XiVO Solutions Documentation¶
XiVO Solutions is a complete solution for entreprise communications and contact centre infrastructure. XiVO is a PABX application, XiVO-CC is an application suite for contact centers.
Table of Contents¶
XiVO Documentation¶
XiVO is an application suite based on several free existing components including Asterisk, and our own developments to provide communication services (IPBX, Unified Messaging, ...) to businesses.
XiVO is free software. Most of its distinctive components, and XiVO as a whole, are distributed under the GPLv3 license.
You may also check the XiVO blog for more information.
XiVO documentation is also available as a downloadable HTML, EPUB or PDF file. See the downloads page for a list of available files or use the menu on the lower right.
Table of Contents¶
Introduction¶
XiVO is a PABX application based on several free existant components including Asterisk and our own developments. XiVO provides a solution for enterprises who wish to replace or add telephone services (PABX).
XiVO is free software. Most of its distinctive components, and XiVO as a whole, are distributed under the GPLv3 license.
XiVO History¶
XiVO was created in 2005 by Sylvain Boily (Proformatique SARL). The XiVO mark is now owned by Avencall SAS after a merge between Proformatique SARL and Avencall SARL in 2010. The XiVO core team works for Proformatique INC in Quebec City since 2010, after Sylvain Boily moved to Quebec city.
XiVO 1.2 was released on February 3, 2012.
Installation¶
Installing the System¶
Please refer to the section Troubleshooting if ever you have errors during the installation.
There are two official ways to install XiVO:
- using the official ISO image
- using a minimal Debian installation and the XiVO installation script
XiVO can be installed on both virtual (QEMU/KVM, VirtualBox, ...) and physical machines. That said, since Asterisk is sensitive to timing issues, you might get better results by installing XiVO on real hardware.
- Download the ISO image. (latest version) (all versions)
- Boot from the ISO image, select
Install
and follow the instructions. You must select a locale with charset UTF-8. - At the end of the installation, you can continue by running the configuration wizard.
During the installation of Debian, only a proxy that supports proxying http/https requests may eventually be entered. Otherwise GPG key of XiVO repository will not be installed and must be added manually:
wget http://mirror.xivo.solutions/xivo_current.key -O - | apt-key add -
XiVO can be installed directly over a 32-bit or a 64-bit Debian jessie. When doing so, you are strongly advised to start with a clean and minimal installation of Debian jessie.
The latest installation image for Debian jessie can be found at https://www.debian.org/releases/jessie/debian-installer.
The installed Debian must:
- use the architecture
i386
oramd64
- have a default locale with charset UTF-8
In case you want to migrate a XiVO from i386
to amd64
, see Migrate XiVO from i386 (32 bits) to amd64 (64 bits).
Once you have your Debian jessie properly installed, download the XiVO installation script and make it executable:
wget http://mirror.xivo.solutions/xivo_install.sh
chmod +x xivo_install.sh
And run it:
./xivo_install.sh
At the end of the installation, you can continue by running the configuration wizard.
The installation script can also be used to install an archive version of XiVO (14.18 or later only). For example, if you want to install XiVO 16.03:
./xivo_install.sh -a 16.03
When installing an archive version, note that:
- versions 14.18 to 15.19 of XiVO can only be installed on a Debian 7 (wheezy) system
- the 64-bit versions of XiVO are only available starting from 15.16
You may also install development versions of XiVO with this script. These versions may be unstable and should not be used on a production server. Please refer to the usage of the script:
./xivo_install.sh -h
It’s also possible to install XiVO by PXE. It is not documented here.
Running the Wizard¶
After the system installation, you must go through the wizard before being able to use your XiVO. Open your browser and enter your server’s IP address in the navigation bar. (For example: http://192.168.1.10)
You then have to accept the GPLv3 License under which XiVO is distributed.
- Enter the hostname (Allowed characters are :
A-Z a-z 0-9 -
) - Enter the domain name (Allowed characters are :
A-Z a-z 0-9 - .
) - Enter the password for the
root
user of the web interface, - Configure the IP address and gateway used by the VoIP interface
- Finally, modify the DNS server information if needed.
Contexts are used for managing various phone numbers that are used by your system.
- The Internal calls context manages extension numbers that can be reached internally
- The Incalls context manages calls coming from outside of your system
- The Outcalls context manages calls going from your system to the outside
- Enter the entity name (e.g. your organization name) (Allowed characters are :
A-Z a-z 0-9 - .
) - Enter the number interval for you internal context. The interval will define the users’s phone numbers for your system (you can change it afterwards)
- Enter the DID range and DID length for your system.
- You may change the name of your outgoing calls context.
Finally, you can validate your configuration by clicking on the Validate
button.
Note that if you want to change one of the settings you can go backwards in the wizard by clicking
on the Previous
button.
Warning
This is the last time the root
password will be displayed. Take care to note it.
Congratulations, you now have a fully functional XiVO server.
To start configuring XiVO, see Getting Started.
Post Installation¶
Here are a few configuration options that are commonly changed once the installation is completed. Please note that these changes are optional.
When you call internally another phone of the system you would like your phone to display the name of the called person (instead of the dialed number only). To achieve this you must change the following SIP options:
:
- Trust the Remote-Party-ID: yes,
- Send the Remote-Party-ID: select
PAI
The caller ID number on incoming calls depends on what is sent by your operator.
You can modify it via the file /etc/xivo/asterisk/xivo_in_callerid.conf
.
Note
The reverse directory lookup use the caller ID number after it has been modified by
xivo_in_callerid.conf
Examples:
If you use a prefix to dial outgoing numbers (like a 0) you should add a 0 to all the
add =
sections,You may want to display incoming numbers in E.164 format. For example, you can change the
[national1]
section to:callerid = ^0[1-9]\d{8}$ strip = 1 add = +33
To enable the changes you have to restart xivo-agid:
service xivo-agid restart
Configure your locale and default time zone device template =>
by editing the default templateConfigure the timezone in =>
If needed, reconfigure your timezone for the system:
dpkg-reconfigure tzdata
You should also select default codecs. It obviously depends on the telco links, the country, the phones, the usage, etc. Here is a typical example for Europe (the main goal in this example is to select only G.711 A-Law instead of both G.711 A-Law and G.711 µ-Law by default):
SIP :
:Customize codec : enabled
Codec list:
G.711 A-Law G.722 G.729A H.264
IAX2 :
:Customize : enabled
Codec list:
G.711 A-Law G.722 G.729A H.264
Getting Started¶
This section will show you how to create a user with a SIP line. This simple use case covers what a lot of people need to start using a phone. You can use these steps for configuring a phone (e.g a softphone, an Analog-to-Digital switch or a SIP phone).
This tutorial doesn’t cover how to automatically provision a supported device. For this, consult the provisionning section.
We first need to log into the XiVO web interface. The web interface is where you can administer the whole system.

Logging into the XiVO
When logged in, you will see a page with all the status information about your system. This page helps you monitor the health of your system and gives you information about your network. Please note the IP address of your server, you will need this information later on when you will configure your device (e.g. phone)

System informations
To configure a device for a user, start by navigating to the IPBX menu. Hover over the Services tab, a dropdown menu will appear. Click on IPBX.

Menu IPBX
Select the Users setting in the left menu.

Users settings
From here, press on the “plus” sign. A pop up will appear where you can click on Add.

Adding a new line
We now have the form that will allow us to create a new user. The three most important fields are ‘First name’, ‘Last name’ and ‘Language’. Fill in the fields and click on Save at the bottom. For our example, we will create a used called ‘Alice Wonderland’.

User information
Afterwards, click on the “Lines” tab.

Lines menu
Enter a number for your phone. If you click inside the field, you will see the range of numbers you can use. For our example, we will use ‘1000’.

Line information
By default, the selected protocol is SIP, which is what we want for now. Click on Save to create the line.

Save
We now have a user named ‘Alice Wonderland’ with the phone number ‘1000’.

User added information
Now we need to go get the SIP username and password to configure our phone. Go back to the IPBX menu on the left, and click on ‘Lines’.

Lines information
You will see a line associated with the user we just created. Click on the pencil icon to edit the line.

Edit line
We can now see the username and password for the SIP line. you can configure your phone using the IP for your server, the username and the password.

General line information
Upgrading¶
Upgrading a XiVO is done by executing commands through a terminal on the server. You can connect to the server either through SSH or with a physical console.
To upgrade your XiVO to the latest version, you must use the xivo-upgrade
script. You can
start an upgrade with the command:
xivo-upgrade
Note
- You can’t use xivo-upgrade if you have not run the wizard yet
- Upgrading from a version prior to XiVO 1.2 is not supported.
- When upgrading XiVO, you must also upgrade all associated XiVO Clients. There is currently no retro-compatibility on older XiVO Client versions.
This script will update XiVO and restart all services.
There are 2 options you can pass to xivo-upgrade:
-d
to only download packages without installing them. This will still upgrade the package containing xivo-upgrade and xivo-service.-f
to force upgrade, without asking for user confirmation
xivo-upgrade
uses the following environment variables:
XIVO_CONFD_PORT
to set the port used to query the HTTP API of xivo-confd (default is 9486)
Upgrade procedure¶
- Consult the roadmaps starting from your current version to the current prod version.
- Read all existing Upgrade Notes (see below) starting from your version to the latest version.
- For custom setups, follow the required procedures described below (e.g. HA cluster).
- To download the packages beforehand, run
xivo-upgrade -d
. This is not mandatory, but it does not require stopping any service, so it may be useful to reduce the downtime of the server while upgrading. - When ready, run
xivo-upgrade
which will start the upgrade process. Telephony services will be stopped during the process - When finished, check that all services are running (the list is displayed at the end of the upgrade).
- Check that services are correctly working like SIP registration, ISDN link status, internal/incoming/outgoing calls, XiVO Client connections etc.
Version-specific upgrade procedures¶
To follow Avencall’s official releases you must switch to xivo.solutions mirrors. In order to do that follow the following procedure:
Download the
switch-to-xivo-solutions
script:wget http://mirror.xivo.solutions/debian/tools/migration-tools/switch-to-xivo-solutions.sh chmod +x ./switch-to-xivo-solutions.sh
Execute the script:
./switch-to-xivo-solutions.sh ... Your XiVO has been switched to xivo.solutions successfully. Votre XiVO a été basculé vers xivo.solutions avec succès.
Update the sources list:
apt-get update
When upgrading from XiVO 14.11 or earlier, you must do the following, before the normal upgrade:
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
In those versions, xivo-upgrade keeps XiVO on the same version. You must do the following, before the normal upgrade:
echo "deb http://mirror.xivo.solutions/debian/ xivo-five main" > /etc/apt/sources.list.d/xivo-upgrade.list \
&& apt-get update \
&& apt-get install xivo-fai \
&& rm /etc/apt/sources.list.d/xivo-upgrade.list \
&& apt-get update
When upgrading from XiVO 13.24 or earlier, you must do the following, before the normal upgrade:
Ensure that the file
/etc/apt/sources.list
is not configured onarchive.debian.org
. Instead, it must be configured with a non-archive mirror, but still on thesqueeze
distribution, even if it is not present on this mirror. For example:deb http://ftp.us.debian.org/debian squeeze main
Add
archive.debian.org
in another file:cat > /etc/apt/sources.list.d/squeeze-archive.list <<EOF deb http://archive.debian.org/debian/ squeeze main EOF
And after the upgrade:
rm /etc/apt/sources.list.d/squeeze-archive.list
When upgrading from XiVO 13.03 or earlier, you must do the following, before the normal upgrade:
wget http://mirror.xivo.solutions/xivo_current.key -O - | apt-key add -
When upgrading from XiVO 12.13 or earlier, you must do the following, before the normal upgrade:
apt-get update
apt-get install debian-archive-keyring
Upgrading from 1.2.0 or 1.2.1 requires a special procedure before executing xivo-upgrade
:
apt-get update
apt-get install xivo-upgrade
/usr/bin/xivo-upgrade
Upgrading a cluster¶
Here are the steps for upgrading a cluster, i.e. two XiVO with High Availability (HA):
On the master : deactivate the database replication by commenting the cron in
/etc/cron.d/xivo-ha-master
On the slave, deactivate the xivo-check-master-status script cronjob by commenting the line in
/etc/cron.d/xivo-ha-slave
On the slave, start the upgrade:
xivo-slave:~$ xivo-upgrade
When the slave has finished, start the upgrade on the master:
xivo-master:~$ xivo-upgrade
When done, launch the database replication manually:
xivo-master:~$ xivo-master-slave-db-replication <slave ip>
Reactivate the cronjobs (see steps 1 and 2)
Upgrading to/from an archive version¶
An archive version refers to a XiVO installation whose version is frozen: you can’t upgrade it until you manually change the upgrade server.
Using archive versions enable you to upgrade your XiVO to a specific version, in case you don’t want to upgrade to the latest (which is not recommended, but sometimes necessary). You will then be able to upgrade your newer archive version to the latest version or to an even newer archive version.
Warning
These procedures are complementary to the upgrade procedure listed in Version-specific upgrade procedures. You must follow the version-specific procedure before running the following procedures.
Archive packages are named as follow:
XiVO version | Archive package name |
---|---|
1.2 to 1.2.12 | pf-fai-xivo-1.2-skaro-1.2.1 |
12.14 to 13.24 | xivo-fai-skaro-13.04 |
13.25 to 14.17 | xivo-fai-14.06 |
14.18+ | packages removed |
Archive version < 13.25:
apt-get update
apt-get install {xivo-fai,xivo-fai-skaro}/squeeze-xivo-skaro-$(cat /usr/share/pf-xivo/XIVO-VERSION)
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
xivo-upgrade
Archive version >= 13.25 and < 14.18:
apt-get update
apt-get install xivo-fai
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
xivo-upgrade
Archive version >= 14.18:
xivo-dist xivo-five
xivo-upgrade
As a result, xivo-upgrade will upgrade XiVO to the latest stable version.
Non-archive version means any “normal” way of installing XiVO (ISO install, script install over pre-installed Debian, xivo-upgrade).
Downgrades are not supported: you can only upgrade to a greater version.
We only support upgrades to archive versions >= 13.25, e.g. you can upgrade a 12.16 to 14.16, but not 12.16 to 13.16
apt-get install xivo-fai-13.25
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
You are now considered in an archived version, see the section Upgrade from an older archive version to a newer archive version below.
xivo-dist xivo-15.12
apt-get update
apt-get install xivo-upgrade/xivo-15.12
xivo-upgrade
Downgrades are not supported: you can only upgrade to a greater version.
We only support upgrades to archive versions >= 13.25, e.g. you can upgrade a 12.16 to 14.16, but not 12.16 to 13.16
cat > /etc/apt/sources.list.d/squeeze-archive.list <<EOF
deb http://archive.debian.org/debian/ squeeze main
EOF
apt-get update
apt-get install {xivo-fai,xivo-fai-skaro}/squeeze-xivo-skaro-1.2.3
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
apt-get update
apt-get install xivo-fai-14.16
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
apt-get update
apt-get install xivo-upgrade/xivo-14.16
cat > /etc/apt/preferences.d/50-xivo-14.16.pref <<EOF
Package: *
Pin: release a=xivo-five
Pin-Priority: -10
Package: *
Pin: release a=xivo-14.16
Pin-Priority: 700
EOF
xivo-upgrade
rm /etc/apt/preferences.d/50-xivo-14.16.pref
rm /etc/apt/sources.list.d/squeeze-archive.list
apt-get update
apt-get update
apt-get install xivo-fai
apt-get purge xivo-fai-13.25
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
apt-get update
apt-get install xivo-fai-14.16
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
apt-get update
apt-get install xivo-upgrade/xivo-14.16
cat > /etc/apt/preferences.d/50-xivo-five.pref <<EOF
Package: *
Pin: release a=xivo-five
Pin-Priority: -10
EOF
xivo-upgrade
rm /etc/apt/preferences.d/50-xivo-five.pref
apt-get update
apt-get install xivo-fai
sed -i 's/xivo\.fr/xivo.solutions/g' /etc/apt/sources.list.d/*.list
apt-get update
apt-get install xivo-dist
xivo-dist xivo-15.11
apt-get purge 'xivo-fai*'
apt-get update
apt-get install xivo-upgrade/xivo-15.11
xivo-upgrade
xivo-dist xivo-15.12
apt-get update
apt-get install xivo-upgrade/xivo-15.12
xivo-upgrade
Upgrading from i386 (32 bits) to amd64 (64 bits)¶
i386
(32 bits) to amd64
(64 bits)¶There is no fully automated method to migrate XiVO from i386
to amd64
.
The procedure is:
- Upgrade your
i386
machine to XiVO >= 15.13 - Install a XiVO
amd64
using the same version as the upgraded XiVO i386 - Make a backup of your XiVO
i386
by following the backup procedure - Copy the backup tarballs to the XiVO
amd64
- Restore the backup by following the restore procedure
Before starting the services after restoring the backup on the XiVO amd64
, you should ensure
that there won’t be a conflict between the two machines, e.g. two DHCP servers on the same broadcast
domain, or both XiVO fighting over the same SIP trunk register. You can disable the XiVO i386
by
running:
xivo-service stop
But be aware the XiVO i386
will be enabled again after you reboot it.
Troubleshooting¶
When upgrading XiVO, if you encounter problems related to the system locale, see PostgreSQL localization errors.
If xivo-upgrade fails or aborts in mid-process, the system might end up in a faulty condition. If in doubt, run the following command to check the current state of xivo’s firewall rules:
iptables -nvL
If, among others, it displays something like the following line (notice the DROP and 5060):
0 0 DROP udp -- * * 0.0.0.0/0 0.0.0.0/0 udp dpt:5060
Then your XiVO will not be able to register any SIP phones. In this case, you must delete the DROP rules with the following command:
iptables -D INPUT -p udp --dport 5060 -j DROP
Repeat this command until no more unwanted rules are left.
Upgrade Notes¶
Consult the 2016.04 Roadmap
Upgrading from 2016.03:
xivo-dist xivo-2016.04
xivo-upgrade
For upgrade from older versions see Version-specific upgrade procedures.
To upgrade to this version, use the switch-to-xivo-solutions
script:
wget http://mirror.xivo.solutions/debian/tools/migration-tools/switch-to-xivo-solutions.sh
chmod +x ./switch-to-xivo-solutions.sh
./switch-to-xivo-solutions.sh
At the end the script should say:
Your XiVO has been switched to xivo.solutions successfully.
Votre XiVO a été basculé vers xivo.solutions avec succès.
Then you have to points towards this version:
xivo-dist xivo-2016.03
apt-get update
And finally do the normal upgrade procedure.
Consult the 16.08.2 release post
This release fixes a number of known issues that were present in the 16.08. Mainly it:
- ships with asterisk 13.10.0 which fixes a number of crashes, instabilities and memory leak,
- fixes a number of upgrade problems,
- fixes a status (presence, phone) update problem in XiVO Client,
- add the possibility to create a XiVO Client login in the format name@domain.tld
Consult the 16.08 Roadmap
CTI Protocol is now in version 2.2
Some security features have been added to the XiVO provisioning server. To benefit from these new features, you’ll need to update your xivo-provd plugins to meet the system requirements.
If you have many phones that are connected to your XiVO through a NAT equipment, you should review the default configuration to make sure that the IP address of your NAT equipment don’t get banned unintentionally by your XiVO.
Newly created groups and queues now ignore call forward requests from members by default. Previously, call forward requests from members were always followed. This only applies to call forward configured directly on the member’s phone: call forward configured via *21 have always been ignored in these cases.
Note that during the upgrade, the previous behaviour is kept for already existing queues and groups.
This behaviour is now configurable per queue/group, via the “Ignore call forward requests from members” option under the “Application” tab. We recommend enabling this option.
Consult the 16.07 Roadmap
- If you were affected by the bug #6213, i.e. if your agent login time statistics were incorrect since your upgrade to XiVO 15.20 or later, and you want to fix your statistics for that period of time, you’ll need to manually apply a fix.
Consult the 16.06 Roadmap
Consult the 16.05 Roadmap
- The
view
,add
,edit
,delete
anddeleteall
actions of the “lines” web service provided by the web interface have been removed. As a reminder, note that the web services provided by the web interface are deprecated.
Consult the 16.04 Roadmap
- CTI Protocol is now in version 2.1
- The field Rightcall Code from under Services tab will overwrite all password call permissions for the user.
- Faxes stored on FTP servers are now converted to PDF by default. See Using the FTP backend if you want to keep the old behavior of storing faxes as TIFF files.
Consult the 16.03 Roadmap
- The new section in the web interface will only be visible by a non-root administrator after adding the corresponding permissions in the administrator configuration.
- Update the switchboard configuration page for the statistics in Configuration for multiple switchboards.
- The API for associating a line to a device has been replaced. Consult the xivo-confd REST API changelog for further details
- The configuration parameters of xivo_ldap_user plugin of xivo-auth has been changed. See xivo_ldap plugin.
- The user’s email is now a unique constraint. Every duplicate email will be deleted during the migration. (This does not apply to the voicemail’s email)
Consult the 16.02 Roadmap
- The experimental xivo_ldap_voicemail plugin of xivo-auth has been removed. Use the new xivo_ldap plugin.
- Bus messages in the xivo exchange are now sent with the content-type application/json. Some libraries already do the message conversion based the content-type. Kombu users will receive a python dictionnary instead of a string containing json when a message is received.
- xivo-ctid encryption is automatically switched on for every XiVO server and XiVO Client >= 16.02. If you really don’t want encryption, you must disable it manually on the server after the upgrade. In that case, XiVO Clients will ask whether to accept the connection the first time.
Consult the 16.01 Roadmap
- The page has been removed. Consequently, every Web Services Access has now all access rights on the web services provided by the web interface. These web services are deprecated and will be removed soon.
- During the upgrade, if no CA certificates were trusted at the system level, all the CA
certificates from the ca-certificates package will be added. This is done to resolve an issue with
installations from the ISO and PXE. In the (rare) case you manually configured the ca-certificates
package to trust no CA certificates at all, you’ll need to manually reconfigure it via
dpkg-reconfigure ca-certificates
after the upgrade. - xivo-ctid uses xivo-auth to authenticate users. See Authentication.
- the service_discovery section of the xivo-ctid configuration has changed. If you have set up Contact and Presence Sharing, you should update your xivo-ctid configuration.
- the CTI Protocol is now versioned and a message will be displayed if the server and a client have incompatible protocol versions.
Consult the 15.20 Roadmap
- Debian has been upgraded from version 7 (wheezy) to 8 (jessie).
- CSV webservices in the web interface have been removed. Please use the xivo-confd REST API instead.
- The CSV import format has been changed. Consult CSV Migration for further details.
- xivo-ctid now uses STARTTLS for the client connections.
- For users already using the CTIS protocol the client can be configured to use the default port (5003)
Please consult the following detailed upgrade notes for more information:
The upgrade to XiVO 15.20 or later will take longer than usual, because the whole Debian system will be upgraded.
The database management system (postgresql) will also be upgraded from version 9.1 to version 9.4 at the same time. This will upgrade the database used by XiVO. This operation should take at most a few minutes.
After the upgrade, the system will need to be rebooted.
If you are upgrading from XiVO 13.24 or earlier, you’ll need to first upgrade to Debian 7 (wheezy) before being able to upgrade to Debian 8 (jessie). To do so, you’ll have to:
- Run
xivo-upgrade
a first time, which will upgrade your XiVO to version 15.19 (Debian 7) - Reboot your system
- Run
xivo-upgrade
a second time, which will upgrade your XiVO to the latest version (Debian 8) - Reboot your system
Consult the Debian 7 (wheezy) Upgrade Notes for more information on the first upgrade.
- Run
Make sure your have sufficient space for the upgrade. You might run into trouble if you have less than 2 GiB available in the file system that holds the /var and / directories.
If you have customized the Debian system of your XiVO in some nontrivial way, you might want to review the official Debian release notes before the upgrade. Most importantly, you should:
- Make sure you don’t have any unofficial sources in your /etc/apt/sources.list or /etc/apt/sources.list.d directory. If you were using the wheezy-backports source, you must remove it.
- Remove packages that were automatically installed and are not needed anymore, by running
apt-get autoremove --purge
. - Purge removed packages. You can see the list of packages in this state by running
dpkg -l | awk '/^rc/ { print $2 }'
and purge all of them withapt-get purge $(dpkg -l | awk '/^rc/ { print $2 }')
- Remove
.dpkg-old
,.dpkg-dist
and.dpkg-new
files from previous upgrade. You can see a list of these files by runningfind /etc -name '*.dpkg-old' -o -name '*.dpkg-dist' -o -name '*.dpkg-new'
.
Check that customization to your configuration files is still effective.
During the upgrade, new version of configuration files are going to be installed, and these might override your local customization. For example, the vim package provides a new
/etc/vim/vimrc
file. If you have customized this file, after the upgrade you’ll have both a/etc/vim/vimrc
and/etc/vim/vimrc.dpkg-old
file, the former containing the new version of the file shipped by the vim package while the later is your customized version. You should merge back your customization into the new file, then delete the.dpkg-old
file.You can see a list of affected files by running
find /etc -name '*.dpkg-old'
. If some files shows up that you didn’t modify by yourself, you can ignore them.Purge removed packages. You can see the list of packages in this state by running
dpkg -l | awk '/^rc/ { print $2 }'
and purge all of them withapt-get purge $(dpkg -l | awk '/^rc/ { print $2 }')
If you had customizations in one of these files:
/etc/default/asterisk
/etc/default/consul
/etc/default/xivo-ctid
Then you’ll need to review your customizations to make sure they still work with systemd. This is necessary since these 3 files aren’t read under systemd.
For
/etc/default/asterisk
, only the CONFD_* options are automatically migrated to/etc/systemd/system/asterisk.service.d/auto-sysv-migration.conf
.For
/etc/default/consul
, only the WAIT_FOR_LEADER and CONFIG_DIR options are automatically migrated to/etc/systemd/system/consul.service.d/auto-sysv-migration.conf
.For
/etc/default/xivo-ctid
, only the XIVO_CTID_AMI_PROXY option is automatically migrated to/etc/systemd/system/xivo-ctid.service.d/auto-sysv-migration.conf
.Reboot your system. It is necessary for the upgrade to the Linux kernel and init system (systemd) to be effective.
Here’s a non-exhaustive list of changes that comes with XiVO on Debian 8:
In Debian 7, the
halt
command powered off the machine. In Debian 8, the command halts the system, but does not power off the machine. To halt the machine and turn it off, use thepoweroff
orshutdown
command.With the init system switch from SysV to systemd, you should now use the
systemctl
command to manage services (i.e. start/stop/status) instead of theservice
command or/etc/init.d/<service>
, although these two methods should still work fine.If you are new to systemd, you can find some basic usage on the systemd page of the Debian Wiki.
The bootlogd package is not installed by default anymore, since it is not needed with systemd. If you want to see the boot messages, use the
journalctl -b
command instead.The virtual terminals (tty1 to tty6) now shows up earlier during the boot, before all services have been started.
The way the ami-proxy is configured for xivo-ctid has changed. If your XiVO was using the ami-proxy, the configuration will be automatically upgraded.
Customization to asterisk and consul startup is now done by customizing the systemd unit file (by creating a drop-in file for example) instead of editing the
/etc/default/asterisk
and/etc/default/consul
files. These files are not used anymore.
If your system is using a swap partition or file and is using more memory than it can fit in the RAM, then system power-off or reboot might hangs indefinitely. This is due to a limitation in the current systemd version.
If you find yourself in this case, you should try allocating more RAM to your system. Otherwise, you can try stopping the xivo services using
xivo-service stop
before rebooting to lessen the likelihood of this problem.
This page describes how to migrate CSV files from the legacy format to the new format. Consult the API documentation on user imports for further details.
- Only data encoded as UTF-8 will be accepted
- The pipe delimiter (
|
) has been replaced by a comma (,
)- Double-quotes (
"
) must be escaped by writing them twice (e.gRobert ""Bob"" Jenkins
)
Fields have been renamed in the new CSV format. Use the following table to rename your fields. Fields marked as N/A are no longer supported.
Old name New name entityid entity_id firstname firstname lastname lastname language language outcallerid outgoing_caller_id mobilephonenumber mobile_phone_number agentnumber N/A bosssecretary N/A callerid N/A enablehint supervision_enabled enablexfer call_transfer_enabled enableclient cti_profile_enabled profileclient cti_profile_name username username password password phonenumber exten context context protocol line_protocol linename sip_username linesecret sip_secret incallexten incall_exten incallcontext incall_context incallringseconds incall_ring_seconds voicemailname voicemail_name voicemailnumber voicemail_number voicemailcontext voicemail_context voicemailpassword voicemail_password voicemailemail voicemail_email voicemailattach voicemail_attach_audio voicemaildelete voicemail_delete_messages voicemailaskpassword voicemail_ask_password
Consult the 15.19 Roadmap
- The sound file
/usr/share/asterisk/sounds/fr_FR/une.wav
has been moved to/usr/share/asterisk/sounds/fr_FR/digits/1F.wav
. - If you would like to use the new “transfer to voicemail” feature
from the People Xlet, you’ll need to update your directory definition and your directory display, i.e.:
- edit your “internal” directory definition (Services / CTI server / Directories / Definitions) and add a field “voicemail” with value “voicemail_number”
- edit your display (Services / CTI server / Directories / Display filters) and add a row with title “Voicemail”, field type “voicemail” and field name “voicemail”
- restart xivo-dird
- It is now possible to send an email to a user with a configured email address in the people xlet. See Views to add the appropriate field to your configured displays.
- The Contacts xlet (aka. Search) has been removed in favor of the People Xlet. You may need to do some manual configuration in the directories for the People Xlet to be fully functional. See the detailed upgrade notes for more details.
- If you need context separation in the People Xlet, you will have to manually configure xivo-dird to keep it working, see Context separation. This procedure is only temporary, later versions will handle the context separation automatically.
- xivo-agentd now uses mandatory token authentication for its REST API. If you have custom development using this service, update your program accordingly.
- Some actions that used to be available in the contact xlets are not
implemented in the people xlet yet.
- Cancel transfer is only available using the switchboard xlet
- Hanging up a call is only possible using the switchboard xlet
- Call interception is not available anymore
- Conference room invitation is not available anymore
Please consult the following detailed upgrade notes for more information:
When upgrading your XiVO to 15.19, there are some features in the directories that could not be upgraded automatically, because it risked breaking some manual configurations.
After you upgrade your XiVO, your CTI displays in
may look like this:
You should update your displays to make them look like:

This will give you a Xlet People looking like this:


You can find more details about the field types in Integration of XiVO dird with the rest of XiVO.
Without context separation, you only need one contact source for all the users of your XiVO.
However, if you need context separation, each context is considered as a separate independant source of contacts, each with a different context filter. For this, you need:
- one contact source per context (a file in
/etc/xivo-dird/sources.d
), so that we have a source containing only the contacts from one context - one profile per context (equivalent to ) so that users in one context only see people from the same context.
Each source should look like this one, e.g. the context is named INSIDE
:
confd_config:
host: localhost
https: false
port: 9487
timeout: 4
verify_certificate: false
version: '1.1'
first_matched_columns: [exten]
format_columns:
directory: "R\xE9pertoire XiVO Interne"
location: '{description}'
mobile: '{mobile_phone_number}'
name: '{firstname} {lastname}'
number: '{exten}'
sda: '{userfield}'
voicemail: '{voicemail_number}'
searched_columns: [firstname, lastname, userfield, description]
type: xivo
unique_column: id
name: internal_INSIDE # <--- each source has a different name, one per context
extra_search_params:
context: INSIDE # <--- each source filters users according to one context
The parameters in this file have the same effect than
and put together.You may generate these config files from xivo-confgen dird/sources.yml
. Be sure to have name
and extra_search_params
correct for each source file.
Now that we have our contact sources, we need our search profiles.
Create a new file to override the profiles generated by xivo-confgen. You only need one file, which will define all your profiles at once.
xivo-confgen dird/services.yml >> /etc/xivo-dird/conf.d/001-context-separation.yml
In this file, there is a list of services (favorites, lookup, ...) where each profile has a set of sources. You need to match one profile to the right internal source for each service. For example, to have context separation between contexts INSIDE and INDOORS:
services:
favorites:
__default_phone:
sources: [xivodir, internal, ldaptest, personal]
__switchboard_directory:
sources: [xivodir, ldaptest, personal]
INSIDE:
sources: [xivodir, internal_INSIDE, ldaptest, personal] # <--- profile INSIDE uses the source internal_INSIDE
INDOORS:
sources: [xivodir, internal_INDOORS, ldaptest, personal] # <--- profile INDOORS uses the source internal_INDOORS
lookup:
__default_phone:
sources: [xivodir, internal, ldaptest, personal]
__switchboard_directory:
sources: [xivodir, ldaptest, personal]
INSIDE:
sources: [xivodir, internal_INSIDE, ldaptest, personal] # <--- same HERE
INDOORS:
sources: [xivodir, internal_INDOORS, ldaptest, personal] # <--- and HERE
Consult the 15.18 Roadmap
- The provd_pycli command (deprecated in 15.06) has been removed in favor of xivo-provd-cli. If you have custom scripts referencing provd_pycli, you’ll need to update them.
- The xivo-agentctl command (deprecated in 15.06) has been removed in favor of xivo-agentd-cli. If you have custom scripts referencing xivo-agentctl, you’ll need to update them.
- xivo-agentd now uses HTTPS. If you have custom development using this service, update your configuration accordingly. The xivo-agentd-client library, used to interact with xivo-agentd, has also been updated to use HTTPS by default.
- xivo-confd ports 50050 and 50051 have been removed. Please use 9486 and 9487 instead
Configuration File Upgrade Notes
The file format of configuration files for daemons exposing an HTTP/S API has changed. The following services have been affected :
- xivo-agentd
- xivo-amid
- xivo-auth
- xivo-confd
- xivo-ctid
- xivo-dird
- xivo-dird-phoned
Ports and listening addresses are now organised in the following fashion:
rest_api:
https:
enabled: true
port: 9486
listen: 0.0.0.0
certificate: /usr/share/xivo-certs/server.crt
private_key: /usr/share/xivo-certs/server.key
ciphers: "ALL:!aNULL:!eNULL:!LOW:!EXP:!RC4:!3DES:!SEED:+HIGH:+MEDIUM"
http:
enabled: true
port: 9487
listen: 127.0.0.1
If you have any custom configuration files for these daemons, please modify them accordingly. Consult Network for further details on which network services are available for each daemon.
Consult the 15.17 Roadmap
- Online call recording is now done via automixmon instead of automon. This has no impact unless you have custom dialplan that is passing directly the “w” or “W” option to the Dial or Queue application. In these cases, you should modify your dialplan to pass the “x” or “X” option instead.
- The remote directory service available from supported phones is now provided by the new unified directory service, i.e. xivo-dird. Additional upgrade steps are required to get the full benefit of the new directory service; see the detailed upgrade notes.
- The field
enableautomon
has been renamed toenableonlinerec
in the users web services provided by the web-interface (these web services are deprecated). - The agent status dashboard now shows that an agent is calling or receiving a non ACD call while in wrapup or paused.
- SIP endpoints created through the REST API will not appear in the web interface until they have been associated with a line
- Due to limitations in the database, only a limited number of optional parameters can be configured on a SIP endpoint. Consult the xivo-confd REST API changelog for further details
Please consult the following detailed upgrade notes for more information:
If you are not using the remote directory from your phones, you can safely skip this page.
Starting from XiVO 15.17, the remote directory used by the phones is now provided by the new directory service, composed principally of xivo-dird and xivo-dird-phoned. It was previously provided by the XiVO web interface.
This brings a few changes for the administrators, the biggest one being that lookup from both the XiVO client and phones are now configured at the same place, namely the (incorrectly named) Directories page.
section, with some advanced configuration only available in the configuration files. This means that lookup from the phones can now also display results from CSV or web services directories. For details on how to configure directories, refer to theFor users, the biggest change is that they can now consult their personal contacts (that they added from their XiVO client) when doing a search from their phone.
The following options have been removed from the web interface, in the
page:- the
Phone number type
field - the
Attributes
tab
The phone number type is now configurable on a per source basis (and for all type of source, not
just LDAP), in telephoneNumber
that you want to be displayed on your phone with
the suffix “(Office)”, just make sure that your directory definition is configured with a field
named phone_office
with the value {telephoneNumber}
.
By default, the following fields are available:
phone
: doesn’t add a suffixphone_office
: add a “(Office)” suffixphone_mobile
: add a “(Mobile)” suffixphone_home
: add a “(Home)” suffixphone_other
: add a “(Other)” suffix
Note
These fields will automatically be added in your LDAP directory definitions during the upgrade, so you may only need to review your directory configuration.
This list of fields and the suffix associated to it is currently only configurable in the xivo-dird configuration files, in the views/displays_phone section.
This is causing 2 functional changes:
- Previously, the suffix displayed was translated in function of the phone’s language. This is not possible anymore, and you’ll have to edit the configuration files if you want the suffix to be in a different language than english.
- For “custom” phone number type, you’ll have to add a new entry in the configuration files and add the correspond field in the directory definition.
In XiVO 15.16, the Attributes
tab would allow a “fallback” mechanism, where if an LDAP attribute
for a record was missing/empty, another attribute would be used. In XiVO 15.17, this mechanism is
available (for all type of sources) by mapping the first attribute to a field name phone
, the
second to a field name phone1
, etc. The fallback mechanism is available on the fields phone
,
phone_office
, phone_mobile
, phone_home
, phone_other
and display_name
.
The following options have been removed from the web interface, in the
page:- the
LDAP filters
tab
LDAP sources used for lookup from the phone are now selected in the same place as for the XiVO client, i.e. in
. A consequence of that is that it’s not possible anymore to have sources only used for lookup from phone and other sources only used for lookup from the XiVO client.Note
The LDAP filters that were used for phone lookup will be automatically added to all the profiles during the upgrade.
After upgrading your XiVO to 15.17 or later, you should do the following steps.
This step is optional, although strongly recommended.
For the users to be able to search their personal contacts from their phone, the phone configuration needs to be updated. This means:
- Installing new xivo-provd plugins or upgrading existing plugins
- Restarting all affected phones
See the provisioning section for more information on installing or upgrading plugins.
Here’s the list of plugins which have received modifications to be compatible with the new directory service:
Name | Version |
---|---|
xivo-aastra-3.3.1-SP4 | 1.5 |
xivo-aastra-4.1.0 | 1.5 |
xivo-cisco-sccp-9.0.3 | 0.8 |
xivo-cisco-sccp-cipc-2.1.2 | 0.8 |
xivo-cisco-sccp-legacy | 0.8 |
xivo-cisco-sccp-wireless-1.4.5 | 0.8 |
xivo-cisco-spa-7.5.5 | 0.12 |
xivo-cisco-spa-legacy | 0.12 |
xivo-polycom-4.0.4 | 1.4 |
xivo-polycom-5.3.0 | 1.5 |
xivo-snom-8.7.5.17 | 1.5 |
xivo-technicolor-ST2022-4.78-1 | 0.4 |
xivo-technicolor-ST2030-2.74 | 0.3 |
xivo-technicolor-TB30-1.74.0 | 0.3 |
xivo-yealink-v70 | 1.24 |
xivo-yealink-v72 | 1.24 |
xivo-yealink-v73 | 1.24 |
xivo-yealink-v80 | 1.24 |
Plugins with greater version number or greater firmware-version number are also compatible.
If the xivo-provd plugins are not updated or the phone are not rebooted, the user will by default
only be able to search in the “internal” and “xivodir” directory definitions. If you want to add or
remove sources for these phones, you’ll need to edit xivo-dird configuration files. More precisely,
you’ll need to edit the sources associated to the profile named default_phone
.
If there’s a firewall (or a NAT equipement) between your XiVO and your phones, you must know that the port used for the directory lookup from the phone has changed from port TCP/80 to port TCP/9498. The new port is going to be used only by phones which are using a compatible plugins (see list above) and have been rebooted; otherwise, the port TCP/80 will still be used.
During the upgrade, new LDAP directory definitions might be created and fields to existing one might be added.
For example, if you had an LDAP filter which was used for directory lookup from your phones, then a
corresponding LDAP directory definition will be created if nonexistent, and otherwise be updated to
make sure the display_name
and phone_office
(or another field, depending on the phone number
type of your LDAP filter) fields are defined. The directory definition will also be added to all the
direct directories entries, i.e. added to all items in the page.
If you were using LDAP filters with custom phone number types, the custom part will be lost, and to get back the same behaviour, you’ll need to modify xivo-dird configuration files and update the field’s name in your directory definition.
Also, if you have other directory defintions that you now want to use from your phones (e.g. CSV
directories), make sure that their configuration is working, i.e. that they have a display_name
and phone
fields. During the upgrade, these fields are automatically added to the directory
defintion “xivodir”, “internal” and for LDAP source, like described above.
Consult the 15.16 Roadmap
- The directory column type “mobile” was removed in favor of the new “callable” type. If you have hand-written configuration files for xivo-dird, in section “views”, subsection “displays”, all keys “type” with value “mobile” must be changed to value “callable”.
- The
xivo-auth
backend interface has changed,get_acls
is nowget_consul_acls
. All unofficial back ends must be adapted and updated. No action is required for “normal” installations. - Voicemails can now be deleted even if they are associated to a user.
Consult the 15.15 Roadmap
Voicemail Upgrade Notes
- Voicemail webservices in the web interface have been removed. Please use the xivo-confd REST API instead.
- Voicemail IMAP configuration has been migrated to the new
Advanced
tab.- Voicemail option
Disable password checking
has been converted toAsk password
. The value has also been inverted. (e.g. IfDisable password checking
was false,Ask password
is true.)Ask password
is activated by default.- After an upgrade, if ever you have errors when searching for voicemails, please try clearing cookies in your web browser.
- A voicemail must be dissociated from any user prior to being deleted. Voicemail are dissociated by editing the user and clicking on the
Delete voicemail
button in theVoicemail
tab. This constraint will disappear in future versions.- Deleting a user will dissociate any voicemail that was attached, but will not delete it nor any messages.
- Creating a line is no longer necessary when attaching a voicemail to a user.
- The following fields have been modified when importing a CSV file:
Old name | New name | Required ? | New default value |
---|---|---|---|
voicemailmailbox | voicemailnumber | yes | |
voicemailskippass | voicemailaskpassword | no | 1 |
voicemailcontext | yes |
Directories
- Concatenated fields in directories are now done in the directory definitions instead of the displays
- The field column in directory displays are now field names from the directory definition. No more {db-*} are required
- In the directory definitions fields can be modified using a python format string with the fields comming from the source.
- Most of the configuration for xivo-dird is now generated from xivo-confgen using the values in the web interface.
- The remote directory xlet has been removed in favor of the new people xlet.
See Directories and Integration of XiVO dird with the rest of XiVO for more details
- Consult the 15.14 Roadmap
- Default password for
xivo-polycom-4.0.4
plugin version >= 1.3 is now 9486 (i.e. the word “xivo” on a telephone keypad). - Default password for
xivo-polycom-5.3.0
plugin version >= 1.4 is now 9486. - Caller id management for users in confd has changed. Consult the xivo-confd REST API changelog.
- The Local Directory Xlet is replaced with the People Xlet. Contacts are automatically migrated to the server. Note that the CSV format for importing contacts has changed (see People Xlet for more information).
- Consult the 15.13 Roadmap
- Asterisk has been upgraded from version 11.17.1 to 13.4.0, which is a major Asterisk upgrade.
- An ARI user has been
added to
/etc/asterisk/ari.conf
. If you have configured Asterisk HTTP server to bind on a publicly reachable address (in/etc/asterisk/http.conf
), then you should update your configuration to prevent unauthorized access on your Asterisk. - The xivo-dird configuration option source_to_display_columns has been removed in favor of the new option format_columns. All source configuration using the source_to_display_columns must be updated. A migration script will automatically modify source configuration in the /etc/xivo-dird/sources.d directory.
Please consult the following detailed upgrade notes for more information:
You might be impacted by the upgrade to Asterisk 13 if you have:
- custom dialplan
- custom Asterisk configuration
- custom application using AGI, AMI or any other Asterisk interface
- custom application exploiting CEL or queue_log
- custom Asterisk modules (e.g. codec_g729a.so)
- customized Asterisk in some other way
- DAHDI trunks using SS7 signaling
If you find yourself in one of these cases, you should make sure that your customizations still work with Asterisk 13.
If you are upgrading from Asterisk 1.8, you should also check the Asterisk 1.8 to 11 upgrade notes.
Some of the more common changes to look for:
SS7 support is not available in the Asterisk package of XiVO between version 15.13 and 16.08 inclusively.
All channel and global variable names are evaluated in a case-sensitive manner. In previous versions of Asterisk, variables created and evaluated in the dialplan were evaluated case-insensitively, but built-in variables and variable evaluation done internally within Asterisk was done case-sensitively.
The SetMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use the CHANNEL function’s
musicclass
setting instead.The WaitMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use MusicOnHold with a
duration
parameter instead.The SIPPEER dialplan function no longer supports using a colon as a delimiter for parameters. The parameters for the function should be delimited using a comma.
The SIPCHANINFO dialplan function was deprecated and has been removed. Users of the function should use the CHANNEL function instead.
For SIP, the codec preference order in an SDP during an offer is slightly different than previous releases. Prior to Asterisk 13, the preference order of codecs used to be:
- Our preferred codec
- Our configured codecs
- Any non-audio joint codecs
Now, in Asterisk 13, the preference order of codecs is:
- Our preferred codec
- Any joint codecs offered by the inbound offer
- All other codecs that are not the preferred codec and not a joint codec offered by the inbound offer
Queue strategy
rrmemory
(Round robin memory) now has a predictable order. Members will be called in the order that they are added to the queue. For agents, this means they will be called in the order they are logged.When performing queue pause/unpause on an interface without specifying an individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at least one member of any queue exists for that interface. This has an impact on the agent performance statistics; an agent must be a member of at least 1 queue for its pause time to show up in the statistics.
You can see the complete list of changes from the Asterisk website:
- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12
- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13
- http://git.asterisk.org/gitweb/?p=asterisk/asterisk.git;a=blob;f=CHANGES;h=d0363f7c3b03cec5f71b3806535c4f9d2b2baa02;hb=refs/heads/13
The AGI protocol did not change between Asterisk 11 and Asterisk 13; if you have custom AGI applications, you only need to make sure that the dialplan applications and functions you are using from the AGI are still valid.
List of known bugs and limitations for Asterisk 13 in XiVO:
When direct media is active and DTMF are sent using SIP INFO, DTMF are not working properly. It is also impossible to do an attended transfer from the XiVO client in these conditions.
- Consult the 15.12 Roadmap
- The certificate used for HTTPS in the web interface will be regenerated if the default certificate was used. Your browser will complain about the new certificate, and it is safe to accept it (see #3656). See also HTTPS certificate.
- If you have an HA configuration, then you should run
xivo-sync -i
on the master node to setup file synchronization between the master and the slave. File synchronization will then be done automatically every hour via rsync and ssh. - xivo-auth and xivo-dird now use HTTPS, if you have custom development using these services, update your configuration accordingly.
- Consult the 15.11 Roadmap
- The call records older than 365 days will be periodically removed. The first automatic purge will occur in the night after the upgrade. See Purge Logs for more details.
- Consult the 15.10 Roadmap
- Consult the 15.09 Roadmap
- Consult the 15.08 Roadmap
- The Dialer Xlet has been integrated in Identity Xlet.
- Consult the 15.07 Roadmap
- Consult the 15.06 Roadmap
- The provd client has been moved into a new python package, xivo_provd_client. If you have custom scripts using this client, you’ll need to update them. See http://projects.xivo.io/issues/5469 for more information.
- The provd_pycli command name has been deprecated in favor of xivo-provd-cli. These 2 commands do the same thing, the only difference being the name of the command. The provd_pycli command name will be removed in 15.18, so if you have custom scripts referencing provd_pycli, you’ll need to update them.
- The xivo-agentctl command name has been deprecated in favor of xivo-agentd-cli. These 2 commands do the same thing, the only difference being the name of the command. The xivo-agentctl command name will be removed in 15.18, so if you have custom scripts referencing xivo-agentctl, you’ll need to update them.
- Consult the 15.05 Roadmap
- The Xlet identity has been modified to follow the new XiVO Client design which implies the removal of some details.
- Consult the 15.04 Roadmap
- Consult the 15.03 Roadmap
- Consult the 15.02 Roadmap
- Consult the 15.01 Roadmap
- The confd REST API is now more restrictive on HTTP headers. Particularly, the
headers Accept and Content-Type must be set to (typically)
application/json
. - The following configuration files have been created:
/etc/xivo-agid/config.yml
/etc/xivo-call-logd/config.yml
/etc/xivo-amid/config.yml
/etc/xivo-agentd/config.yml
- Consult the 14.24 Roadmap
The following security vulnerability has been fixed:
- XIVO-2014-01: Queues and groups permit callers to make unwanted calls
- Consult the 14.23 Roadmap
- The “waiting calls / logged agents ratio” queue diversion scenario has been renamed to “number of waiting calls per logged agents”.
- A new community section was added to the official documentation for all user-contributed documentation.
- Consult the 14.22 Roadmap
- The sheet event Dial on queues is now only sent to the ringing agent. The sheet is also sent a little later during the call, when the ringing agent is known.
- Consult the 14.21 Roadmap
- The confd REST API is now accessible via HTTPS on port 9486 and via HTTP on port 9487 (localhost only). These ports are replacing the 50051 and 50050 ports respectively. It will still be possible to access the confd REST API via the 50051 and 50050 ports for the next year, but you are advised to update your confd REST API clients as soon as possible.
- The old (unsupported) ami-proxy is now replaced by an ami-proxy built in xivo-ctid. You must uninstall the old ami-proxy before activating the built-in version. See troubleshooting xivo-ctid to learn how to activate.
- Consult the 14.20 Roadmap
- Default parameters for all Cisco SPA ATA plugins have changed to be better suited for european faxes.
- Following the POODLE attack (CVE-2014-3566), SSL 3.0 has been disabled for the web interface and the xivo-confd REST API.
If you have Aastra phones and are using the remote directory on them, consult the following detailed upgrade notes:
Starting from XiVO 14.20, it is not possible anymore to use SSL 3.0 when connecting to XiVO using HTTPS.
This has the unfortunate consequence of breaking the remote directory on Aastra phones
configured by the xivo-aastra
provisioning plugins in version 1.2 and earlier.
To be able to use the remote directory on your Aastra phones on XiVO 14.20 or later, you’ll need to take one of the following actions:
This is the recommended solution. This can be done either before or after the upgrade. You’ll have to:
- Upgrade your
xivo-aastra
plugin to version 1.3 or later - Restart/synchronize all your phones
The correction is only available for plugin xivo-aastra-3.3.1-SP2
and later. If you are using an
older plugin (xivo-aastra-3.2.2-SP3
for example), then you’ll need to install a newer plugin
and update all your phones to use the new plugin.
If you were already using custom templates, make sure to update them so that the phones access the remote directory via HTTP instead of HTTPS. This can be done using the following command:
find /var/lib/xivo-provd/plugins/xivo-aastra* -name '*.tpl' -exec sed -i '/X_xivo_phonebook_ip/s/\bhttps:/http:/' {} \;
If you can’t or don’t want to update to a newer plugin, you can instead update the templates used by the plugin. This can be done either before or after the upgrade. You’ll have to:
- Update the templates so that the directory is accessed via HTTP
- Restart/synchronize all your phones
In this specific case, it is safe to directly modify the templates used by the plugin instead of creating custom templates. To update the templates, you can use the following command:
find /var/lib/xivo-provd/plugins/xivo-aastra* -name '*.tpl' -exec sed -i '/X_xivo_phonebook_ip/s/\bhttps:/http:/' {} \;
If you can’t restart/synchronize your phones, the last solution is to re-enable SSL 3.0 on your XiVO. This should only be used as a temporary solution to give you more time to plan a firmware upgrade for your phones. This can be done only after the upgrade. You’ll have to:
- Update nginx configuration
- Reload nginx
This can be done using the following commands:
sed -i 's/ssl_protocols .*/ssl_protocols SSLv3 TLSv1 TLSv1.1 TLSv1.2;/' /etc/nginx/sites-available/xivo
service nginx reload
- Consult the 14.19 Roadmap
- Consult the 14.18 Roadmap
- xivo-fai packages were replaced with xivo-dist : a new tool to handle repositories sources. Upon upgrade, xivo-dist is installed and run and all xivo-fai packages are purged. Consult xivo-dist use cases
- Consult the 14.17 Roadmap
- DAHDI configuration file
/etc/dahdi/modules
is no more created by default and must now be maintained manually. No action is needed upon upgrade but be aware that the upstream sample file is now available in/usr/share/dahdi/modules.sample
. See dahdi modules documentation for detailed info. - The new CCSS feature will not be enabled upon upgrade, you must explicitly enable it in the menu.
- Consult the 14.16 Roadmap
- See the changelog for xivo-confd’s REST API
- DAHDI is upgraded to 2.10.0. If the upgrade process asks about
/etc/dahdi/modules
, we recommend that you keep the old version of the file. - Asterisk now inserts CEL and queue log entries via the ODBC asterisk modules instead of the pgsql modules.
Consult the 14.15 Roadmap
Duplicate function keys will be deleted upon upgrade. If multiple function keys pointing to the same destination are detected for a given user, only the one with the lowest position will be kept. To see the list of deleted function keys, check the xivo-upgrade log file such as:
grep MIGRATE_FK /var/log/xivo-upgrade.log
These notes only apply to:
- Digium TE133/TE131 cards that are in firmware version 780017 or earlier
- Digium TE435/TE235 cards that are in firmware version e0017 or ealier
Warning
The system will need to be power cycled after the upgrade. Your cards will not be usable until then.
First, you need to install the latest firmware for your TE133/TE131 or TE435/TE235 cards:
xivo-fetchfw install digium-te133
xivo-fetchfw install digium-te435
Then stop all the services and reload the DAHDI modules. Reloading the DAHDI module might take up to 30 seconds:
xivo-service stop
service dahdi stop
service dahdi start
Following this manipulation, you should see something similar at the end of the /var/log/messages
file:
dahdi: Telephony Interface Unloaded
dahdi: Version: 2.9.2
dahdi: Telephony Interface Registered on major 196
wcte13xp 0000:03:0c.0: Firmware version 780017 is running, but we require version 780019.
wcte13xp 0000:03:0c.0: firmware: agent loaded dahdi-fw-te133.bin into memory
wcte13xp 0000:03:0c.0: Found dahdi-fw-te133.bin (version: 780019) Preparing for flash
wcte13xp 0000:03:0c.0: Uploading dahdi-fw-te133.bin. This can take up to 30 seconds.
wcte13xp 0000:03:0c.0: Delaying reset. Firmware load requires a power cycle
wcte13xp 0000:03:0c.0: Running firmware version: 780017
wcte13xp 0000:03:0c.0: Loaded firmware version: 780019 (Will load after next power cycle)
wcte13xp 0000:03:0c.0: FALC version: 5
wcte13xp 0000:03:0c.0: Setting up global serial parameters for T1
wcte13xp 0000:03:0c.0: VPM450: firmware dahdi-fw-oct6114-032.bin not available from userspace
For the firmware update to complete, you must halt the machine (a reboot won’t be enough) before restarting it.
- Consult the 14.14 Roadmap
- See the changelog for REST API
- Upon an important freeze of Asterisk, Asterisk will be restarted. See the associated ticket for more information.
- Consult the 14.13 Roadmap
- See the changelog for REST API
- Skills-based routing: for an agent which doesn’t have the skill X, the rule X < 10 was previously evaluated to true, since not having the skill X was equivalent to having it with a value of 0. This behaviour has changed, and the same expression is now evaluated to false. If you are using skills-based routing, you’ll need to check that your rules are still doing what you expect. See skill evaluation for more information.
- Consult the 14.12 Roadmap
- All provisioning plugins were modified. Although not mandatory, it is strongly advised to update all used plugins.
- The function key ‘Activate voicemail’ was removed as it was a duplicate of existing function key ‘Enable voicemail’. All users having the ‘Activate voicemail’ function key will have to be reconfigured with a ‘Enable voicemail’ function key in order to keep the equivalent feature.
- Log files have changed for the following daemons (previously in
/var/log/daemon.log
):- xivo-provd:
/var/log/xivo-provd.log
- xivo-agid:
/var/log/xivo-agid.log
- xivo-sysconfd:
/var/log/xivo-sysconfd.log
- xivo-provd:
- Consult the 14.11 Roadmap
- The API URL
/lines/<id>/extension
is now deprecated. Use/lines/<id>/extensions
instead.
- Consult the 14.10 Roadmap
- Custom MOH have been fixed, but can not be used for playing uploaded files anymore. See Music on Hold.
- Consult the 14.09 Roadmap
- REST API 1.0 is no more. All code, tests and documentation was removed from XiVO. All code developped for REST API 1.0 must now be adapted to use REST API 1.1.
- Consult the 14.08 Roadmap
- The
xivo
database has been merged into theasterisk
database. The database schema has also been altered in a way that it might make the upgrade longer than usual.
Please consult the following detailed updated notes for more information:
The xivo
database has been merged into the asterisk
database in XiVO 14.08.
This has an impact on:
- The restore procedure. There’s only one database to restore now. Also, the procedure to restore the data while keeping the system configuration has been updated.
- The data that is replicated between the master and the slave in a high availability cluster.
Previously, all the configuration that was under the “Configuration” menu of the web interface was not replicated between the master and slave. This is now replicated, except for:
- HA settings
- All the network configuration (i.e. everything under the section)
- All the support configuration (i.e. everything under the section)
The call center statistics have also been excluded from the replication.
The way the replication is done has also been updated, which makes it faster.
When upgrading to XiVO 14.08, the database schema will be altered.
This will result in a longer upgrade time if you have a lots of rows in the queue_log table.
You can see the number of rows in your queue_log table with:
sudo -u postgres psql -c "SELECT count(*) FROM queue_log" asterisk
On ordinary hardware, you can expect that it will take ~10 minutes for every 2.5 million of rows. So if you have 5 million of rows in your queue_log table, you can expect that the upgrade will take an extra 20 minutes.
It is possible to reduce the amount of additional time the upgrade will take by either removing rows from the table or altering the table before the upgrade.
Both these commands can be run while the XiVO services are up.
For example, if you want to remove all the rows before march 2014, you can use:
sudo -u postgres psql -c "DELETE FROM queue_log WHERE \"time\" < '2014-03-01'" asterisk
If you want to alter the table before the upgrade, you can use:
sudo -u postgres psql -c "ALTER TABLE queue_log ADD COLUMN id SERIAL PRIMARY KEY; GRANT ALL ON SEQUENCE queue_log_id_seq TO asterisk" asterisk
Note
It is recommended to execute this command when there’s no activity on the system.
The way the database is initially provisioned and the way it is altered during an upgrade has also been changed.
In XiVO 14.07 and earlier, the database was provisioned by executing
the /usr/share/xivo-manage-db/datastorage/asterisk.sql
SQL script.
Starting with XiVO 14.08, the xivo-init-db
is responsible for provisioning
the database. This script should not be used by an administrator in normal
circumstance.
Starting with XiVO 14.08, database migration are done with the help of
alembic instead
of the asterisk-XXX.sql and xivo-XXX.sql scripts. The alembic migration
scripts can be found inside the /usr/share/xivo-manage-db
directory.
Otherwise, the xivo-check-db
and xivo-update-db
commands have been
updated to work with both the old and the new systems and are still the official
way to check the database state and update the database respectively.
- Consult the 14.07 Roadmap
- Configuration for phones used for the switchboard has changed.
Please consult the following detailed updated notes for more information:
The xivo-aastra-switchboard and xivo-snom-switchboard plugins have been removed and their functionalities are now provided by the generic xivo-aastra and xivo-snom plugins respectively.
The upgrade is not done automatically, so please follow the Upgrade Procedure section below.
Although you are strongly advised to upgrade your switchboard phone configuration, backwards compatibility with the old system will be maintained.
Note that if you need to install a switchboard for a previous version of XiVO, the old xivo-aastra-switchboard and xivo-snom-switchboard plugins can be found in the archive repository.
This procedure should be executed after the upgrade to 14.07 or later: the options used in this procedure are not available in versions before 14.07.
The following upgrade procedure suppose that you are using an Aastra phone as your switchboard phone. The same upgrade procedure apply for Snom phones, with the only difference being the different plugin name.
- Update the list of installable plugins.
- Install the latest xivo-aastra plugin, or upgrade it to the latest version if it is already installed.
- Install the needed language files and firmware files.
- For each phone used for the switchboard, change the plugin and activate the switchboard option:
- Select the generic xivo-aastra plugin.
- Check the “switchboard” checkbox.
- Synchronize the phone.
- Once this is completed, you can uninstall the xivo-aastra-switchboard plugin.
An unofficial script that automates this procedure is also available on github:
cd /tmp
wget --no-check-certificate https://raw.githubusercontent.com/xivo-pbx/xivo-tools/master/scripts/migrate_switchboard_1407.py
python migrate_switchboard_1407.py
- Consult the 14.06 Roadmap
- The XiVO client now uses Qt 5 instead of Qt 4. There is nothing to be aware of unless you are building your own version of it.
- Consult the 14.05 Roadmap
- The CTI Protocol has been updated.
- The specification of the ‘answered-rate’ queue statistic has changed to exclude calls on a closed queue
- The switchboard can now choose which incoming call to answer
- The package versions do not necessarily contain the current XiVO version, it may contain older
versions. Only the package
xivo
is guaranteed to have the current XiVO version.
Please consult the following detailed updated notes for more information:
These notes only apply to Digium TE133 or TE134 cards that are in firmware version 770017 or earlier.
Warning
The system will need to be power cycled after the upgrade. Your cards will not be usable until then.
First, you need to install the latest firmware for your TE133 or TE134 cards:
xivo-fetchfw install digium-te133
xivo-fetchfw install digium-te134
Then stop all the services and reload the DAHDI modules. Reloading the DAHDI module might take up to 30 seconds:
xivo-service stop
service dahdi stop
service dahdi start
Following this manipulation, you should see something similar at the end of the /var/log/messages
file:
dahdi: Telephony Interface Unloaded
dahdi: Version: 2.9.0
dahdi: Telephony Interface Registered on major 196
wcte13xp 0000:03:0c.0: Firmware version 6f0017 is running, but we require version 780017.
wcte13xp 0000:03:0c.0: firmware: agent loaded dahdi-fw-te134.bin into memory
wcte13xp 0000:03:0c.0: Found dahdi-fw-te134.bin (version: 780017) Preparing for flash
wcte13xp 0000:03:0c.0: Uploading dahdi-fw-te134.bin. This can take up to 30 seconds.
wcte13xp 0000:03:0c.0: Delaying reset. Firmware load requires a power cycle
wcte13xp 0000:03:0c.0: Running firmware version: 6f0017
wcte13xp 0000:03:0c.0: Loaded firmware version: 780017 (Will load after next power cycle)
wcte13xp 0000:03:0c.0: FALC version: 5
wcte13xp 0000:03:0c.0: Setting up global serial parameters for T1
wcte13xp 0000:03:0c.0: VPM450: firmware dahdi-fw-oct6114-032.bin not available from userspace
wcte13xp 0000:03:0c.0: Found a Wildcard TE132/TE134 (SN: 1TE134F - DF05132600690 - B1 - 20130702)
For the firmware update to complete, you must halt the machine (a reboot won’t be enough) before restarting it.
Important modification have been made to the internal structure of the SCCP channel driver, xivo-libsccp.
The modifications mostly affect administrators; users are not affected.
Major changes are:
- Improved support for live modifications; no more manual intervention in the asterisk CLI is needed.
- Improved handling of concurrency; crash and deadlock due to concurrency problems should not occur anymore.
The following commands have been removed because they were not needed:
- sccp resync
- sccp set directmedia
- sccp show lines
- sccp update config
The behavior of the following commands have been changed:
module reload chan_sccp
reloads the module configuration, without interrupting the telephony service. A device will only be resetted/restarted if needed, and only once the device is idle. Some changes don’t even require the device to be resetted.sccp show config
output format has been changed a little.sccp show devices
only show the connected devices instead of all the devices. This might change in the future. To get a list of all the devices, usesccp show config
.
The format of the sccp.conf
configuration file has been changed. This
will only impact you if you are using xivo-libsccp without using XiVO.
The format has been changed because the module is now using the ACO module from asterisk, which expect configuration file to have a specific format.
See sccp.conf.sample for a configuration file example.
Each SCCP session/connection now use 3 file descriptors instead of 1 previously. On XiVO, the file descriptor limit for the asterisk process is 8192, which means that the increase in used file descriptors should not be a problem, even on a large installation.
- Consult the 14.04 Roadmap
- Live reload of the configuration can be enabled and disabled using the REST API
- The generation of call logs for unanswered calls from the XiVO client have been improved.
- Consult the 14.03 Roadmap
- A migration script adds an index on the linkedid field in the cel table. Tests have shown that this operation can last up to 11.5 minutes on a XiVO Corporate with 18 millions CELs. xivo-upgrade will thus be slightly longer.
- Two new daemons are now operationnal, xivo-amid and xivo-call-logd:
- xivo-amid constantly reads the AMI and sends AMI events to the RabbitMQ bus
- xivo-call-logd generates call-logs in real time based on AMI LINKEDID_END events read on the bus
- An increase in load average is expected with the addition of these two new daemons.
- The cron job calling xivo-call-logs now runs once a day at 4:25 instead of every 5 minutes.
- Consult the 14.02 Roadmap
- PHP Web services has been removed from documentation
- REST API 1.0 Web services has been removed from documentation
- REST API 1.1 User-Line-Extension service is replaced by User-Line and Line-Extension services
- Consult the 14.01 Roadmap
- The following paths have been renamed:
/etc/pf-xivo
to/etc/xivo
/var/lib/pf-xivo
to/var/lib/xivo
/usr/share/pf-xivo
to/usr/share/xivo
You must update any dialplan or configuration file using these paths
- Consult the 13.25 Roadmap
- Debian has been upgraded from version 6 (squeeze) to 7 (wheezy).
Please consult the following detailed upgrade notes for more information:
- The upgrade will take longer than usual, because the whole Debian system will be upgraded
- The system must be restarted after the upgrade, because the Linux kernel will also be upgraded
In case XiVO is using a LDAP server through SSL/TLS (LDAPS), the documentation instructed you to
append the certificate to /etc/ssl/certs/ca-certificates.crt
. However, this is the wrong way
to add a new certificate, because it will be erased by the upgrade.
To keep your certificate installed through the upgrade, you must follow the instructions given in the LDAP documentation.
GRUB installations on cloned virtual machines may lead to unbootable systems, if not fixed properly before restarting the system. If xivo-upgrade detects your system is in a broken state, it will display a few commands to repair the GRUB installation.
- Consult the 13.24 Roadmap
- Default Quality of Service (QoS) settings have been changed for SCCP. The IP packets containing audio media are now marked with the EF DSCP.
- Consult the 13.23 Roadmap
- The New call softkey has been removed from SCCP phones in connected state. To start a new call, the user will have to press Hold then New call. This is the same behavior as a Call Manager.
- Some softkeys have been moved on SCCP phones. We tried to keep the keys in the same position at any given time. As an example, the transfer key will not become End call while transfering a call. Note that this is a work in progress and some models still need some tweaking.
- Consult the 13.22 Roadmap
- PostgreSQL will be upgraded from 9.0 to 9.1. The upgrade of XiVO will take longer than usual, depending on the size of the database. Usually, the database grows with the number of calls processed by XiVO. The upgrade will be stopped if not enough space is available on the XiVO server.
- Consult the 13.21 Roadmap
- It is no more possible to delete a device associated to a line using REST API.
- Consult the 13.20 Roadmap
- xivo-libsccp now supports direct media on wifi phone 7920 and 7921
- xivo-confd now implements a voicemail list
- Since XiVO 13.18 was not released, the 13.19 release contains all developments of both 13.18 and 13.19, therefore please consult both Roadmaps :
- Consult the 13.19 Roadmap
- Consult the 13.18 Roadmap
- Call logs are now generated automatically, incrementally and regularly. Call logs generated before 13.19 will be erased one last time.
- The database was highly modified for everything related to devices : table devicefeatures does not exist anymore and now relies on information from xivo-provd.
- Consult the 13.17 Roadmap
- There is a major change to call logs. They are no longer available as a web report but only as a csv export. See the call logs documentation. Furthermore, call logs are now fetched from xivo-confd REST API.
- Paging group numbers are now exclusively numeric. All non-numeric paging group numbers are converted to their numeric-only equivalent while upgrading to XiVO 13.17 ( *58 becomes 58, for example).
- Consult the 13.16 Roadmap
- A migration script modifies the user and line related-tables and the way users, lines and extensions are associated. As a consequence of this script, it is not possible any more to associate a user and a line without extensions. Existing associations between users and one or more lines having no extensions will be removed. Users and lines will still exist unassociated.
- The call logs page is able to display partial results of big queries, instead of displaying a blank page.
- Two new CEL messages are now enabled : LINKEDID_END and BRIDGE_UPDATE. Those events will only exist in CEL for calls passed after upgrading to XiVO 13.16.
- The new REST API now makes possible to associate multiple user to a given line and/or extension. There are currently some limitations on how those users and lines can be manipulated using the web interface.
- There was no production release of XiVO 13.15. All 13.15 developments are included in the official 13.16 release.
- Consult the 13.14 Roadmap
- The latest Polycom plugin enables the phone lock feature with a default user password of ‘123’. All Polycom phones used with XiVO also have a default admin password. In order for the phone lock feature to be secure, one should change every phone’s admin AND user passwords.
- WebServices for SIP trunks/lines: field
nat
: valueyes
changed toforce_rport,comedia
- The database has beed updated in order to remove deprecated tables (generalfeatures, extenumbers, extenhash, cost_center).
- Consult the 13.13 Roadmap
- Consult the 13.12 Roadmap
- CTI protocol: Modified values of agent
availability
. Read CTI Protocol changelog - Clean-up was made related to the minimization of the XiVO Client. Some visual differences have been observed on Mac OS X that do not affect the XiVO Client in a functional way.
- Consult the 13.11 Roadmap
- Asterisk has been upgraded from version 11.3.0 to 11.4.0
API changes:
- Dialplan variable XIVO_INTERFACE_0 is now XIVO_INTERFACE
- Dialplan variable XIVO_INTERFACE_NB and XIVO_INTERFACE_COUNT have been removed
- The following fields have been removed from the lines and users web services
- line_num
- roles_group
- rules_order
- rules_time
- rules_type
- Consult the 13.10 Roadmap
API changes:
- CTI protocol: for messages of class
getlist
and functionupdateconfig
, theconfig
object/dictionary does not have arules_order
key anymore.
- Consult the 13.09 Roadmap
- The Restart CTI server link has been moved from to .
- The Agent Status Dashboard has been optimized.
- The Directory xlet can now be used to place call.
- Consult the 13.08 Roadmap
- asterisk has been upgraded from version 1.8.21.0 to 11.3.0, which is a major asterisk upgrade.
- The switchboard’s queue now requires the xivo_subr_switchboard preprocess subroutine.
- A fix to bug #4296 introduced functional changes due to the order in which sub-contexts are included. Please refer to ticket for details.
Please consult the following detailed upgrade notes for more information:
Table of modules that were available in the asterisk 1.8 package but that are not available anymore in the asterisk 11 package:
Name | Description | Loaded in AST1.8 | Asterisk Status | Replaced By |
---|---|---|---|---|
app_dahdibarge | Barge in on DAHDI channel application | Yes | Deprecated | app_chanspy |
app_readfile | Stores output of file into a variable | Yes | Deprecated | func_env (FILE()) |
app_saycountpl | Say polish counting words | Yes | Deprecated | say.conf |
app_setcallerid | Set CallerID Presentation Application | Yes | Deprecated | func_callerid |
cdr_sqlite | SQLite CDR Backend | No | Removed | cdr_sqlite3_custom |
chan_gtalk | Gtalk Channel Driver | No | Deprecated | chan_motif |
chan_jingle | Jingle Channel Driver | No | Deprecated | chan_motif |
chan_vpb | Voicetronix API driver | No | Supported | |
format_sln16 | Raw Signed Linear 16KHz Audio support | Yes | Removed | format_sln |
res_ais | SAForum AIS | No | Removed | res_corosync |
res_jabber | AJI - Asterisk Jabber Interface | No | Deprecated | res_xmpp |
List of modules that were loaded in asterisk 1.8 but that are not loaded anymore in asterisk 11 (see modules.conf):
- res_calendar.so
- res_calendar_caldav.so
- res_calendar_ews.so
- res_calendar_exchange.so
- res_calendar_icalendar.so
- res_config_sqlite.so
- res_stun_monitor.so
List of debian packages that are not available anymore for asterisk 11:
- asterisk-config
- asterisk-mysql
- asterisk-web-vmail
Note
These packages were not installed by default for asterisk 1.8.
If you are using some custom dialplan or AGIs, it is your responsibility to make sure it still works with asterisk 11. See the External Links for more information.
The switchboard’s queue now uses a preprocess subroutine named xivo_subr_switchboard. This preprocess subroutine will be associated with all queues named __switchboard that have no preprocess subroutine defined before the upgrade.
If your switchboard queue is named anything other than __switchboard you should add the preprocess subroutine manually.
If your switchboard queue already has a preprocess subroutine, you should add a Gosub(xivo_subr_switchboard) to you preprocess subroutine.
Warning
This change is only applied to the switchboard distribution queue, not the queue for calls on hold.
- Consult the 13.07 Roadmap
- Agent Status Dashboard has more features and less limitations. See related agent status dashboard documentation
- XiVO call centers have no more notion of ‘disabled agents’. All previously disabled agents in web interface will become active agents after upgrading.
- asterisk has been upgraded from version 1.8.20.1 to 1.8.21.0. Please note that in XiVO 13.08, asterisk will be upgraded to version 11.
- DAHDI has been upgraded from version 2.6.1 to 2.6.2.
- libpri has been upgraded from version 1.4.13 to 1.4.14.
- PostgreSQL upgraded from version 9.0.4 to 9.0.13
- Consult the 13.06 Roadmap
- The new Agent Status Dashboard has a few known limitations. See related dashboard xlet known issues section
- Status Since counter in xlet list of agents has changed behavior to better reflect states of agents in queues as seen by asterisk. See Ticket #4254 for more details.
- Consult the 13.05 Roadmap
- The bug #4228 concerning BS filter only applies to 13.04 servers installed from scratch. Please upgrade to 13.05.
- The order of softkeys on SCCP phones has changed, e.g. the Bis button is now at the left.
- Consult the 13.04 Roadmap
- Upgrade procedure for HA Cluster has changed. Refer to Specific Procedure : Upgrading a Cluster.
- Configuration of switchboards has changed. Since the directory xlet can now display any column from the lookup source, a display filter has to be configured and assigned to the __switchboard_directory context. Refer to Directory xlet documenttion.
- There is no more context field directly associated with a call filter. Boss and secretary users associated with a call filter must necessarily be in the same context.
- Consult the 12.24 Roadmap
- XiVO 12.24 has some limitations mainly affecting the contact center features due to the rewriting of the code handling agents.
Please consult the following detailed upgrade notes for more information:
In order to fix problems related to Asterisk freezing through the chan_agent module, XiVO 12.24 implements a new way of managing agents.
The contact center XiVO 12.24 does not implement all the features available in 12.22. Therefore, you must not upgrade your XiVO if you depend on these features. These features will be reimplemented in the future starting with version 13.01.
- Skill-based routing
- Penalities
- Call listening
Agents must be logged out for the following operations:
- Adding or removing agents from the queues
- When changing the name of a queue (only the name, not the displayed name)
You can logoff all the agents with the following command:
xivo-agentctl -c "logoff all"
Subroutines on users are currently no longer executed when an agent receives a call from the queue
HA for the contact center is not supported for the moment. When switching from a master to a slave, you must relog all your agents.
The “Available” / “In use” statuses for agents that are logged in do not work for the moment.
In XiVO 12.22, an agent is seen as “In use” when:
- The agent’s phone is ringing or has answered a call coming only from a queue
In XiVO 12.24:
- The agent’s phone is “In use” no matter where the call comes from
In XiVO 12.22, an agent is seen as “Available” when:
- The agent is not in pause/wrapup and his phone isn’t ringing/in conversation for a call coming from a queue
In XiVO 12.24:
- The agent is not in pause/wrapup and his phone is in the “idle” state
The “Agent linked” event no longer exists in XiVO 12.24. xivo-upgrade will automatically migrate “Agent linked” / “Agent unlinked” sheets to the “linked” / “unlinked” event.
There is no longer a difference between a “static” or “dynamic” membership. All agent memberships are now considered “static”. Membership changes between the web interface and the XiVO client are now synchronized.
Please note that when upgrading, the following actions will take place automatically:
- All agents will be logged off before migrating
- All agents with a “dynamic” membership will be removed from their queues
Another change is in effect beginning with XiVO 12.24: the field
profileclient
in the CSV user import sees its values change.
Old value | New value |
---|---|
client | Client |
agent | Agent |
switchboard | Switchboard |
agentsup | Supervisor |
oper | removed |
clock | removed |
XiVO Client¶
What is the XiVO Client¶
The XiVO Client is an application that you install on your computer and is connected to the XiVO server. This application offers the following features:
- search contacts and show their presence, phone status
- make calls through your phone (the XiVO Client is NOT a softphone, it is complementary to the phone)
- access your voicemail through your phone
- enable call forwards, call filtering
- show the history of your calls
- list conference rooms and members
- send faxes
It also offers some call center features:
- show screen popups or open URLs when you receive/answer a call
- list agents, queues, calls in queues
- login/logoff, pause/unpause other agents (for supervisors)
- listen/whisper to agents through you phone (for supervisors)
A lot of those features are modular and may be enabled for each user by choosing which Xlets they can see.
Getting the XiVO client¶
Binaries of the XiVO Client are available on our mirror. (latest version) (all versions)
Warning
The installed version of the XiVO Client must match the XiVO server’s version installation. With our current architecture, there is no way to guarantee that the XiVO server will be retro-compatible with older versions of the XiVO Client. Non-matching XiVO server and XiVO Clients versions might lead to unexpected behaviour.
Choose the version you want and in the right directory, get :
- the
.exe
file for Windows - the
.deb
file for Ubuntu or Debian (i386 or amd64, depending on your computer) - the
.dmg
file for Mac OS
For Windows, double-click on the file and follow the instructions. You can also install it silently:
xivoclient-14.XX-x86.exe /S
For Ubuntu/Debian, double-click on the file or execute the following command:
$ gdebi xivoclient-*.deb
For Mac OS, double-click on the file and drag-and-drop the inner file on the Application entry of the Finder.
The XiVO Client should then be available in the applications menu of each platform.
If you want to build your own XiVO Client, see Building the XiVO Client.
Connection to the server¶
To connect to the server using the XiVO client you need a user name, a password and the server’s address. Optionally, it is possible to login an agent while connecting to the server.
Xlets¶
Xlets are features of the XiVO Client. It is the contraction of XiVO applets. To select which xlets are displayed in your client, see CTI profiles.
The Conference room list tab show all available conference rooms configured on the XiVO. The user can click on a conference number to join the conference. When a user joins a conference, his phone will ring and the conference will be joined when the user answers the phone.
When clicking on a conference room a new tab is opened for the selected conference room. The new tab contains information about the members of the conference. The name and number of the member will be displayed when available. Users can also mute and unmute themselves using the microphone icon on the left.
Warning
This xlet should only be used with a Switchboard profile. It is not meant to be used alone.
The goal of the directory xlet is to allow the user to search through XiVO users, directory entries and arbitrary numbers to be able to call and transfer calls to these destinations.

The list of entries in the xlet is searched using the top field. Entries are filtered by column content. The entry list will initially appear as empty.
If the current search term is a valid number, it will be displayed in the result list with no name to allow transfer to numbers that are not currently in the phonebook or configured on the XiVO.
- Users available
- Users ringing
- Users talking
- Users
- Mobile phone
- External contacts
- Current search (not a contact)
Phonebook searches are triggered after the user has entered 3 characters. Results from remote directories will appear after 1 second.
If a directory entry as the same number as a mobile or a phone configured on the XiVO, it’s extra columns will be added to the corresponding entry instead of creating a new line in the search result.
For example:
If User 1 has number 1000 and is also in a configured LDAP with a location in “Québec”, if the display filter contains the Location column, the entry for User 1 will show “Québec” in the Location column after the search results are received.
The directory xlet needs a special context named __switchboard_directory. In
add a new context with the following parameters :- Name :
__switchboard_directory
- Type of context : Other
- Display name : Switchboard

A new display filter must be created for the directory xlet.

The following fields must be configured with the correct value for the Field type column in order for entries to be displayed in the xlet:
- status is the column that will be used to display the status icon, the title can be empty
- name is displayed in the Name column of the xlet
- number_office is displayed in the Number column with a phone icon in the xlet
- number_mobile is displayed in the Number column with a mobile icon in the xlet
- number_... any other field starting with number_ will be displayed in the Number column of the xlet with a generic directory icon
- Any other field will be displayed in their own column of the directory xlet
The values in the Field name column must contain values that were created in the Directory definition.
The title used for the Number column is the title of the first field whose type starts with number_.
Note
The field title of the first number column will be used for the header title in the xlet.
Warning
Make sure that the fields entered in the display format are also available in the directory definition, otherwise the filter will not return any results
The new Display filter has to be assigned to the __switchboard_directory context

You can then choose which directories will be searched by the Xlet.
Warning
You must not select internal directory, as it is already handled.
To search in ldap directories, you must have an LDAP server configured. See LDAP for more details.
If you already have an LDAP filter configured for the Remote directory Xlet, you can use it.
If not, please refer to Add a LDAP Filter.
You must use the new LDAP filter in the Context and filter association step.
The Choose a file to send field is used to select which file you want to send. Only PDF documents are supported.
The Choose destination number field is the fax destination, directory search can be used to find the fax number in available directories.
The history xlet allow the user to view his last calls. The user can filter by sent, received and missed calls.

The user can click on the number to initiate a new call with a given correspondent.
Warning
- The column content is only refreshed when moving from one view to the other.
- The Sent calls tab displays only the phone number of the called party (the name column will be void).
The Identity Xlet allows you to make calls from your computer, via your phone. This means that you can enter the number that you want to dial on your computer, then your phone rings and when you answer it, the called phone will ring.

You can enter the number you want to dial in the text box and then click the button or press enter to dial it.
If you dial an invalid extension (a number is an extension), your phone will ring and you will be told that the extension is not valid.
The People XLet lists the people of your company and personal contacts, giving you access to their phone, status and other information configured by the administrator.

- Display results of the search
- Display favorite contacts
- Search contacts
- Call a contact
- Transfer a call to a contact
- Transfer a call to a contact’s voicemail
- Chat with a contact
- Send an email to a contact
- Bookmark/unmark the contact as a favorite

- View all personal contacts
- Edit or remove a personal contact
- Create a personal contact
- Import personal contacts from a CSV file
- Export personal contacts to a CSV file
- Delete all personal contacts

- XiVO Client status (see Presence Option)
- Phone status (see page)
- Agent status (logged in or logged out)
Note
Most information (e.g. columns displayed, allowed actions, searched directories, etc.) is configurable through the web interface.
Imported files should have the following structure:
firstname,lastname,number,email,company,fax,mobile
John,Doe,5555551111,my@email,xivo,5555552222,5555553333
- The field order is not important.
- The file must be encoded in
UTF-8
. - Invalid lines of the CSV file will be skipped and an error will be displayed in the import report.
The file has the same structure as the import file, with a supplementary field: id
, which is the
internal contact ID from XiVO.
- The first line (the list of field names) is ordered in alphabetical order.
- The file will be encoded in
UTF-8
.
It is possible to copy a contact’s number or email address to the system’s clipboard. To do so, right click on a contact’s action menu and select the value you wish to copy.
Note
When using a mac without a right mouse button use ctrl-Left click to show the copy menu.
The service xlet allows the user to enable and disable telephony services such as call forwarding, call filter and do not disturb.

The available service list is configured from the web interface in
.The right side of the Services section contains services that are available to a given profile.

Configuration¶
The XiVO Client configuration options can be accessed under
.This page allows the user to set his network information to connect to the xivo-ctid server.
- Server is the IP address of the server.
- Backup server is the IP address of the backup server.
- Port is the port on which xivo-ctid is listening for connections. (default: 5003)
- STARTTLS is used to specify that a secure connect should be used
Note
To use STARTTLS, the server needs to be configured to accept encrypted connection.
Handling callto: and tel: URLs¶
The XiVO Client can handle telephone number links that appear in web pages. The client will automatically dial the number when you click on a link.
Note
You must already be logged in for automatic dialing to work, otherwise the client will simply start up and wait for you to log in.
Warning
The option in the XiVO Client
must be disabled, else you will launch one new XiVO Client with every click.callto:
links will work out-of-the-box in Safari and other web browsers
after installing the client.
tel:
links will open FaceTime after installing the client. To make the
XiVO Client the default application to open tel:
URLs in Safari.
- Open the FaceTime application
- Connect using your apple account
- Open the FaceTime preferences
- Change the Default for calls entry to xivoclient.app

Note
The tel:
URL works out-of-the-box in versions of mac osx before 10.10.
XiVO Client is associated with callto:
and tel:
upon installation. Installing other
applications afterward could end up overriding these associations. Starting with Windows Vista, it is possible
to configure these associations via the Default Programs. Users can access Default Programs from Control
Panel or directly from the Start menu.
The following popups might appear when you open a callto:
or tel:
link for the first time in
Internet Explorer:


Simply click on allow to dial the number using the XiVO Client.
Note
If you do not want these warnings to appear each time, do not forget to check/uncheck the checkbox at the bottom of the popups.
Currently, callto:
or tel:
links are only supported in Firefox. There is no configuration
needed.
Currently, callto:
or tel:
links are only supported in Firefox. If the XiVO Client is not
listed in the proposition when you open the link, browse your files to find
/usr/bin/xivoclient
.
If, for some reason, Firefox does not recognize callto:
or tel:
URIs you can manually
associate them to the XiVO Client using the following steps:
- Type
about:config
in the URL bar - Click the I’ll be careful, I promise ! button to close the warning
- Right-click anywhere in the list and select New -> Boolean
- Enter
network.protocol-handler.external.callto
as preference name - Select
false
as value - Repeat steps 3 to 6, but replace
callto
bytel
at step 4
The next time that you click on a telephone link, Firefox will ask you to choose an application. You will then be able to choose the XiVO client for handling telephone numbers.
System¶
DHCP Server¶
XiVO includes a DHCP server used for assisting in the provisioning of phones and other devices. (See Basic Configuration for the basic setup). This section contains additional notes on how to configure more advanced options that may be helpful when integrating the server with different VOIP subnets.
DHCP Server can be activated through the XiVO Web Interface
:By default, it will only answer to DHCP requests coming from the VoIP subnet (defined in the
eth0
After saving your modifications, click on Apply system configuration so that the new settings can take effect.
By default, the XiVO DHCP server uses the XiVO’s IP address as the routing address. To change this you must create a custom-template:
Create a custom template for the
dhcpd_subnet.conf.head
file:mkdir -p /etc/xivo/custom-templates/dhcp/etc/dhcp/ cd /etc/xivo/custom-templates/dhcp/etc/dhcp/ cp /usr/share/xivo-config/templates/dhcp/etc/dhcp/dhcpd_subnet.conf.head .
Edit the custom template:
vim dhcpd_subnet.conf.head
In the file, replace the string
#XIVO_NET4_IP#
by the routing address of your VoIP network, for example:option routers 192.168.2.254;
Re-generate the dhcp configuration:
xivo-update-config
DHCP server should have been restarted and should now use the new routing address.
By default, the XiVO DHCP server serves only known hosts. That is:
- either hosts which MAC address prefix (the OUI) is known
- or hosts which Vendor Identifier is known
Known OUIs and Vendor Class Identifiers are declared in /etc/dhcp/dhcpd_update/*
files.
If you want your XiVO DHCP server to serve also unknown hosts (like PCs) follow these instructions:
Create a custom template for the
dhcpd_subnet.conf.tail
file:mkdir -p /etc/xivo/custom-templates/dhcp/etc/dhcp/ cd /etc/xivo/custom-templates/dhcp/etc/dhcp/ cp /usr/share/xivo-config/templates/dhcp/etc/dhcp/dhcpd_subnet.conf.tail .
Edit the custom template:
vim dhcpd_subnet.conf.tail
And add the following line at the head of the file:
allow unknown-clients;
Re-generate the dhcp configuration:
xivo-update-config
DHCP server should have been restarted and should now serve all network equipments.
If your telephony devices aren’t located on the same site and the same broadcast domain as the XiVO DHCP server, you will have to add the option DHCP Relay to the site’s router. This parameter will allow the DHCP requests from distant devices to be transmitted to the IP address you specify as DHCP Relay.
Warning
Please make sure that the IP address used as DHCP Relay is the same as one of XiVO’s interfaces, and that this interface is configured to listen to DHCP requests (as decribed in previous part). Also verify that routing is configured between the distant router and the choosen interface, otherwise DHCP requests will never reach the XiVO server.
This section describes how to configure XiVO to serve other subnets that the VOIP subnet. As you can’t use the Web Interface to declare other subnets (for example to address DATA subnet, or a VOIP subnet that isn’t on the same site that XiVO server), you’ll have to do the following configuration on the Command Line Interface.
First thing to do is to create a directory and to copy into it the configuration files:
mkdir /etc/dhcp/dhcpd_sites/
cp /etc/dhcp/dhcpd_subnet.conf /etc/dhcp/dhcpd_sites/dhcpd_siteXXX.conf
cp /etc/dhcp/dhcpd_subnet.conf /etc/dhcp/dhcpd_sites/dhcpd_lanDATA.conf
Note
In this case we’ll create 2 files for 2 differents subnets. You can change the name of the files,
and create as many files as you want in the folder /etc/dhcp/dhcpd_sites/
. Just adapt
this procedure by changing the name of the file in the different links.
After creating one or several files in /etc/dhcp/dhcpd_sites/
, you have to edit the file
/etc/dhcp/dhcpd_extra.conf
and add the according include statement like:
include "/etc/dhcp/dhcpd_sites/dhcpd_siteXXX.conf";
include "/etc/dhcp/dhcpd_sites/dhcpd_lanDATA.conf";
Once you have created the subnet in the DHCP server, you must edit each configuration file (the
files in /etc/dhcp/dhcpd_sites/
) and modify the different parameters. In section
subnet, write the IP subnet and change the following options (underlined fields in the
example):
subnet 172.30.8.0 netmask 255.255.255.0 {
subnet-mask:
option subnet-mask 255.255.255.0;
broadcast-address:
option broadcast-address 172.30.8.255;
routers (specify the IP address of the router that will be the default gateway of the site):
option routers 172.30.8.1;
In section pool, modify the options:
pool {
log (add the name of the site or of the subnet):
log(concat("[", binary-to-ascii(16, 8, ":", hardware), "] POOL VoIP Site XXX"));
range (it will define the range of IP address the DHCP server can use to address the devices of that subnet):
range 172.30.8.10 172.30.8.200;
Warning
XiVO only answers to DHCP requests from supported devices. In case of
you need to address other equipment, use the option allow unknown-clients; in the
/etc/dhcp/dhcpd_sites/
files
At this point, you can apply the changes of the DHCP server with the command:
service isc-dhcp-server restart
After that, XiVO will start to serve the DHCP requests of the devices located on other sites or
other subnets than the VOIP subnet. You will see in /var/log/daemon.log
all the DHCP
requests received and how they are handled by XiVO.
Mail¶
This section describes how to configure the mail server shipped with XiVO (Postfix) and the way XiVO handles mails.
In
, the following options can be configured:- Domain Name messaging : the server’s displayed domain. Will appear in “Received” mail headers.
- Source address of the server : domain part of headers “Return-Path” and “From”.
- Relay SMTP and FallBack relay SMTP : relay mail servers.
- Rewriting shipping addresses : Canonical address Rewriting. See Postfix canonical documentation for more info.
Warning
Postfix, the mail server shipped with XiVO, should be stopped on an installed XiVO with no valid and reachable DNS servers configured. If Postfix is not stopped, messages will bounce in queues and could end up affecting core pbx features.
If you need to disable Postfix here is how you should do it:
systemctl stop postfix
systemctl disable postfix
If you ever need to enable Postfix again:
systemctl enable postfix
systemctl start postfix
Alternatively, you can empty Postfix’s queues by issuging the following commands on the XiVO server:
postsuper -d ALL
Network¶
This section describes how to configure additional network devices that may be used to better accomodate more complex network infrastructures. Network interfaces are managed in the XiVO web interface via the page
.XiVO offers 2 types of interfaces: VoIP and Data. The VoIP interface is used by the DHCP server, provisioning server, and phone devices connected to your XiVO. These services will use the information provided on the VoIP interface for their configuration. For example, the DHCP server will only listen on the VoIP interface by default.
To change these settings, you must either create a new interface or edit an existing one and change its type. When adding a new VoIP interface, the type of the old one will automatically be changed to Data.
In this example, we’ll add and configure the eth1 network interface on our XiVO.
First, we see there’s already an unconfigured network interface named eth1 on our system:

To add and configure it, we click on the small plus button next to it, and we get to this page:

In our case, since we want to configure this interface with static information (i.e. not via DHCP), we fill the following fields:

Note that since our eth0 network interface already has a default gateway,
we do not enter information in the Default gateway
field for our eth1 interface.
Once the changes have been saved, the action Apply network configuration will appear in bold. This action must be clicked in order for the changes to take effect.

Apply after modify interface
In this example, the XiVO already has 2 network interfaces configured:

Listing the network interfaces
To add and configure a new VLAN interface, we click on the small plus button in the top right corner,

and we get to this page:

In our case, since we want to configure this interface with static information:

Click on Save list the network interfaces:

- The new virtual interface has been successfully created.
Note
Do not forget after you finish the configuration of the network to apply it with the button: Apply network configuration
After applying the network configuration:

Network configuration successfully apply
Static routes cannot be added via the web interface. However, you may add static routes to your XiVO by following following the steps described below. This procedure will ensure that your static routes are applied at startup (i.e. each time the network interface goes up).
Create the file
/etc/network/if-up.d/xivo-routes
:touch /etc/network/if-up.d/xivo-routes chmod 755 /etc/network/if-up.d/xivo-routes
Insert the following content:
#!/bin/sh if [ "${IFACE}" = "<network interface>" ]; then ip route add <destination> via <gateway> ip route add <destination> via <gateway> fi
Fields <network interface>, <destination> and <gateway> should be replaced by your specific configuration. For example, if you want to add a route for 192.168.50.128/25 via 192.168.17.254 which should be added when eth0 goes up:
#!/bin/sh if [ "${IFACE}" = "eth0.2" ]; then ip route add 192.168.50.128/25 via 192.168.17.254 fi
Note
The above check is to ensure that the route will be applied only if the correct interface goes up. This check should contain the actual name of the interface (i.e. eth0 or eth0.2 or eth1 or ...). Otherwise the route won’t be set up in every cases.
Warning
Manually changing the MTU is risky. Please only proceed if you are aware of what you are doing.
These steps describe how to change the MTU:
#. Create the file :file:`/etc/network/if-up.d/xivo-mtu`::
touch /etc/network/if-up.d/xivo-mtu chmod 755 /etc/network/if-up.d/xivo-mtu
Insert the following content:
#!/bin/sh # Set MTU per iface if [ "${IFACE}" = "<data interface>" ]; then ip link set ${IFACE} mtu <data mtu> elif [ "${IFACE}" = "<voip interface>" ]; then ip link set ${IFACE} mtu <voip mtu> fi
Change the <data interface> to the name of your interface (e.g. eth0), and the <data mtu> to the new MTU (e.g. 1492),
Change the <voip interface> to the name of your interface (e.g. eth1), and the <voip mtu> to the new MTU (e.g. 1488)
Note
In the above example you can set a different MTU per interface. If you don’t need a per-interface MTU you can simply write:
#!/bin/sh
ip link set ${IFACE} mtu <my mtu>
Backup¶
A backup of the database and the data are launched every day with a logrotate task. It is run at 06:25 a.m. and backups are kept for 7 days.
Logrotate task:
/etc/logrotate.d/xivo-backup
Logrotate cron:
/etc/cron.daily/logrotate
You can retrieve the backup from the web-interface in
page.Otherwise, with shell access, you can retrieve them in /var/backups/xivo
.
In this directory you will find db.tgz
and data.tgz
files for the database and data
backups.
Backup scripts:
/usr/sbin/xivo-backup
Backup location:
/var/backups/xivo
Here is the list of folders and files that are backed-up:
/etc/asterisk/
/etc/consul/
/etc/crontab
/etc/dahdi/
/etc/dhcp/
/etc/hostname
/etc/hosts
/etc/ldap/
/etc/network/if-up.d/xivo-routes
/etc/network/interfaces
/etc/ntp.conf
/etc/profile.d/xivo_uuid.sh
/etc/resolv.conf
/etc/ssl/
/etc/systemd/
/etc/wanpipe/
/etc/xivo-agentd/
/etc/xivo-agid/
/etc/xivo-amid/
/etc/xivo-auth/
/etc/xivo-call-logd/
/etc/xivo-confd/
/etc/xivo-confgend-client/
/etc/xivo-ctid/
/etc/xivo-ctid-ng/
/etc/xivo-dird/
/etc/xivo-dird-phoned/
/etc/xivo-dxtora/
/etc/xivo-purge-db/
/etc/xivo-websocketd/
/etc/xivo/
/usr/local/bin/
/usr/local/sbin/
/usr/share/xivo/XIVO-VERSION
/var/lib/asterisk/
/var/lib/consul/
/var/lib/xivo-provd/
/var/lib/xivo/
/var/log/asterisk/
/var/spool/asterisk/
/var/spool/cron/crontabs/
The following files/folders are excluded from this backup:
- folders:
/var/lib/consul/checks
/var/lib/consul/raft
/var/lib/consul/serf
/var/lib/consul/services
/var/lib/xivo-provd/plugins/*/var/cache/*
/var/spool/asterisk/monitor/
/var/spool/asterisk/meetme/
- files
/var/lib/xivo-provd/plugins/xivo-polycom*/var/tftpboot/*.ld
- log files, coredump files
- audio recordings
- and, files greater than 10 MiB or folders containing more than 100 files if they belong to one of
these folders:
/var/lib/xivo/sounds/
/var/lib/asterisk/sounds/custom/
/var/lib/asterisk/moh/
/var/spool/asterisk/voicemail/
/var/spool/asterisk/monitor/
The database asterisk
from PostgreSQL is backed up. This include almost everything that is
configured via the web interface.
Warning
A backup file may take a lot of space on the disk. You should check the free space on the partition before creating one.
You can manually create a database backup file named db-manual.tgz
in /var/tmp
by
issuing the following commands:
xivo-backup db /var/tmp/db-manual
You can manually create a data backup file named data-manual.tgz
in /var/tmp
by
issuing the following commands:
xivo-backup data /var/tmp/data-manual
Restore¶
A backup of both the configuration files and the database used by a XiVO installation is done
automatically every day.
These backups are created in the /var/backups/xivo
directory and are kept for 7 days.
- You must restore a backup on the same version of XiVO that was backed up (though the
architecture –
i386
oramd64
– may differ) - You must restore a backup on a machine with the same hostname and IP address
- Be aware that this procedure applies only to XiVO >= 14.08 (see 14.08).
Warning
Before restoring a XiVO on a fresh install you have to setup XiVO using the wizard (see Running the Wizard section).
Stop monit and all the xivo services:
xivo-service stop
System files are stored in the data.tgz file located in the /var/backups/xivo
directory.
This file contains for example, voicemail files, musics, voice guides, phone sets firmwares, provisioning server configuration database.
To restore the file
tar xvfp /var/backups/xivo/data.tgz -C /
Warning
- This will destroy all the current data in your database.
- You have to check the free space on your system partition before extracting the backups.
Database backups are created as db.tgz
files in the /var/backups/xivo
directory.
These tarballs contains a dump of the database used in XiVO.
In this example, we’ll restore the database from a backup file named db.tgz
placed in the home directory of root.
First, extract the content of the db.tgz
file into the /var/tmp
directory and go inside
the newly created directory:
tar xvf db.tgz -C /var/tmp
cd /var/tmp/pg-backup
Drop the asterisk database and restore it with the one from the backup:
sudo -u postgres dropdb asterisk
sudo -u postgres pg_restore -C -d postgres asterisk-*.dump
To finalize the restore, see After Restoring The System.
When restoring the database, if you encounter problems related to the system locale, see PostgreSQL localization errors.
System configuration like network interfaces is stored in the database. It is possible to keep this configuration and only restore xivo data.
Rename the asterisk database to asterisk_previous:
sudo -u postgres psql -c 'ALTER DATABASE asterisk RENAME TO asterisk_previous'
Restore the asterisk database from the backup:
sudo -u postgres pg_restore -C -d postgres asterisk-*.dump
Restore the system configuration tables from the asterisk_previous database:
sudo -u postgres pg_dump -c -t dhcp -t netiface -t resolvconf asterisk_previous | sudo -u postgres psql asterisk
Drop the asterisk_previous database:
sudo -u postgres dropdb asterisk_previous
Warning
Restoring the data.tgz file also restores system files such as host hostname, network interfaces, etc. You will need to reapply the network configuration if you restore the data.tgz file.
Resynchronize the xivo-auth keys:
xivo-update-keys
Update systemd runtime configuration:
source /etc/profile.d/xivo_uuid.sh
systemctl set-environment XIVO_UUID=$XIVO_UUID
systemctl daemon-reload
Restart the services you stopped in the first step:
xivo-service start
You may also reboot the system.
HTTPS certificate¶
X.509 certificates are used to authorize and secure communications with the server. They are mainly used for HTTPS, but can also be used for SIPS, CTIS, WSS, etc.
There are two categories of certificates in XiVO:
- the default certificate, used for HTTPS in the web interface, REST APIs and WebSockets
- the certificates created and managed via the web interface
This article is about the former. For the latter, see Telephony certificates.
XiVO uses HTTPS where possible. The certificates are generated at install time (or
during the upgrade to 15.12+). The main certificate is placed in
/usr/share/xivo-certs/server.crt
.
However, this certificate is self-signed, and HTTP clients (browser or REST API client) will complain about this default certificate because it is not signed by a trusted Certification Authority (CA).
To make the HTTP client accept this certificate, you have two choices:
- configure your HTTP client to trust the self-signed XiVO certificate by adding a new trusted CA.
The CA certificate (or bundle) is the file
/usr/share/xivo-certs/server.crt
. - replace the self-signed certificate with your own trusted certificate.
For this, follow these steps:
- Replace the following files with your own private key/certificate pair:
- Private key:
/usr/share/xivo-certs/server.key
- Certificate:
/usr/share/xivo-certs/server.crt
Change the hostname of XiVO for each XiVO component: the different processes of XiVO heavily use HTTPS for internal communication, and for these connection to establish successfully, all hostnames used must match the Common Name (CN) of your certificate. Basically, you must replace all occurrences of
localhost
(the default hostname) with your CN in the configuration of the XiVO services. For example:mkdir /etc/xivo/custom cat > /etc/xivo/custom/custom-certificate.yml << EOF consul: host: xivo.example.com auth: host: xivo.example.com confd: host: xivo.example.com dird: host: xivo.example.com ajam: host: xivo.example.com agentd: host: xivo.example.com EOF for config_dir in /etc/xivo-*/conf.d/ ; do ln -s "/etc/xivo/custom/custom-certificate.yml" "$config_dir/010-custom-certificate.yml" done
Also, you must replace
localhost
in the definition of your directories in the web interface under .If your certificate is not self-signed, and you obtained it from a third-party CA that is trusted by your system, you must enable the system-based certificate verification. By default, certificate verification is set to consider
/usr/share/xivo-certs/server.crt
as the only CA certificate.The options are the following:
- Consul:
verify: True
- Other XiVO services:
verify_certificate: True
The procedure is the same as 2. with more configuration for each service. For example:
cat > /etc/xivo/custom/custom-certificate.yml << EOF consul: host: xivo.example.com verify: True auth: host: xivo.example.com verify_certificate: True dird: host: xivo.example.com verify_certificate: True ...
Setting
verify_certificate
toFalse
will disable the certificate verification, but the connection will still be encrypted. This is pretty safe as long as XiVO services stay on the same machine, however, this is dangerous when XiVO services are separated by an untrusted network, such as the Internet.- Consul:
Ensure your CN resolves to a valid IP address with either:
- a DNS entry
- an entry in
/etc/hosts
resolving your CN to 127.0.0.1. Note that/etc/hosts
will be rewritten occasionally by xivo-sysconfd. To make the change persistent, you can:- modify
/usr/share/xivo-sysconfd/templates/resolvconf/hosts
instead (which will be rewritten when xivo-sysconfd is upgraded...) - then add a script in
/usr/share/xivo-upgrade/pre-start.d
to re-apply the modification to/usr/share/xivo-sysconfd/templates/resolvconf/hosts
after eachxivo-upgrade
.
- modify
Restart all XiVO services:
xivo-service restart all
Configuration Files¶
This section describes some of the XiVO configuration files.
Usually, the configuration is read from two locations: a configuration file config.yml
and a
configuration directory conf.d
.
Files in the conf.d
extra configuration directory:
- are used in alphabetical order and the first one has priority
- are ignored when their name starts with a dot
- are ignored when their name does not end with
.yml
For example:
.01-critical.yml
:
log_level: critical
02-error.yml.dpkg-old
:
log_level: error
10-debug.yml
:
log_level: debug
20-nodebug.yml
:
log_level: info
The value that will be used for log_level
will be debug
since:
10-debug.yml
comes before20-nodebug.yml
in the alphabetical order..01-critical.yml
starts with a dot so is ignored02-error.yml.dpkg-old
does not end with.yml
so is ignored
Configuration files for every service running on a XiVO server will respect these rules:
- Default configuration directory in
/etc/xivo-{service}/conf.d
(e.g./etc/xivo-agentd/conf.d/
) - Default configuration file in
/etc/xivo-{service}/config.yml
(e.g./etc/xivo-agentd/config.yml
)
The files /etc/xivo-{service}/config.yml
should not be modified because they will be
overridden during upgrades. However, they may be used as examples for creating additional
configuration files as long as they respect the Configuration priority. Any exceptions to
these rules are documented below.
- Default configuration directory:
/etc/xivo-agentd/conf.d
- Default configuration file:
/etc/xivo-agentd/config.yml
- Default configuration directory:
/etc/xivo-amid/conf.d
- Default configuration file:
/etc/xivo-amid/config.yml
- Default configuration directory:
/etc/xivo-auth/conf.d
- Default configuration file:
/etc/xivo-auth/config.yml
- Default configuration directory:
/etc/xivo-confgend/conf.d
- Default configuration file:
/etc/xivo-confgend/config.yml
- Default templates directory:
/etc/xivo-confgend/templates
- Default configuration directory:
/etc/xivo-ctid/conf.d
- Default configuration file:
/etc/xivo-ctid/config.yml
- Default configuration directory:
/etc/xivo-dao/conf.d
- Default configuration file:
/etc/xivo-dao/config.yml
This configuration is read by many XiVO programs in order to connect to the Postgres database of XiVO.
- Default configuration directory:
/etc/xivo-dird-phoned/conf.d
- Default configuration file:
/etc/xivo-dird-phoned/config.yml
- Default configuration directory:
/etc/xivo-websocketd/conf.d
- Default configuration file:
/etc/xivo-websocketd/config.yml
- Path:
/etc/xivo/asterisk/xivo_ring.conf
- Purpose: This file can be used to change the ringtone played by the phone depending on the origin of the call.
Warning
Note that this feature has not been tested for all phones and all call flows. This page describes how you can customize this file but does not intend to list all validated call flows or phones.
This file xivo_ring.conf
consists of :
- profiles of configuration (some examples for different brands are already included:
[aastra]
,[snom]
etc.) - one section named
[number]
where you apply the profile to an extension or a context etc.
Here is the process you should follow if you want to use/customize this feature :
Create a new profile, e.g.:
[myprofile-aastra]
Change the
phonetype
accordingly, in our example:[myprofile-aastra] phonetype = aastra
Chose the ringtone for the different type of calls (note that the ringtone names are brand-specific):
[myprofile-aastra] phonetype = aastra intern = <Bellcore-dr1> group = <Bellcore-dr2>
Apply your profile, in the section
[number]
to a given list of extensions (e.g. 1001 and 1002):
1001@default = myprofile-aastra 1002@default = myprofile-aastraor to a whole context (e.g. default):
@default = myprofile-aastra
Restart
xivo-agid
service:service xivo-agid restart
- Path:
/etc/xivo/web-interface/ipbx.ini
- Purpose: This file specifies various configuration options and paths related to Asterisk and used by the web interface.
Here is a partial glimpse of what can be configured in file ipbx.ini
:
Enable/Disable modification of SIP line username and password:
[user] readonly-idpwd = "true"
When editing a SIP line, the username and password fields cannot be modified via the web interface. Set this option to false to enable the modification of both fields. This option is set to “true” by default.
Warning
This feature is not fully tested. It should be used only when absolutely necessary and with great care.
Consul¶
The default consul installation in XiVO uses the
configuration file in /etc/consul/xivo/*.json
. All files in this directory
are installed with the package and should not be modified by the
administrator. To use a different configuration, the adminstrator can add it’s
own configuration file at another location and set the new configuration directory by creating a
systemd unit drop-in file in the /etc/systemd/system/consul.service.d
directory.
The default installation generates a master token that can be retrieved in
/var/lib/consul/master_token
. This master token will not be used if a new
configuration is supplied.
The following environment variables can be overridden in a systemd unit drop-in file:
CONFIG_DIR
: the consul configuration directoryWAIT_FOR_LEADER
: should the “start” action wait for a leader ?
Example, in /etc/systemd/system/consul.service.d/custom.conf
:
[Service]
Environment=CONFIG_DIR=/etc/consul/agent
Environment=WAIT_FOR_LEADER=no
It is possible to run consul on another host and have the local consul node run as an agent only.
To get this kind of setup up and running, you will need to follow the following steps.
For a 32 bits system
wget --no-check-certificate https://releases.hashicorp.com/consul/0.5.2/consul_0.5.2_linux_386.zip
For a 64 bits system
wget --no-check-certificate https://releases.hashicorp.com/consul/0.5.2/consul_0.5.2_linux_amd64.zip
unzip consul_0.5.2_linux_386.zip
Or
unzip consul_0.5.2_linux_amd64.zip
mv consul /usr/bin/consul
mkdir -p /etc/consul/xivo
mkdir -p /var/lib/consul
adduser --quiet --system --group --no-create-home \
--home /var/lib/consul consul
On the new consul host, modify /etc/consul/xivo/config.json
to include to following lines.
"bind_addr": "0.0.0.0",
"client_addr": "0.0.0.0",
"advertise_addr": "<consul-host>"
# on the consul host
scp root@<xivo-host>:/lib/systemd/system/consul.service /lib/systemd/system
systemctl daemon-reload
scp -r root@<xivo-host>:/etc/consul /etc
scp -r root@<xivo-host>:/usr/share/xivo-certs /usr/share
consul agent -data-dir /var/lib/consul -config-dir /etc/consul/xivo/
Note
To start consul with the systemd unit file, you may need to change owner and group
(consul:consul) for all files inside /etc/consul
, /usr/share/xivo-certs
and /var/lib/consul
Create the file /etc/consul/agent/config.json
with the following content
{
"acl_datacenter": "<node_name>",
"datacenter": "xivo",
"server": false,
"bind_addr": "0.0.0.0",
"advertise_addr": "<xivo_address>",
"client_addr": "127.0.0.1",
"bootstrap": false,
"rejoin_after_leave": true,
"data_dir": "/var/lib/consul",
"enable_syslog": true,
"disable_update_check": true,
"log_level": "INFO",
"ports": {
"dns": -1,
"http": -1,
"https": 8500
},
"retry_join": [
"<remote_host>"
],
"cert_file": "/usr/share/xivo-certs/server.crt",
"key_file": "/usr/share/xivo-certs/server.key"
}
node_name
: Arbitrary name to give this node,xivo-paris
for example.remote_host
: IP address of your new consul. Be sure the host is accessible from your XiVO and check the firewall. See the documentation here.xivo_address
: IP address of your xivo.
This file should be owned by consul user.
chown -R consul:consul /etc/consul/agent
Add or modify /etc/systemd/system/consul.service.d/custom.conf
to include the following lines:
[Service]
Environment=CONFIG_DIR=/etc/consul/agent
Restart your consul server.
service consul restart
Add a file in /etc/xivo-ctid/conf.d/remote_consul.yml
with the following content
rest_api:
http:
listen: 0.0.0.0
service_discovery:
advertise_address: <xivo-ctid-host>
check_url: http://<xivo-ctid-host>:9495/0.1/infos
xivo-ctid-host
: Hostname to reach xivo-ctid
Log Files¶
Every XiVO service has its own log file, placed in /var/log
.
The Asterisk log files are managed by logrotate.
It’s configuration files /etc/logrotate.d/asterisk
and /etc/asterisk/logger.conf
The message log level is enabled by default in logger.conf
and contains notices, warnings
and errors. The full log entry is commented in logger.conf
and should only be enabled when
verbose debugging is required. Using this option in production would produce VERY large log files.
- Files location:
/var/log/asterisk/*
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-agentd.log
- Rotate configuration:
/etc/logrotate.d/xivo-agentd
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-agid.log
- Rotate configuration:
/etc/logrotate.d/xivo-agid
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-amid.log
- Rotate configuration:
/etc/logrotate.d/xivo-amid
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-auth.log
- Rotate configuration:
/etc/logrotate.d/xivo-auth
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-call-logd.log
- Rotate configuration:
/etc/logrotate.d/xivo-call-logd
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-confd.log
- Rotate configuration:
/etc/logrotate.d/xivo-confd
- Number of archived files: 15
- Rotation frequence: Daily
The xivo-confgend daemon output is sent to the file specified with the --logfile
parameter when
launched with twistd.
The file location can be changed by customizing the xivo-confgend.service unit file.
- File location:
/var/log/xivo-confgend.log
- Rotate configuration:
/etc/logrotate.d/xivo-confgend
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-ctid.log
- Rotate configuration:
/etc/logrotate.d/xivo-ctid
- Number of archived log files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-ctid-ng.log
- Rotate configuration:
/etc/logrotate.d/xivo-ctid-ng
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-dird.log
- Rotate configuration:
/etc/logrotate.d/xivo-dird
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-dird-phoned.log
- Rotate configuration:
/etc/logrotate.d/xivo-dird-phoned
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-dxtora.log
- Rotate configuration:
/etc/logrotate.d/xivo-dxtora
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-provd.log
- Rotate configuration:
/etc/logrotate.d/xivo-provd
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-purge-db.log
- Rotate configuration:
/etc/logrotate.d/xivo-purge-db
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-stat.log
- Rotate configuration:
/etc/logrotate.d/xivo-stat
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-sysconfd.log
- Rotate configuration:
/etc/logrotate.d/xivo-sysconfd
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-upgrade.log
- Rotate configuration:
/etc/logrotate.d/xivo-upgrade
- Number of archived files: 15
- Rotation frequence: Daily
- File location:
/var/log/xivo-web-interface/*.log
- Rotate configuration:
/etc/logrotate.d/xivo-web-interface
- Number of archived files: 21
- Rotation frequence: Daily
- File location:
/var/log/xivo-websocketd.log
- Rotate configuration:
/etc/logrotate.d/xivo-websocketd
- Number of archived files: 15
- Rotation frequence: Daily
Nginx¶
XiVO use nginx as a web server and reverse proxy.
In its default configuration, the nginx server listens on port TCP/80 and TCP/443 and allows these services to be used:
- web interface (xivo-web-interface)
- API documentation (xivo-swagger-doc)
Starting from XiVO 16.13, an administrator can easily modify the configuration to allow additional services to be used (e.g. xivo-auth or xivo-confd).
To do so, an administrator only has to create a symbolic link inside the
/etc/nginx/locations/http-enabled
directory to the corresponding file in the
/etc/nginx/locations/http-available
directory, and then reload nginx with
systemctl reload nginx
. A similar operation must be done for HTTPS.
For example, to enable all the available services:
ln -sf /etc/nginx/locations/http-available/* /etc/nginx/locations/http-enabled
ln -sf /etc/nginx/locations/https-available/* /etc/nginx/locations/https-enabled
systemctl reload nginx
To disable all the services other than the web interface:
rm /etc/nginx/locations/http-enabled/* /etc/nginx/locations/https-enabled/*
ln -s /etc/nginx/locations/http-available/xivo-web-interface /etc/nginx/locations/http-enabled
ln -s /etc/nginx/locations/https-available/xivo-web-interface /etc/nginx/locations/https-enabled
systemctl reload nginx
NTP¶
XiVO has a NTP server, that must be synchronized to a reference server. This can be a public one or customized for specific target networking architecture. XiVO’s NTP server is used by default as NTP server for the devices time reference.
Show NTP service status:
service ntp status
Stop NTP service:
service ntp stop
Start NTP service:
service ntp start
Restart NTP service:
service ntp restart
Show NTP synchronization status:
ntpq -p
Edit
/etc/ntp.conf
Give your NTP reference servers:
server 192.168.0.1 # LAN existing NTP Server server 0.debian.pool.ntp.org iburst dynamic # default in ntp.conf server 1.debian.pool.ntp.org iburst dynamic # default in ntp.conf
If no reference server to synchronize to, add this to synchronize locally:
server 127.127.1.0 # local clock (LCL) fudge 127.127.1.0 stratum 10 # LCL is not very reliable
Restart NTP service
Check NTP synchronization status.
Warning
If #5 shows that NTP doesn’t use NTP configuration in /etc/ntp.conf
, maybe have
you done a dhclient
for one of your network interface and the dhcp server that gave the IP
address also gave a NTP server address. Thus you might check if the file /var/lib/ntp/ntp.conf.dhcp
exists, if yes, this is used for NTP configuration prior to /etc/ntp.conf
. Remove it and
restart NTP, check NTP synchronization status, then it should work.
Proxy Configuration¶
If you use XiVO behind an HTTP proxy, you must do a couple of manipulations for it to work correctly.
Create the /etc/apt/apt.conf.d/90proxy
file with the following content:
Acquire::http::Proxy "http://domain\username:password@proxyip:proxyport";
Proxy information is set via the
page.This step is needed if you use the DHCP server of the XiVO. Otherwise the DHCP configuration won’t be correct.
Proxy information is set via the /etc/xivo/dhcpd-update.conf
file.
Edit the file and look for the [proxy]
section.
This step is not needed if you don’t use xivo-fetchfw.
Proxy information is set via the /etc/xivo/xivo-fetchfw.conf
file.
Edit the file and look for the [proxy]
section.
Service Discovery¶
XiVO uses consul for service discovery. When a daemon is started, it registers itself on the configured consul node.
Consul template may be used to generate the configuration files for each daemons that requires the availability of another service. Consul template can also be used to reload the appropriate service.
Service Authentication¶
XiVO services expose more and more resources through REST API, but they also ensure that the access is restricted to the authorized programs. For this, we use an authentication daemon who delivers authorizations via tokens.
Here is the call flow to access a REST resource of a XiVO service:
- Create a username/password (also called service_id/service_key) with the right ACLs, via Web Services Access.
- Create a token with these credentials and the backend xivo-service.
- Use this token to access the REST resource defined by the ACL.

Call flow of service authentication
- Service
- Service who needs to access a REST resource.
- xivo-{daemon}
- Server that exposes a REST resource. This resource must have an attached ACL.
- xivo-auth
- Server that authenticates the Service and validates the required ACL with the token.
XiVO services directly use this system to communicate with each other, as you can see in their Web Services Access.
xivo-auth¶
xivo-auth is a scalable, extendable and configurable authentication service. It uses an HTTP interface to emit tokens to users who can then use those tokens to identify and authenticate themselves with other services compatible with xivo-auth.
The HTTP API reference is at http://api.xivo.io.
- POST
/0.1/token
, fieldexpiration
: only integers are accepted, floats are now invalid. - Experimental backend
ldap_user_voicemail
has been removed. - New backend
ldap_user
has been added.
- POST
/0.1/token
do not accept anymore argumentbackend_args
- New backend
ldap_user_voicemail
has been added. WARNING this backend is EXPERIMENTAL.
- HEAD and GET now take a new
scope
query string argument to check ACLs - Backend interface method
get_acls
is now namedget_consul_acls
- Backend interface method
get_acls
now returns a list of ACLs - HEAD and GET can now return a
403
if an ACL access is denied
- POST
/0.1/token
accept new argumentbackend_args
- Signature of backend method
get_ids()
has a new argumentargs
- New method
get_acls
for backend has been added - New backend
service
has been added
xivo-auth contains 4 major components, an HTTP interface, a celery worker, authentication backends and a consul client. All operations are made through the HTTP interface, tokens are stored by consul as well as the persistence for some of the data attached to tokens. The celery worker is used to schedule tasks that outlive the lifetime of the xivo-auth process. Backends are used to test if a supplied username/password combination is valid and provide the xivo-user-uuid.
xivo-auth is made of the following modules and packages.
the plugin package contains the xivo-auth backends that are packaged with xivo-auth.
The http module is the implementation of the HTTP interface.
- Validate parameters
- Calls the backend the check the user authentication
- Forward instructions to the token_manager
- Handle exceptions and return the appropriate status_code
The controller is the plumbin of xivo-auth, it has no business logic.
- Start the HTTP application
- Start the celery worker
- Load all enabled plugins
- Instanciate the token_manager
The token modules contains the business logic of xivo-auth.
- Creates and delete tokens
- Creates ACLs for XiVO
- Schedule token expiration
- Read/write token data to consul
The tasks module contains implementation of celery tasks that are executed by the worker.
- Called by the celery worker
- Forwards instructions to the token manager
This is a place holder for a global variable for the celery app. It will be removed and should not be used.
Other modules that should not need documentation are helpers, config, interfaces
xivo-auth is meant to be easy to extend. This section describes how to add features to xivo-auth.
xivo-auth allows its administrator to configure one or many sources of authentication. Implementing a new kind of authentication is quite simple.
- Create a python module implementing the backend interface.
- Install the python module with an entry point xivo_auth.backends
An example backend implementation is available here.
Backend name: xivo_admin
Purpose: Authenticate a XiVO administrator. The login/password is configured in
.Backend name: xivo_service
Purpose: Authenticate a XiVO Web Services Access. The login/password is configured in .
Backend name: xivo_user
Purpose: Authenticate a XiVO user. The login/password is configured in
in the CTI client section.Backend name: ldap_user
Purpose: Authenticate with an LDAP user.
For example, with the given configuration:
ldap:
uri: ldap://example.org
bind_dn: cn=xivo,dc=example,dc=org
bind_password: bindpass
user_base_dn: ou=people,dc=example,dc=org
user_login_attribute: uid
user_email_attribute: mail
When an authentication request is received for username alice
and password userpass
, the
backend will:
- Connect to the LDAP server at example.org
- Do an LDAP “bind” operation with bind DN
cn=xivo,dc=example,dc=org
and passwordbindpass
- Do an LDAP “search” operation to find an LDAP user matching
alice
, using:- the base DN
ou=people,dc=example,dc=org
- the filter
(uid=alice)
- a SUBTREE scope
- the base DN
- If the search returns exactly 1 LDAP user, do an LDAP “bind” operation with the user’s DN and the
password
userpass
- If the LDAP “bind” operation is successful, search in XiVO a user with an email matching the
mail
attribute of the LDAP user - If a XiVO user is found, success
To use an anonymous bind instead, the following configuration would be used:
ldap:
uri: ldap://example.org
bind_anonymous: True
user_base_dn: ou=people,dc=example,dc=org
user_login_attribute: uid
user_email_attribute: mail
The backend can also works in a “no search” mode, for example with the following configuration:
ldap:
uri: ldap://example.org
user_base_dn: ou=people,dc=example,dc=org
user_login_attribute: uid
user_email_attribute: mail
When the server receives the same authentication request as above, it will directly do an
LDAP “bind” operation with the DN uid=alice,ou=people,dc=example,dc=org
and password
userpass
, and continue at step 5.
Note
User’s email and voicemail’s email are two separate things. This plugin only use the user’s email.
uri
- the URI of the LDAP server. Can only contain the scheme, host and port of an LDAP URL.
user_base_dn
- the base dn of the user
user_login_attribute
- the attribute to login a user
user_email_attribute
(optional)- the attribute to match with the XiVO user’s email (default: mail)
bind_dn
(optional)- the bind DN for searching for the user DN.
bind_password
(optional)- the bind password for searching for the user DN.
bind_anonymous
(optional)- use anonymous bind for searching for the user DN (default: false)
xivo-auth is used through HTTP requests, using HTTPS. Its default port is 9497. As a user, the most common operation is to get a new token. This is done with the POST method.
Alice retrieves a token using her username/password:
$ # Alice creates a new token, using the xivo_user backend, expiring in 10 minutes
$ curl -k -X POST -H 'Content-Type: application/json' -u 'alice:s3cre7' "https://localhost:9497/0.1/token" -d '{"backend": "xivo_user", "expiration": 600}';echo
{"data": {"issued_at": "2015-06-05T10:16:58.557553", "token": "1823c1ee-6c6a-0cdc-d869-964a7f08a744", "auth_id": "63f3dc3c-865d-419e-bec2-e18c4b118224", "xivo_user_uuid": "63f3dc3c-865d-419e-bec2-e18c4b118224", "expires_at": "2015-06-05T11:16:58.557595"}}
In this example Alice used here XiVO CTI client login alice
and password s3cre7
. The
authentication source is determined by the backend in the POST data.
Alice could also have specified an expiration time on her POST request. The expiration value is the number of seconds before the token expires.
After retrieving her token, Alice can query other services that use xivo-auth and send her token to those service. Those services can then use this token on Alice’s behalf to access her personal storage.
If Alice wants to revoke her token before its expiration:
$ curl -k -X DELETE -H 'Content-Type: application/json' "https://localhost:9497/0.1/token/1823c1ee-6c6a-0cdc-d869-964a7f08a744"
See http://api.xivo.io for more details about the HTTP API.
See Service Authentication for details about the authentication process.
A service that requires authentication and identification can use xivo-auth to externalise the burden of authentication. The new service can then accept a token as part of its operations to authenticate the user using the service.
Once a service receives a token from one of its user, it will need to check the validity of that token. There are 2 forms of verification, one that only checks if the token is valid and the other returns information about this token’s session if it is valid.
Checking if a token is valid:
$ curl -k -i -X HEAD -H 'Content-Type: application/json' "https://localhost:9497/0.1/token/1823c1ee-6c6a-0cdc-d869-964a7f08a744"
HTTP/1.1 204 NO CONTENT
Content-Type: text/html; charset=utf-8
Content-Length: 0
Date: Fri, 05 Jun 2015 14:49:50 GMT
Server: pcm-dev-0
$ # get more information about this token
$ curl -k -X GET -H 'Content-Type: application/json' "https://localhost:9497/0.1/token/1823c1ee-6c6a-0cdc-d869-964a7f08a744";echo
{"data": {"issued_at": "2015-06-05T10:16:58.557553", "token": "1823c1ee-6c6a-0cdc-d869-964a7f08a744", "auth_id": "63f3dc3c-865d-419e-bec2-e18c4b118224", "xivo_user_uuid": "63f3dc3c-865d-419e-bec2-e18c4b118224", "expires_at": "2015-06-05T11:16:58.557595"}}
usage: xivo-auth [-h] [-c CONFIG_FILE] [-u USER] [-d] [-f] [-l LOG_LEVEL]
optional arguments:
-h, --help show this help message and exit
-c CONFIG_FILE, --config-file CONFIG_FILE
The path to the config file
-u USER, --user USER User to run the daemon
-d, --debug Log debug messages
-f, --foreground Foreground, don't daemonize
-l LOG_LEVEL, --log-level LOG_LEVEL
Logs messages with LOG_LEVEL details. Must be one of:
critical, error, warning, info, debug. Default: None
The complete HTTP API documentation is at http://api.xivo.io.
See also the xivo-auth HTTP API Changelog.
xivo-confd¶
xivo-confd is a HTTP server that provides a RESTful API service for configuring and managing basic resources on a XiVO server.
The HTTP API reference is available at http://api.xivo.io.
xivo-confd resources are organised through a plugin mechanism. There are 2 main plugin categories:
- Resource plugins
- A plugin that manages a resource (e.g. users, extensions, voicemails, etc). A resource plugin exposes the 4 basic CRUD operations (Create, Read, Update, Delete) in order to operate on a resource in a RESTful manner.
- Association plugins
- A plugin for associating or dissociating 2 resources (e.g a user and a line). An association
plugin exposes an HTTP action for associating (either
POST
orPUT
) and another for dissociating (DELETE
)
The following diagram outlines the most important parts of a plugin:

Plugin architecture of xivo-confd
- Resource
Class that receives and handles HTTP requests. Resources use flask-restful for handling requests.
There are 2 kinds of resources: ListResource for root URLs and ItemResource for URLs that have an ID. ListResource will handle creating a resource (
POST
) and searching through a list of available resources (GET
). ItemResource handles fetching a single item (GET
), updating (PUT
) and deleting (DELETE
).- Service
Class that handles business logic for a resource, such as what to do in order to get, create, update, or delete a resource. Service classes do not manipulate data directly. Instead, they coordinate what to do via other objects.
There are 2 kinds of services: CRUDService for basic CRUD operations in Resource plugins, and AssociationService for association/dissociation operations in Association plugins.
- Dao
- Data Access Object. Knows how to get data and how to manipulate it, such as SQL queries, files, etc.
- Notifier
- Sends events after an operation has completed. An event will be sent in a messaging queue for each CRUD operation. Certain resources also need to send events to other daemons in order to reload some configuration data. (i.e. asterisk needs to reload the dialplan when an extension is updated)
- Validator
- Makes sure that a resource’s data does not contain any errors before doing something with it. A Validator can be used for validating input data or business rules.
XiVO confgend¶
xivo-confgend is a configuration file generator. It is mainly used to generate the Asterisk configuration files.
xivo-confgend uses drivers to implement the logic required to generate configuration files. It uses stevedore to do the driver instantiation and discovery.
Plugins in xivo-confgend use setuptools’ entry points. That means that installing a new plugin to xivo-confgend requires an entry point in the plugin’s setup.py.
Driver plugin are classes that are used to generate the content of a configuration file.
The implementation of a plugin should have the following properties.
- It’s
__init__
method should take one argument - It should have a
generate
method which will return the content of the file - A setup.py adding an entry point
The __init__
method argument is the content of the configuration of
xivo-confgend. This allows the driver implementor to add values to the
configuration in /etc/xivo-confgend/conf.d/*.yml
and these values will be
available in the driver.
The generate method has no argument, the configuration provided to the
__init__
should be sufficient for most cases. generate
is called within a
scoped_session
of xivo-dao, allowing the usage of xivo-dao without prior setup
in the driver.
The namespaces used for entry points in xivo-confgend have the following form:
xivo_confgend.<resource>.<filename>
as an example, a generator for sip.conf would have the following namespace:
xivo_confgend.asterisk.sip.conf
Here is a typical setup.py
:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 | #!/usr/bin/env python
# -*- coding: utf-8 -*-
# Copyright 2016 by Avencall
# SPDX-License-Identifier: GPL-3.0+
from setuptools import setup
from setuptools import find_packages
setup(
name='XiVO confgend driversample',
version='0.0.1',
description='An example driver',
packages=find_packages(),
entry_points={
'xivo_confgend.asterisk.sip.conf': [
'my_driver = src.driver:MyDriver',
],
}
)
|
With the following package structure:
.
├── setup.py
└── src
└── driver.py
driver.py
:
1 2 3 4 5 6 7 8 9 10 11 12 | # -*- coding: utf-8 -*-
# Copyright 2016 by Avencall
# SPDX-License-Identifier: GPL-3.0+
class MyDriver(object):
def __init__(self, config):
self._config = config
def generate(self):
return 'Hello World!'
|
To enable this plugin, you need to:
Install the plugin with:
python setup.py install
Create a config file in
/etc/xivo-confgend/conf.d
:plugins: asterisk.sip.conf: my_driver
Restart xivo-confgend:
systemctl restart xivo-confgend
XiVO dird¶
xivo-dird is the directory server for XiVO. It offers a simple REST interface to query all directories that are configured. xivo-dird is extendable with plugins.
- Added phonebook imports
- POST
0.1/tenants/<tenant>/phonebooks/<phonebook_id>/contacts/import
- POST
- Added a new internal phonebook with a CRUD interface
- Added a new backend to do lookups in the new phonebook
- The ldap plugins ldap_network_timeout default value has been incremented from 0.1 to 0.3 seconds
- Added the
voicemail
type in Views configuration - Removed reverse endpoints in REST API:
- GET
/0.1/directories/reverse/<profile>/me
- GET
- Added reverse endpoints in REST API:
- GET
/0.1/directories/reverse/<profile>/<xivo_user_uuid>
- GET
/0.1/directories/reverse/<profile>/me
- GET
- Added directories endpoints in REST API:
- GET
/0.1/directories/input/<profile>/aastra
- GET
/0.1/directories/lookup/<profile>/aastra
- GET
/0.1/directories/input/<profile>/polycom
- GET
/0.1/directories/lookup/<profile>/polycom
- GET
/0.1/directories/input/<profile>/snom
- GET
/0.1/directories/lookup/<profile>/snom
- GET
/0.1/directories/lookup/<profile>/thomson
- GET
/0.1/directories/lookup/<profile>/yealink
- GET
- Added more cisco endpoints in REST API:
- GET
/0.1/directories/input/<profile>/cisco
- GET
- Endpoint
/0.1/directories/lookup/<profile>/cisco
accepts a newlimit
andoffset
query string arguments.
- Added cisco endpoints in REST API:
- GET
/0.1/directories/menu/<profile>/cisco
- GET
/0.1/directories/lookup/<profile>/cisco
- GET
- Added more personal contacts endpoints in REST API:
- GET
/0.1/personal/<contact_id>
- PUT
/0.1/personal/<contact_id>
- POST
/0.1/personal/import
- DELETE
/0.1/personal
- GET
- Endpoint
/0.1/personal
accepts a newformat
query string argument.
- Added personal contacts endpoints in REST API:
- GET
/0.1/directories/personal/<profile>
- GET
/0.1/personal
- POST
/0.1/personal
- DELETE
/0.1/personal/<contact_id>
- GET
- Signature of backend method
list()
has a new argumentargs
- Argument
args
for backend methodslist()
andsearch()
has a new keytoken_infos
- Argument
args
for backend methodload()
has a new keymain_config
- Methods
__call__()
andlookup()
of service pluginlookup
take a newtoken_infos
argument
- Added authentication on all REST API endpoints
- Service plugins receive the whole configuration, rather than only their own section
There are three sources of configuration for xivo-dird:
- the command line options
- the main configuration file
- the sources configuration directory
The command-line options have priority over the main configuration file options.
Default location: /etc/xivo-dird/config.yml
. Format: YAML
The default location may be overwritten by the command line options.
Here’s an example of the main configuration file:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 | debug: False
foreground: False
log_filename: /var/log/xivo-dird.log
log_level: info
pid_filename: /var/run/xivo-dird/xivo-dird.pid
source_config_dir: /etc/xivo-dird/sources.d
user: www-data
rest_api:
wsgi_socket: /var/run/xivo-dird/xivo-dird.sock
enabled_plugins:
backends:
- csv
- ldap
- phonebook
services:
- lookup
views:
- cisco_view
- default_json
views:
displays:
switchboard_display:
-
title: Firstname
default: Unknown
field: firstname
type: name
-
title: Lastname
default: Unknown
field: lastname
type: name
default_display:
-
title: Firstname
field: fn
type: name
-
title: Location
default: Canada
field: country
-
title: Number
field: number
type: number
displays_phone:
default:
name:
- display_name
number:
-
field:
- phone
-
field:
- phone_mobile
name_format: "{name} (Mobile)"
profile_to_display:
default: default_display
switchboard: switchboard_display
profile_to_display_phone:
default: default
services:
lookup:
default:
sources:
- my_csv
- ldap_quebec
timeout: 0.5
switchboard:
sources:
- my_csv
- xivo_phonebook
- ldap_quebec
timeout: 1
sources:
my_source:
name: my_source
type: ldap
ldap_option1: value
ldap_option2: value
...
|
- debug
- Enable log debug messages. Overrides
log_level
. Default:False
. - foreground
- Foreground, don’t daemonize. Default:
False
. - log_filename
- File to write logs to. Default:
/var/log/xivo-dird.log
. - log_level
- Logs messages with LOG_LEVEL details. Must be one of:
critical
,error
,warning
,info
,debug
. Default:info
. - pid_filename
- File used as lock to avoid multiple xivo-dird instances. Default:
/var/run/xivo-dird/xivo-dird.pid
. - source_config_dir
- The directory from which sources configuration are read. See
Sources Configuration. Default:
/etc/xivo-dird/sources.d
. - user
- The owner of the process. Default:
www-data
.
This sections controls which plugins are to be loaded at xivo-dird startup. All plugin types must have at least one plugin enabled, or xivo-dird will not start. For back-end plugins, sources using a back-end plugin that is not enabled will be ignored.
- displays
A dictionary describing the content of each display. The key is the display’s name, and the value are the display’s content.
The display content is a list of fields. Each field is a dictionary with the following keys:
- title: The label of the field
- default: The default value of the field
- type: An arbitrary identifier of the field. May be used by consumers to identify the field without matching the label. For meaningful values inside XiVO, see Integration of XiVO dird with the rest of XiVO.
- field: the key of the data from the source that will be used for this field.
The display may be used by a plugin view to configure which fields are to be presented to the consumer.
- displays_phone
A dictionary describing the content of phone-related displays. Like
displays
, the key is the display’s name and the value is the display’s content. These displays are used by phone-related view plugins, like thecisco_view
plugin.The display content contains 2 keys,
name
andnumber
.The value of the
name
key is a list of source result fields. For a given source result, the first field that will return a non-empty value will be used as the display name on the phone. For example, ifname
is configured with["display_name", "name"]
and you have a source result with fields{"display_name": "", "name": "Bob"}
, then “Bob” will be displayed on the phone.The value of the
number
key is a list of number item. Each item is composed of a dictionary containing at least afield
key, and optionally aname_format
key. For example, if you have the following number configuration:name: - display_name number: - field: - phone - field: - phone_mobile name_format: "{name} (Mobile)"
and you have a source result
{"display_name": "Bob", "phone": "101", "phone_mobile": "102"}
, then 2 results will be displayed on your phone:- “Bob”, with number “101”
- “Bob (Mobile)”, with number “102”
The
name_format
value is a python format string. There’s two substitution variables available,{name}
and{number}
.- profile_to_display
- A dictionary associating a profile to a display. It allows xivo-dird to use the right display when a consumer makes a query with a profile. The key is the profile name and the value is the display name.
- profile_to_display_phone:
- A dictionary associating a profile to a phone display. This is similar to
profile_to_display
, but only used by phone-related view plugins.
This section is a dictionary whose keys are the service plugin name and values are the configuration of that service. Hence the content of the value is dependent of the service plugin. See the documentation of the service plugin (Stock Plugins Documentation).
This section is a dictionary whose keys are the source name and values are the configuration for that source. See the Sources Configuration section for more details about source configuration.
There are two ways to configure sources:
- in the sources section of the main configuration
- in files of a directory, one file for each source:
- Default directory location
/etc/xivo-dird/sources.d
- Files format: YAML
- File names are ignored
- Each file listed in this directory will be read and used to create a data source for xivo-dird.
- Default directory location
Here is an example of a CSV source configuration in its own file:
1 2 3 4 5 6 7 8 9 10 | type: csv
name: my_contacts_in_a_csv_file
file: /usr/local/share/my_contacts.csv
unique_column: id
searched_columns:
- fn
- ln
format_columns:
name: "{fn} {ln}"
number: "{num}"
|
This is strictly equivalent in the main configuration file:
1 2 3 4 5 6 7 8 9 10 11 12 13 | sources:
my_contacts_in_a_csv_file:
type: csv
name: my_contacts_in_a_csv_file
file: /usr/local/share/my_contacts.csv
unique_column: id
searched_columns:
- fn
- ln
source_to_display_columns:
ln: lastname
fn: firstname
num: number
|
- type
- the type of the source. It must be the same than the name of one of the enabled back-end plugins.
- name
- is the name of this given configuration. The name is used to associate the source to profiles. The value is arbitrary, but it must be unique across all sources.
Warning
Changing the name of the source will make all favorites in that source disappear. There is currently no tool to help you migrate favorites between source names, so choose your source names carefully.
The other options are dependent on the source type (the back-end used). See the documentation of the back-end plugin (Stock Plugins Documentation). However, the following keys should be present in all source configurations:
- first_matched_columns (optional)
- the columns used for the reverse lookup. Any column having the search term will be a reverse lookup result.
- format_columns (optional)
- a mapping between result fields and a format string. The new key will be added to the result, if this name already exists in the result, it will be replaced with the new value. The syntax is a python format string. See https://docs.python.org/2/library/string.html#formatspec for a complete reference.
- searched_columns (optional)
- the columns used for the lookup. Any column containing the search term substring will be a lookup result.
- unique_column (optional)
- This column is what makes an entry unique in this source. The
unique_column
is used to build theuid
that is passed to the list method to fetch a list of results by unique ids. This is necessary for listing and identifying favorites.

xivo-dird startup flow
The XiVO dird architecture uses plugins as extension points for most of its job. It uses stevedore to do the plugin instantiation and discovery and ABC classes to define the required interface.
Plugins in xivo-dird use setuptools’ entry points. That means that installing a new plugin to xivo-dird requires an entry point in the plugin’s setup.py. Each entry point’s namespace is documented in the appropriate documentation section. These entry points allow xivo-dird to be able to discover and load extensions packaged with xivo-dird or installed separately.
Each kind of plugin does a specific job. There are three kinds of plugins in dird.

xivo-dird HTTP query
All plugins are instantiated by the core. The core then keeps a catalogue of loaded extensions that can be supplied to other extensions.
The following setup.py shows an example of a python library that add a plugin of each kind to xivo-dird:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 | #!/usr/bin/env python
# -*- coding: utf-8 -*-
from setuptools import setup
from setuptools import find_packages
setup(
name='XiVO dird plugin sample',
version='0.0.1',
description='An example program',
packages=find_packages(),
entry_points={
'xivo_dird.services': [
'my_service = dummy:DummyServicePlugin',
],
'xivo_dird.backends': [
'my_backend = dummy:DummyBackend',
],
'xivo_dird.views': [
'my_view = dummy:DummyView',
],
}
)
|
Back-ends are used to query directories. Each back-end implements a way to query a given directory. Each instance of a given back-end is called a source. Sources are used by the services to get results from each configured directory.
Given one LDAP back-end, I can configure a source from the LDAP at alpha.example.com and another source from the other LDAP at beta.example.com. Both of these sources use the LDAP back-end.
- Namespace:
xivo_dird.backends
- Abstract source plugin: BaseSourcePlugin
- Methods:
name
: the name of the source, typically retrieved from the configuration injected toload()
load(args)
: set up resources used by the plugin, depending on the config.args
is a dictionary containing:- key
config
: the source configuration for this instance of the back-end - key
main_config
: the whole configuration of xivo-dird
- key
unload()
: free resources used by the plugin.search(term, args)
: The search method returns a list of dictionary.- Empty values should be
None
, instead of empty string. args
is a dictionary containing:- key
token_infos
: data associated to the authentication token (see xivo-auth)
- key
- Empty values should be
first_match(term, args)
: The first_match method returns a dictionary.- Empty values should be
None
, instead of empty string. args
is a dictionary containing:- key
token_infos
: data associated to the authentication token (see xivo-auth)
- key
- Empty values should be
list(uids, args)
: The list method returns a list of dictionary from a list of uids. Each uid is a string identifying a contact within the source.args
is a dictionary containing:- key
token_infos
: data associated to the authentication token (see xivo-auth)
- key
See Sources Configuration. The implementation of the back-end should take these values into account and return results accordingly.
The following example add a backend that will return random names and number.
dummy.py
:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 | # -*- coding: utf-8 -*-
import logging
logger = logging.getLogger(__name__)
class DummyBackendPlugin(object):
def name(self):
return 'my_local_dummy'
def load(self, args):
logger.info('dummy backend loaded')
def unload(self):
logger.info('dummy backend unloaded')
def search(self, term, args):
nb_results = random.randint(1, 20)
return _random_list(nb_results)
def list(self, unique_ids):
return _random_list(len(unique_ids))
def _random_list(self, nb_results):
columns = ['Firstname', 'Lastname', 'Number']
return [_random_entry(columns) for _ in xrange(nb_results)]
def _random_entry(self, columns):
random_stuff = [_random_string() for _ in xrange(len(columns))]
return dict(zip(columns, random_stuff))
def _random_string(self):
return ''.join(random.choice(string.lowercase) for _ in xrange(5))
|
Service plugins add new functionality to the dird server. These functionalities are available to views. When loaded, a service plugin receives its configuration and a dictionary of available sources.
Some service examples that come to mind include:
- A lookup service to search through all configured sources.
- A reverse lookup service to search through all configured sources and return a specific field of the first matching result.
Namespace:
xivo_dird.services
Abstract service plugin: BaseServicePlugin
Methods:
load(args)
: set up resources used by the plugin, depending on the config.args
is a dictionary containing:- key
config
: the whole configuration file in dict form - key
sources
: a dictionary of source names to sources
load
must return the service object, which is any kind of python object.- key
unload()
: free resources used by the plugin.
The following example adds a service that will return an empty list when used.
dummy.py
:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 | # -*- coding: utf-8 -*-
import logging
from xivo_dird import BaseServicePlugin
logger = logging.getLogger(__name__)
class DummyServicePlugin(BaseServicePlugin):
"""
This plugin is responsible fow instantiating and returning the
DummyService. It manages its life time and should take care of
its cleanup if necessary
"""
def load(self, args):
"""
Ignores all provided arguments and instantiate a DummyService that
is returned to the core
"""
logger.info('dummy loaded')
self._service = DummyService()
return self._service
def unload(self):
logger.info('dummy unloaded')
class DummyService(object):
"""
A very dumb service that will return an empty list every time it is used
"""
def list(self):
"""
This function must be called explicitly from the view, `list` is not a
special method name for xivo-dird
"""
return []
|
View plugins add new routes to the HTTP application in xivo-dird, in particular the REST API of xivo-dird: they define the URLs to which xivo-dird will respond and the formatting of data received and sent through those URLs.
For example, we can define a REST API formatted in JSON with one view and the same API formatted in XML with another view. Supporting the directory function of a phone is generally a matter of adding a new view for the format that the phone consumes.
- Namespace:
xivo_dird.views
- Abstract view plugin: BaseViewPlugin
- Methods:
load(args)
: set up resources used by the plugin, depending on the config. Typically, register routes on Flask. Those routes would typically call a service.args
is a dictionary containing:- key
config
: the section of the configuration file for all views in dict form - key
services
: a dictionary of services, indexed by name, which may be called from a route - key
http_app
: the Flask application instance - key
rest_api
: a Flask-RestFul Api instance
- key
unload()
: free resources used by the plugin.
The following example adds a simple view: GET /0.1/directories/ping
answers {"message": "pong"}
.
dummy.py
:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 | # -*- coding: utf-8 -*-
import logging
from flask_restful import Resource
logger = logging.getLogger(__name__)
class PingViewPlugin(object):
name = 'ping'
def __init__(self):
logger.debug('dummy view created')
def load(self, args):
logger.debug('dummy view args: %s', args)
args['rest_api'].add_resource(PingView, '/0.1/directories/ping')
def unload(self):
logger.debug('dummy view unloaded')
class PingView(Resource):
"""
Simple API using Flask-Restful: GET /0.1/directories/ping answers "pong"
"""
def get(self):
return {'message': 'pong'}
|
View name: default_json
Purpose: present directory entries in JSON format. The format is detailed in http://api.xivo.io.
View name: headers
Purpose: List headers that will be available in results from default_json
view.
View name: personal_view
Purpose: Expose REST API to manage personal contacts (create, delete, list).
View name: phonebook_view
Purpose: Expose REST API to manage xivo-dird’s internal phonebooks.
View name: aastra_view
Purpose: Expose REST API to search in configured directories for Aastra phone.
View name: cisco_view
Purpose: Expose REST API to search in configured directories for Cisco phone (see CiscoIPPhone_XML_Objects).
View name: polycom_view
Purpose: Expose REST API to search in configured directories for Polycom phone.
View name: snom_view
Purpose: Expose REST API to search in configured directories for Snom phone.
View name: thomson_view
Purpose: Expose REST API to search in configured directories for Thomson phone.
View name: yealink_view
Purpose: Expose REST API to search in configured directories for Yealink phone.
Service name: lookup
Purpose: Search through multiple data sources, looking for entries matching a word.
Example (excerpt from the main configuration file):
1 2 3 4 5 6 | services:
lookup:
default:
sources:
- my_csv
timeout: 0.5
|
The configuration is a dictionary whose keys are profile names and values are configuration specific to that profile.
For each profile, the configuration keys are:
- sources
- The list of source names that are to be used for the lookup
- timeout
- The maximum waiting time for an answer from any source. Results from sources that take longer to answer are ignored. Default: no timeout.
Service name: favorites
Purpose: Mark/unmark contacts as favorites and get the list of all favorites.
Service name: phonebook
Purpose: Add, delete, list phonebooks and phonebook contacts.
Example (excerpt from the main configuration file):
1 2 3 4 5 6 | services:
favorites:
default:
sources:
- my_csv
timeout: 0.5
|
The configuration is a dictionary whose keys are profile names and values are configuration specific to that profile.
For each profile, the configuration keys are:
- sources
- The list of source names that are to be used for the lookup
- timeout
- The maximum waiting time for an answer from any source. Results from sources that take longer to answer are ignored. Default: no timeout.
Service name: reverse
Purpose: Search through multiple data sources, looking for the first entry matching an extension.
Example:
1 2 3 4 5 6 | services:
reverse:
default:
sources:
- my_csv
timeout: 1
|
The configuration is a dictionary whose keys are profile names and values are configuration specific to that profile.
For each profile, the configuration keys are:
- sources
- The list of source names that are to be used for the reverse lookup
- timeout
- The maximum waiting time for an answer from any source. Results from sources that take longer to answer are ignored. Default: 1.
This sections completes the Sources Configuration section.
Back-end name: csv
Purpose: read directory entries from a CSV file.
Limitations:
- the CSV delimiter is not configurable (currently:
,
(comma)).
Example (a file inside source_config_dir
):
1 2 3 4 5 6 7 8 9 10 11 12 13 | type: csv
name: my_csv
file: /var/tmp/test.csv
unique_column: id
searched_columns:
- fn
- ln
first_matched_columns:
- num
format_columns:
lastname: "{ln}"
firstname: "{fn}"
number: "{num}"
|
With the CSV file:
1 2 3 4 | id,fn,ln,num
1,Alice,Abrams,55553783147
2,Bob,Benito,5551354958
3,Charles,Curie,5553132479
|
- file
- the absolute path to the CSV file
Back-end name: csv_ws
Purpose: search using a web service that returns CSV formatted results.
Given the following configuration, xivo-dird would call “https://example.com:8000/ws-phonebook?firstname=alice&lastname=alice” for a lookup for the term “alice”.
Example (a file inside source_config_dir
):
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 | type: csv_ws
name: a_csv_web_service
lookup_url: "https://example.com:8000/ws-phonebook"
list_url: "https://example.com:8000/ws-phonebook"
verify_certificate: False
searched_columns:
- firstname
- lastname
first_matched_columns:
- exten
delimiter: ","
timeout: 16
unique_column: id
format_columns:
number: "{exten}"
|
- lookup_url
- the URL used for directory searches.
- list_url (optional)
- the URL used to list all available entries. This URL is used to retrieve favorites.
- verify_certificate (optional)
- whether the SSL cert will be verified. A CA_BUNDLE path can also be provided. Defaults to True.
- delimiter (optional)
- the field delimiter in the CSV result. Default: ‘,’
- timeout (optional)
- the number of seconds before the lookup on the web service is aborted. Default: 10.
back-end name: dird_phonebook
Purpose: search the xivo-dird’s internal phonebooks
1 2 3 4 5 6 7 8 9 10 11 12 13 | type: dird_phonebook
name: phonebook
db_uri: 'postgresql://asterisk:proformatique@localhost/asterisk'
tenant: default
phonebook_id: 42
phonebook_name: main
first_matched_columns:
- number
searched_columns:
- firstname
- lastname
format_columns:
name: "{firstname} {lastname}"
|
- db_uri
- the URI of the DB used by xivo-dird to store the phonebook.
- tenant
- the tenant of the phonebook to query.
- phonebook_name
- the name of the phonebook used by this source.
- phonebook_id (deprecated, use phonebook_name)
- the id of the phonebook used by this source.
Back-end name: ldap
Purpose: search directory entries from an LDAP server.
Example (a file inside source_config_dir
):
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 | type: ldap
name: my_ldap
ldap_uri: ldap://example.org
ldap_base_dn: ou=people,dc=example,dc=org
ldap_username: cn=admin,dc=example,dc=org
ldap_password: foobar
ldap_custom_filter: (l=québec)
unique_column: entryUUID
searched_columns:
- cn
first_matched_columns:
- telephoneNumber
format_columns:
firstname: "{givenName}"
lastname: "{sn}"
number: "{telephoneNumber}"
|
- ldap_uri
- the URI of the LDAP server. Can only contains the scheme, host and port part of an LDAP URL.
- ldap_base_dn
- the DN of the entry at which to start the search
- ldap_username (optional)
the user’s DN to use when performing a “simple” bind.
Default to an empty string.
When both ldap_username and ldap_password are empty, an anonymous bind is performed.
- ldap_password (optional)
the password to use when performing a “simple” bind.
Default to an empty string.
- ldap_custom_filter (optional)
the custom filter is used to add more criteria to the filter generated by the back end.
Example:
- ldap_custom_filter: (l=québec)
- searched_columns: [cn,st]
will result in the following filter being used for searches.
(&(l=québec)(|(cn=*%Q*)(st=*%Q*)))
If only the custom filter is to be used, leave the
searched_columns
field empty.This must be a valid LDAP filter, where the string
%Q
will be replaced by the (escaped) search term when performing a search.Example:
(&(o=ACME)(cn=*%Q*))
- ldap_network_timeout (optional)
the maximum time, in second, that an LDAP network operation can take. If it takes more time than that, no result is returned.
Defaults to 0.3.
- ldap_timeout (optional)
the maximum time, in second, that an LDAP operation can take.
Defaults to 1.0.
- unique_column (optional)
the column that contains a unique identifier of the entry. This is necessary for listing and identifying favorites.
For OpenLDAP, you should set this option to “entryUUID”.
For Active Directory, you should set this option to “objectGUID” and also set the “unique_column_format” option to “binary_uuid”.
- unique_column_format (optional)
the unique column’s type returned by the queried LDAP server. Valid values are “string” or “binary_uuid”.
Defaults to “string”.
Back-end name: phonebook
Purpose: search directory entries from a XiVO phone book.
Example (a file inside source_config_dir
):
1 2 3 4 5 6 7 8 9 10 11 12 | type: phonebook
name: my_phonebook
phonebook_url: https://example.org/service/ipbx/json.php/restricted/pbx_services/phonebook
phonebook_username: admin
phonebook_password: foobar
first_matched_columns:
- phonebooknumber.office.number
- phonebooknumber.mobile.number
format_columns:
firstname: "{phonebook.firstname}"
lastname: "{phonebook.lastname}"
number: "{phonebooknumber.office.number}"
|
- phonebook_url (optional)
the phonebook’s URL.
Default to
http://localhost/service/ipbx/json.php/private/pbx_services/phonebook
.The URL to use differs depending on if you are accessing the phone book locally or remotely:
- Local:
http://localhost/service/ipbx/json.php/private/pbx_services/phonebook
- Remote:
https://example.org/service/ipbx/json.php/restricted/pbx_services/phonebook
- Local:
- phonebook_username (optional)
the username to use in HTTP requests.
No HTTP authentication is tried when phonebook_username or phonebook_password are empty.
- phonebook_password (optional)
- the password to use in HTTP requests.
- phonebook_timeout (optional)
the HTTP request timeout, in seconds.
Defaults to 1.0.
To be able to access the phone book of a remote XiVO, you must create a web services access user (
) on the remote XiVO.Back-end name: personal
Purpose: search directory entries among users’ personal contacts
You should only have one source of type personal
, because only one will be used to list personal
contacts. The personal
backend needs a working Consul installation. This backend works with the
personal service, which allows users to add personal contacts.
The complete list of fields is in Personal contacts.
Example (a file inside source_config_dir
):
1 2 3 4 5 6 7 8 | type: personal
name: personal
first_matched_columns:
- number
format_columns:
firstname: "{firstname}"
lastname: "{lastname}"
number: "{number}"
|
unique_column
is not configurable, its value is always id
.
Back-end name: xivo
Purpose: add users from a XiVO (may be remote) as directory entries
Example (a file inside source_config_dir
):
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 | type: xivo
name: my_xivo
confd_config:
https: True
host: xivo.example.com
port: 9486
version: 1.1
username: admin
password: password
timeout: 3
unique_column: id
first_matched_columns:
- exten
searched_columns:
- firstname
- lastname
format_columns:
number: "{exten}"
mobile: "{mobile_phone_number}"
|
- confd_config:host
- the hostname of the XiVO (more precisely, of the xivo-confd service)
- confd_config:port
- the port of the xivo-confd service (usually 9486)
- confd_config:version
- the version of the xivo-confd API (should be 1.1)
In the directory displays (also in the main configuration file of xivo-dird, in the views
section), the
following keys are interpreted and displayed in xlet people of the XiVO Client:
title
- The
title
will be shown as a header for the column type
agent
: the field value will be ignored and replaced by an icon showing the status of the agent assigned to the contact (e.g. green icon for logged agent, red icon for unlogged agent, ...)callable
: a dropdown action on thenumber
field will be added to call the field value.email
: a dropdown action on thenumber
field will be added to send an email to the field value.favorite
: the boolean field value will be replaced by an icon showing if the status is favorite (yellow star filled) or not (yellow star empty).name
: a decoration will be added to the field value (typically a color dot) showing the presence status of the contact (e.g. Disconnected, Available, Away, ...)number
: only one number type can be defined per profile. The field value will be:- added a decoration (typically a color dot) showing the status of the phone of the contact (e.g. Offline, Ringing, Talking, ...)
- replaced with a button to call the contact with your phone when using the mouse
personal
: the boolean field value will be used to show a deletion action for the contactvoicemail
: the voicemail number of the contact
See People Xlet features Upgrade Notes for an example with screenshots.
Here are the list of available attributes of a personal contact:
id
company
email
fax
firstname
lastname
mobile
number
To be able to edit and delete personal contacts, you need a column of type personal in your display.
In the web interface under
.- Edit the filter on which you which to enable favorites.
- Add a column with the type personal and display format personal.
Enabling favorites in the XiVO client.
- Add a unique_column to your sources.
- Add a favorite column to your display
The web interface does not allow the administrator to specify the unique_column and unique_column_format. To add these configuration options, add a file to /etc/xivo-dird/sources.d containing the same name than the directory definition and all missing fields.
Example:
Given an ldap directory source using Active Directory named myactivedirectory
:

Add a file /etc/xivo-dird/sources.d/myactivedirectory.yml
with the following content to
enable favorites on this source.
name: myactivedirectory # the same name than the directory definition
unique_column: objectGUID
unique_column_format: binary_uuid
In the web interface under
.- Edit the filter on which you which to enable favorites.
- Add a column with the type favorite and display format favorite.
Some configuration options are not available in the web interface. To add configuration to a source that is configured in the web interface, create a file in /etc/xivo-dird/sources.d/ with the key name matching your web interface configuration and add all missing fields.
Example:
adding a timeout configuration to a CSV web service source
name: my_csv_web_service
timeout: 16
usage: xivo-dird [-h] [-c CONFIG_FILE] [-d] [-f] [-l LOG_LEVEL] [-u USER]
optional arguments:
-h, --help show this help message and exit
-c CONFIG_FILE, --config-file CONFIG_FILE
The path where is the config file. Default: /etc/xivo-dird/config.yml
-d, --debug Log debug messages. Overrides log_level. Default:
False
-f, --foreground Foreground, don't daemonize. Default: False
-l LOG_LEVEL, --log-level LOG_LEVEL
Logs messages with LOG_LEVEL details. Must be one of:
critical, error, warning, info, debug. Default: info
-u USER, --user USER The owner of the process.
A back-end is a connector to query a specific type of directory, e.g. one back-end to query LDAP servers, another back-end to query CSV files, etc.
A source is an instance of a back-end. One backend may be used multiples times to query multiple directories of the same type. For example, I could have the customer-csv and the employee-csv sources, each using the CSV back-end, but reading a different file.
A plugin is an extension point in xivo-dird. It is a way to add or modify the functionality of xivo-dird. There are currently three types of plugins:
- Back-ends to query different types of directories (LDAP, CSV, etc.)
- Services to provide different directory actions (lookup, reverse lookup, etc.)
- Views to expose directory results in different formats (JSON, XML, etc.)
See http://api.xivo.io, section XiVO Dird.
XiVO dird phoned¶
xivo-dird-phoned is an interface to use directory service with phone. It offers a simple REST interface to authenticate a phone and search result from XiVO dird.
xivo-dird-phoned is used through HTTP requests, using HTTP and HTTPS. Its default port is 9498 and 9499. As a user, the common operation is to search through directory from a phone. The phone need to send 2 informations:
- xivo_user_uuid: The XiVO user uuid that the phone is associated. It’s used to search through personal contacts (see personal).
- profile: The profile that the user is associated. It’s used to format results as configured.
Note
Since most phones dont’t support HTTPS, a small protection is to configure authorized_subnets in Configuration Files or in
On command line, type xivo-dird-phoned -h
to see how to use it.
Purge Logs¶
Keeping records of personal communications for long periods may be subject to local legislation, to
avoid personal data retention. Also, keeping too many records may become resource intensive for the
server. To ease the removal of such records, xivo-purge-db
is a process that removes old log
entries from the database. This allows keeping records for a maximum period and deleting older ones.
By default, xivo-purge-db removes all logs older than a year (365 days). xivo-purge-db is run nightly.
Note
Please check the laws applicable to your country and modify days_to_keep
(see below)
in the configuration file accordingly.
The following features are impacted by xivo-purge-db:
More technically, the tables purged by xivo-purge-db
are:
call_log
cel
queue_log
stat_agent_periodic
stat_call_on_queue
stat_queue_periodic
stat_switchboard_queue
We recommend to override the setting days_to_keep
from /etc/xivo-purge-db/config.yml
in a
new file in /etc/xivo-purge-db/conf.d/
.
Warning
Setting days_to_keep
to 0 will NOT disable xivo-purge-db
, and will remove ALL
logs from your system.
See Configuration priority and /etc/xivo-purge-db/config.yml
for more details.
It is possible to purge logs manually. To do so, log on to the target XiVO server and run:
xivo-purge-db
You can specify the number of days of logs to keep. For example, to purge entries older than 365 days:
xivo-purge-db -d 365
Usage of xivo-purge-db
:
usage: xivo-purge-db [-h] [-d DAYS_TO_KEEP]
optional arguments:
-h, --help show this help message and exit
-d DAYS_TO_KEEP, --days_to_keep DAYS_TO_KEEP
Number of days data will be kept in tables
After an execution of xivo-purge-db
, postgresql’s Autovacuum Daemon should perform a
VACUUM ANALYZE automatically (after 1 minute). This command marks memory as reusable but does
not actually free disk space, which is fine if your disk is not getting full. In the case when
xivo-purge-db
hasn’t run for a long time (e.g. upgrading to 15.11 or when
days_to_keep is decreased), some administrator may want to perform
a VACUUM FULL to recover disk space.
Warning
VACUUM FULL will require a service interruption. This may take several hours depending on the size of purged database.
You need to:
$ xivo-service stop
$ sudo -u postgres psql asterisk -c "VACUUM (FULL)"
$ xivo-service start
In the case you want to keep archives of the logs removed by xivo-purge-db, you may install plugins to xivo-purge-db that will be run before the purge.
XiVO does not provide any archive plugin. You will need to develop plugins for your own need. If you want to share your plugins, please open a pull request.
Each plugin is a Python callable (function or class constructor), that takes a dictionary of
configuration as argument. The keys of this dictionary are the keys taken from the configuration
file. This allows you to add plugin-specific configuration in /etc/xivo-purge-db/conf.d/
.
There is an example plugin in the xivo-purge-db git repo.
Archive name: sample
Purpose: demonstrate how to create your own archive plugin.
Each plugin needs to be explicitly enabled in the configuration of xivo-purge-db
. Here is an
example of file added in /etc/xivo-purge-db/conf.d/
:
1 2 3 | enabled_plugins:
archives:
- sample
|
The following example will be save a file in /tmp/xivo_purge_db.sample
with the following
content:
Save tables before purge. 365 days to keep!
1 2 3 4 5 | sample_file = '/tmp/xivo_purge_db.sample'
def sample_plugin(config):
with open(sample_file, 'w') as output:
output.write('Save tables before purge. {0} days to keep!'.format(config['days_to_keep']))
|
The following setup.py
shows an example of a python library that adds a plugin to xivo-purge-db:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 | #!/usr/bin/env python
# -*- coding: utf-8 -*-
from setuptools import setup
from setuptools import find_packages
setup(
name='xivo-purge-db-sample-plugin',
version='0.0.1',
description='An example program',
packages=find_packages(),
entry_points={
'xivo_purge_db.archives': [
'sample = xivo_purge_db_sample.sample:sample_plugin',
],
}
)
|
XiVO service¶
XiVO has many running services. To restart the whole stack, the xivo-service command can be used to make sure the service is restarted in the right order.
Show all services status:
xivo-service status
Stop XiVO services:
xivo-service stop
Start XiVO services:
xivo-service start
Restart XiVO services:
xivo-service restart
The commands above will only act upon XiVO services. Appending an argument
all
will also act upon nginx
and postgresql
. Example:
xivo-service restart all
UDP port 5060 will be closed while services are restarting.
XiVO sysconfd¶
xivo-sysconfd is the system configuration server for XiVO. It does quite a few different things; here’s a non exhaustive list:
- configuring network (interfaces, hostname, DNS)
- configuring high availability
- staring/stopping/restarting services
- reloading asterisk configuration
- sending some events to components (xivo-agentd, xivo-agid and xivo-ctid)
Default location: /etc/xivo/sysconfd.conf
. Format: INI.
The default location may be overwritten by the command line options.
Here’s an example of the configuration file:
[general]
xivo_config_path = /etc/xivo
templates_path = /usr/share/xivo-sysconfd/templates
custom_templates_path = /etc/xivo/sysconfd/custom-templates
backup_path = /var/backups/xivo-sysconfd
[resolvconf]
hostname_file = /etc/hostname
hostname_update_cmd = /etc/init.d/hostname.sh start
hosts_file = /etc/hosts
resolvconf_file = /etc/resolv.conf
[network]
interfaces_file = /etc/network/interfaces
[wizard]
templates_path = /usr/share/xivo-config/templates
custom_templates_path = /etc/xivo/custom-templates
[commonconf]
commonconf_file = /etc/xivo/common.conf
commonconf_cmd = /usr/sbin/xivo-update-config
commonconf_monit = /usr/sbin/xivo-monitoring-update
[openssl]
certsdir = /var/lib/xivo/certificates
[monit]
monit_checks_dir = /usr/share/xivo-monitoring/checks
monit_conf_dir = /etc/monit/conf.d
[request_handlers]
synchronous = true
[bus]
username = guest
password = guest
host = localhost
port = 5672
exchange_name = xivo
exchange_type = topic
exchange_durable = true
- synchronous
If this option is true, when xivo-sysconfd receives a request to reload the dialplan for example, it will wait for the dialplan reload to complete before replying to the request.
When this option is false, xivo-sysconfd reply to the request immediately.
Setting this option to false will speed up some operation (for example, editing a user from the web interface or from xivo-confd), but this means that there will be a small delay (up to a few seconds in the worst case) between the time you create your user and the time you can dial successfully its extension.
Ecosystem¶
Devices¶
In XiVO, there is two kind of devices:
The officially supported devices will be supported across upgrades and phone features are guaranteed to be supported on the latest version.
xivo-provd
plugins for these devices can be installed from the
“officially supported devices” repository.
Aastra has been acquired by Mitel in 2014. In XiVO, the 6700 series and 6800 series phones are still referenced as Aastra phones, for historical and compatibility reasons.
6731i | 6735i | 6737i | 6739i | 6755i | 6757i | |
---|---|---|---|---|---|---|
Provisioning | Y | Y | Y | Y | Y | Y |
H-A | Y | Y | Y | Y | Y | Y |
Directory XIVO | Y | Y | Y | Y | Y | Y |
Funckeys | 8 | 26 | 30 | 55 | 26 | 30 |
Supported programmable keys | ||||||
User with supervision function | Y | Y | Y | Y | Y | Y |
Group | Y | Y | Y | Y | Y | Y |
Queue | Y | Y | Y | Y | Y | Y |
Conference Room with supervision function | Y | Y | Y | Y | Y | Y |
General Functions | ||||||
Online call recording | N | N | N | N | N | N |
Phone status | Y | Y | Y | Y | Y | Y |
Sound recording | Y | Y | Y | Y | Y | Y |
Call recording | Y | Y | Y | Y | Y | Y |
Incoming call filtering | Y | Y | Y | Y | Y | Y |
Do not disturb | Y | Y | Y | Y | Y | Y |
Group interception | Y | Y | Y | Y | Y | Y |
Listen to online calls | Y | Y | Y | Y | Y | Y |
Directory access | Y | Y | Y | Y | Y | Y |
Filtering Boss - Secretary | Y | Y | Y | Y | Y | Y |
Transfers Functions | ||||||
Blind transfer | HK | Y | Y | HK | Y | Y |
Indirect transfer | HK | Y | Y | HK | Y | Y |
Forwards Functions | ||||||
Disable all forwarding | Y | Y | Y | Y | Y | Y |
Enable/Disable forwarding on no answer | Y | Y | Y | Y | Y | Y |
Enable/Disable forwarding on busy | Y | Y | Y | Y | Y | Y |
Enable/Disable forwarding unconditional | Y | Y | Y | Y | Y | Y |
Voicemail Functions | ||||||
Enable voicemail with supervision function | Y | Y | Y | Y | Y | Y |
Reach the voicemail | Y | Y | Y | HK | Y | Y |
Delete messages from voicemail | Y | Y | Y | Y | Y | Y |
Agent Functions | ||||||
Connect/Disconnect a static agent | Y | Y | Y | Y | Y | Y |
Connect a static agent | Y | Y | Y | Y | Y | Y |
Disconnect a static agent | Y | Y | Y | Y | Y | Y |
Parking Functions | ||||||
Parking | Y | Y | Y | Y | Y | Y |
Parking position | Y | Y | Y | Y | Y | Y |
Paging Functions | ||||||
Paging | Y | Y | Y | Y | Y | Y |
Supported expansion modules:
- Aastra® M670i (for Aastra® 35i/37i/39i/53i/55i/57i)
- Aastra® M675i (for Aastra® 35i/37i/39i/55i/57i)
6863i | 6865i | 6867i | 6869i | |
---|---|---|---|---|
Provisioning | Y | Y | Y | NT |
H-A | Y | Y | Y | Y |
Directory XIVO | Y | Y | Y | Y |
Funckeys | 0 | 8 | 38 | 68 |
Supported programmable keys | ||||
User with supervision function | N | Y | Y | Y |
Group | N | Y | Y | Y |
Queue | N | Y | Y | Y |
Conference Room with supervision function | N | Y | Y | Y |
General Functions | ||||
Online call recording | N | Y | Y | Y |
Phone status | N | Y | Y | Y |
Sound recording | N | Y | Y | Y |
Call recording | N | Y | Y | Y |
Incoming call filtering | N | Y | Y | Y |
Do not disturb | N | Y | Y | Y |
Group interception | N | Y | Y | Y |
Listen to online calls | N | Y | Y | Y |
Directory access | N | Y | Y | Y |
Filtering Boss - Secretary | N | Y | Y | Y |
Transfers Functions | ||||
Blind transfer | HK | HK | HK | HK |
Indirect transfer | HK | HK | HK | HK |
Forwards Functions | ||||
Disable all forwarding | N | Y | Y | Y |
Enable/Disable forwarding on no answer | N | Y | Y | Y |
Enable/Disable forwarding on busy | N | Y | Y | Y |
Enable/Disable forwarding unconditional | N | Y | Y | Y |
Voicemail Functions | ||||
Enable voicemail with supervision function | N | Y | Y | Y |
Reach the voicemail | N | Y | Y | Y |
Delete messages from voicemail | N | Y | Y | Y |
Agent Functions | ||||
Connect/Disconnect a static agent | N | Y | Y | Y |
Connect a static agent | N | Y | Y | Y |
Disconnect a static agent | N | Y | Y | Y |
Parking Functions | ||||
Parking | N | Y | Y | Y |
Parking position | N | Y | Y | Y |
Paging Functions | ||||
Paging | N | Y | Y | Y |
Supported expansion modules:
- Aastra® M680 (for Aastra® 6865i/6867i/6869i)
- Aastra® M685 (for Aastra® 6865i/6867i/6869i)
RFP35 | RFP36 | |
---|---|---|
Provisioning | N | N |
H-A | N | N |
Directory XIVO | N | N |
Funckeys | 0 | 0 |
SPA122 | SPA3102 | SPA8000 | |
---|---|---|---|
Provisioning | Y | Y | Y |
H-A | N | N | N |
Directory XIVO | N | N | N |
Funckeys | 0 | 0 | 0 |
For best results, activate DHCP Integration on your XiVO.
These devices can be used to connect faxes. For better success with faxes some parameters must be changed. You can read the Using analog gateways section.
Note
If you want to manually resynchronize the configuration from the ATA device you should use the following url:
http://ATA_IP/admin/resync?http://XIVO_IP:8667/CONF_FILE
where :
- ATA_IP is the IP address of the ATA,
- XIVO_IP is the IP address of your XiVO,
- CONF_FILE is one of
spa3102.cfg
,spa8000.cfg
7905G | 7906G | 7911G | 7912G | 7920 | 7921G | 7940G | 7941G | 7941G-GE | 7942G | 7960G | 7961G | 7962G | |
---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Provisioning | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
H-A | Y | Y | Y | Y | NT | NT | Y | Y | Y | Y | Y | Y | Y |
Directory XIVO | FK | FK | FK | FK | N | N | FK | FK | FK | FK | FK | FK | FK |
Funckeys | 0 | 0 | 0 | 0 | 0 | 0 | 1 | 1 | 1 | 1 | 5 | 5 | 5 |
Supported programmable keys | |||||||||||||
User with supervision function | N | N | N | N | N | N | N | N | N | N | N | N | N |
Group | N | N | N | N | N | N | N | N | N | N | N | N | N |
Queue | N | N | N | N | N | N | N | N | N | N | N | N | N |
Conference Room with supervision function | N | N | N | N | N | N | N | N | N | N | N | N | N |
General Functions | |||||||||||||
Online call recording | N | N | N | N | N | N | N | N | N | N | N | N | N |
Phone status | N | N | N | N | N | N | N | N | N | N | N | N | N |
Sound recording | N | N | N | N | N | N | N | N | N | N | N | N | N |
Call recording | N | N | N | N | N | N | N | N | N | N | N | N | N |
Incoming call filtering | N | N | N | N | N | N | N | N | N | N | N | N | N |
Do not disturb | SK | SK | SK | SK | N | N | SK | SK | SK | SK | SK | SK | SK |
Group interception | N | N | N | N | N | N | N | N | N | N | N | N | N |
Listen to online calls | N | N | N | N | N | N | N | N | N | N | N | N | N |
Directory access | Y | Y | Y | Y | N | N | Y | Y | Y | Y | Y | Y | Y |
Filtering Boss - Secretary | N | N | N | N | N | N | N | N | N | N | N | N | N |
Transfers Functions | |||||||||||||
Blind transfer | N | N | N | N | N | N | N | N | N | N | N | N | N |
Indirect transfer | SK | SK | SK | SK | SK | SK | SK | SK | SK | SK | SK | SK | SK |
Forwards Functions | |||||||||||||
Disable all forwarding | N | N | N | N | N | N | N | N | N | N | N | N | N |
Enable/Disable forwarding on no answer | N | N | N | N | N | N | N | N | N | N | N | N | N |
Enable/Disable forwarding on busy | N | N | N | N | N | N | N | N | N | N | N | N | N |
Enable/Disable forwarding unconditional | N | N | N | N | N | N | N | N | N | N | N | N | N |
Voicemail Functions | |||||||||||||
Enable voicemail with supervision function | N | N | N | N | N | N | N | N | N | N | N | N | N |
Reach the voicemail | SK | SK | SK | SK | N | N | HK | HK | HK | HK | HK | HK | HK |
Delete messages from voicemail | N | N | N | N | N | N | N | N | N | N | N | N | N |
Agent Functions | |||||||||||||
Connect/Disconnect a static agent | N | N | N | N | N | N | N | N | N | N | N | N | N |
Connect a static agent | N | N | N | N | N | N | N | N | N | N | N | N | N |
Disconnect a static agent | N | N | N | N | N | N | N | N | N | N | N | N | N |
Parking Functions | |||||||||||||
Parking | N | N | N | N | N | N | N | N | N | N | N | N | N |
Parking position | N | N | N | N | N | N | N | N | N | N | N | N | N |
Paging Functions | |||||||||||||
Paging | N | N | N | N | N | N | N | N | N | N | N | N | N |
Warning
These phones can only be used in SCCP mode. They are limited to the features supported in XIVO’s SCCP implementation.
To install firmware for xivo-cisco-sccp plugins, you need to manually download
the firmware files from the Cisco website and save them in the
/var/lib/xivo-provd/plugins/$plugin-name/var/cache
directory.
This directory is created by XiVO when you install the plugin (i.e. xivo-cisco-sccp-legacy). If you create the directory manually, the installation will fail.
Warning
Access to Cisco firmware updates requires a Cisco account with sufficient privileges. The account requires paying for the service and remains under the responsibility of the client or partner. Avencall is not responsible for these firmwares and does not offer any updates.
For example, if you have installed the xivo-cisco-sccp-legacy
plugin and you want to install the 7940-7960-fw
, networklocale
and userlocale_fr_FR
package, you must:
- Go to http://www.cisco.com
- Click on “Log In” in the top right corner of the page, and then log in
- Click on the “Support” menu
- Click on the “Downloads” tab, then on “Voice & Unified Communications”
- Select “IP Telephony”, then “Unified Communications Endpoints”, then the model of your phone (in this example, the 7940G)
- Click on “Skinny Client Control Protocol (SCCP) software”
- Choose the same version as the one shown in the plugin
- Download the file with an extension ending in ”.zip”, which is usually the last file in the list
- In the XiVO web interface, you’ll then be able to click on the “install” button for the firmware
The procedure is similar for the network locale and the user locale package, but:
- Instead of clicking on “Skinny Client Control Protocol (SCCP) software”, click on “Unified Communications Manager Endpoints Locale Installer”
- Click on “Linux”
- Choose the same version of the one shown in the plugin
- For the network locale, download the file named “po-locale-combined-network.cop.sgn”
- For the user locale, download the file named “po-locale-$locale-name.cop.sgn, for example “po-locale-fr_FR.cop.sgn” for the “fr_FR” locale
- Both files must be placed in
/var/lib/xivo-provd/plugins/$plugin-name/var/cache
directory. Then install them in the XiVO Web Interface.
Note
Currently user and network locale 11.5.1 should be used for plugins xivo-sccp-legacy and xivo-cisco-sccp-9.4
D40 | D50 | D70 | |
---|---|---|---|
Provisioning | Y | NYT | Y |
H-A | Y | NYT | Y |
Directory XIVO | N | NYT | N |
Funckeys | 2 | 14 | 106 |
Supported programmable keys | |||
User with supervision function | N | NYT | N |
Group | Y | NYT | Y |
Queue | Y | NYT | Y |
Conference Room with supervision function | Y | NYT | Y |
General Functions | |||
Online call recording | N | NYT | N |
Phone status | Y | NYT | Y |
Sound recording | Y | NYT | Y |
Call recording | Y | NYT | Y |
Incoming call filtering | Y | NYT | Y |
Do not disturb | HK | NYT | HK |
Group interception | Y | NYT | Y |
Listen to online calls | N | NYT | N |
Directory access | N | NYT | N |
Filtering Boss - Secretary | Y | NYT | Y |
Transfers Functions | |||
Blind transfer | HK | NYT | HK |
Indirect transfer | HK | NYT | HK |
Forwards Functions | |||
Disable all forwarding | Y | NYT | Y |
Enable/Disable forwarding on no answer | Y | NYT | Y |
Enable/Disable forwarding on busy | Y | NYT | Y |
Enable/Disable forwarding unconditional | Y | NYT | Y |
Voicemail Functions | |||
Enable voicemail with supervision function | Y | NYT | Y |
Reach the voicemail | HK | NYT | HK |
Delete messages from voicemail | Y | NYT | Y |
Agent Functions | |||
Connect/Disconnect a static agent | Y | NYT | Y |
Connect a static agent | Y | NYT | Y |
Disconnect a static agent | Y | NYT | Y |
Parking Functions | |||
Parking | N | NYT | N |
Parking position | N | NYT | N |
Paging Functions | |||
Paging | Y | NYT | Y |
Note
Some function keys are shared with line keys
Particularities:
- For best results, activate DHCP Integration on your XiVO.
- Impossible to do directed pickup using a BLF function key.
- Only supports DTMF in RFC2833 mode.
- Does not work reliably with Cisco ESW520 PoE switch. When connected to such a switch, the D40 tends to reboot randomly, and the D70 does not boot at all.
- It’s important to not edit the phone configuration via the phones’ web interface when using these phones with XiVO.
- Paging doesn’t work.
The Mitel 6700 Series and 6800 Series SIP Phones are supported in XiVO. See the Aastra section.
The following analog VoIP gateways are supported:
SN4112 | SN4114 | SN4116 | SN4118 | SN4316 | SN4324 | SN4332 | |
---|---|---|---|---|---|---|---|
Provisioning | Y | Y | Y | Y | Y | Y | Y |
H-A | Y | Y | Y | Y | Y | Y | Y |
XiVO only supports configuring the FXS ports of these gateways. It does not support configuring the FXO ports. If you have a gateway on which you would like to configure the FXO ports, you’ll need to write the FXO ports configuration manually by creating a custom template for your gateway.
It’s only possible to enter a provisioning code on the first FXS port of a gateway. For example, if you have a gateway with 8 FXS ports, the first port can be configured by dialing a provisioning code from it, but ports 2 to 7 can only be configured via the XiVO web interface. Also, if you dial the “reset to autoprov” extension from any port, the configuration of all the ports will be reset, not just the port on which the extension was dialed. These limitations might go away in the future.
These gateways are configured with a few regional parameters (France by default). These parameters are easy to change by writing a custom template.
Telnet access and web access are enabled by default. You should change the default password by setting an administrator password via a XiVO “template device”.
By downloading and installing the Patton firmwares, you agree to the Patton Electronics Company conditions.
To provision a gateway that was previously configured manually, use the following commands on your gateway (configure mode), replacing XIVO_IP by the IP address of your XiVO server:
profile provisioning PF_PROVISIONING_CONFIG
destination configuration
location 1 http://XIVO_IP:8667/$(system.mac).cfg
activation reload graceful
exit
provisioning execute PF_PROVISIONING_CONFIG
|SoundPoint IP | |SoundStation IP | |Business Media Phone | |||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
SPIP331 | SPIP335 | SPIP450 | SPIP550 | SPIP560 | SPIP650 | SPIP5000 | SPIP6000 | SPIP7000 | VVX101 | VVX201 | VVX300 | VVX310 | VVX400 | VVX410 | VVX500 | VVX600 | |
Provisioning [1] | NT [1] | Y | Y | Y | NT [1] | NT [1] | NT [1] | Y | NT [1] | Y | Y | Y | Y | Y | Y | Y | NYT |
H-A | N | Y | N | Y | N | N | N | N | N | Y | Y | Y | Y | Y | Y | Y | N |
Directory XIVO | N | N | N | FK | N | N | N | N | N | N | N | FK | FK | FK | FK | FK | N |
Funckeys | N | 0 | 2 | 3 | 3 | 47 | 0 | 0 | 0 | 0 | 0 | 6 | 6 | 12 | 12 | 12 | 0 |
Supported programmable keys | |||||||||||||||||
User with supervision function | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Group | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Queue | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Conference Room with supervision function | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
General Functions | |||||||||||||||||
Online call recording | NYT | N | NYT | N | NYT | NYT | NYT | NYT | NYT | N | N | N | N | N | N | N | NYT |
Phone status | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Sound recording | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Call recording | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Incoming call filtering | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Do not disturb | NYT | SK | NYT | HK | NYT | NYT | NYT | NYT | NYT | SK | SK | SK | SK | SK | SK | SK | NYT |
Group interception | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Listen to online calls | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Directory access | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Filtering Boss - Secretary | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Transfers Functions | |||||||||||||||||
Blind transfer | NYT | SK | NYT | N | NYT | NYT | NYT | NYT | NYT | SK | SK | HK | HK | HK | HK | SK | NYT |
Indirect transfer | NYT | SK | NYT | HK | NYT | NYT | NYT | NYT | NYT | SK | SK | HK | HK | HK | HK | SK | NYT |
Forwards Functions | |||||||||||||||||
Disable all forwarding | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Enable/Disable forwarding on no answer | NYT | SK | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Enable/Disable forwarding on busy | NYT | SK | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Enable/Disable forwarding unconditional | NYT | SK | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Voicemail Functions | |||||||||||||||||
Enable voicemail with supervision function | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Reach the voicemail | NYT | SK | NYT | HK | NYT | NYT | NYT | NYT | NYT | SK | SK | HK | HK | HK | HK | SK | NYT |
Delete messages from voicemail | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Agent Functions | |||||||||||||||||
Connect/Disconnect a static agent | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Connect a static agent | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Disconnect a static agent | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Parking Functions | |||||||||||||||||
Parking | NYT | N | NYT | N | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Parking position | NYT | N | NYT | N | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Paging Functions | |||||||||||||||||
Paging | NYT | N | NYT | Y | NYT | NYT | NYT | NYT | NYT | Y | Y | Y | Y | Y | Y | Y | NYT |
Particularities:
The latest Polycom firmwares can take a lot of time to download and install due to their size (~650 MiB). For this reason, these files are explicitly excluded from the XiVO backups.
For directed call pickup to work via the BLF function keys, you need to make sure that the option Set caller-id in dialog-info+xml notify is enabled on your XiVO. This option is located on the page, in the Signaling tab.
Also, directed call pickup via a BLF function key will not work if the extension number of the supervised user is different from its caller ID number.
Default password is 9486 (i.e. the word “xivo” on a telephone keypad).
On the VVX101 and VVX201, to have the two line keys mapped to the same SIP line, create a custom template with the following content:
{% extends 'base.tpl' -%} {% block remote_phonebook -%} {% endblock -%} {% block model_specific_parameters -%} reg.1.lineKeys="2" {% endblock -%}
This is especially useful on the VVX101 since it supports a maximum of 1 SIP line and does not support function keys.
Note
(XiVO HA cluster) BLF function key saved on the master node are not available.
Supported expansion modules:
- Polycom® VVX Color Expansion (for Polycom® VVX 300/310/400/410/500/600)
- Polycom® VVX Paper Expansion (for Polycom® VVX 300/310/400/410/500/600)
- Polycom® SoundPoint IP Backlit (for Polycom® SoundPoint 650)
Warning
Polycom® VVX® Camera are not supported.
370 | 710 | 715 | 720 | D725 | D745 | 760 | D765 | 821 | 870 | |
---|---|---|---|---|---|---|---|---|---|---|
Provisioning | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
H-A | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Directory XIVO | HK | SK | SK | HK | HK | HK | HK | HK | HK | HK |
Funckeys | 12 | 5 | 5 | 18 | 18 | 32 | 16 | 16 | 12 | 15 |
Supported programmable keys | ||||||||||
User with supervision function | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Group | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Queue | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Conference Room with supervision function | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
General Functions | ||||||||||
Online call recording | N | N | N | N | N | N | N | N | N | N |
Phone status | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Sound recording | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Call recording | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Incoming call filtering | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Do not disturb | HK | SK | SK | HK | HK | HK | HK | HK | HK | HK |
Group interception | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Listen to online calls | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Directory access | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Filtering Boss - Secretary | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Transfers Functions | ||||||||||
Blind transfer | Y | SK | SK | HK | HK | HK | HK | HK | HK | HK |
Indirect transfer | Y | SK | SK | HK | HK | HK | HK | HK | HK | HK |
Forwards Functions | ||||||||||
Disable all forwarding | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Enable/Disable forwarding on no answer | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Enable/Disable forwarding on busy | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Enable/Disable forwarding unconditional | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Voicemail Functions | ||||||||||
Enable voicemail with supervision function | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Reach the voicemail | HK | HK | HK | HK | HK | HK | HK | HK | HK | HK |
Delete messages from voicemail | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Agent Functions | ||||||||||
Connect/Disconnect a static agent | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Connect a static agent | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Disconnect a static agent | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Parking Functions | ||||||||||
Parking | Y | N | N | N | N | N | N | N | Y | Y |
Parking position | Y | N | N | N | N | N | N | N | Y | Y |
Paging Functions | ||||||||||
Paging | Y | Y | Y | Y | Y | Y | Y | Y | Y | Y |
Supported expansion modules:
- Snom® Vision (for Snom® 7xx series and Snom® 8xx series)
- Snom® D7 (for Snom® 7xx series)
Note
For some models, function keys are shared with line keys
There’s the following known limitations/issues with the provisioning of Snom phones in XiVO:
- If you are using Snom phones with HA, you should not assign multiple lines to the same device.
- The Snom D745 has limited space for function key labels: long labels might be split in a suboptimal way.
- When using a D7 expansion module, the function key label will not be shown on the first reboot or resynchronization. You’ll need to reboot or resynchronize the phone a second time for the label to be shown properly.
- After a factory reset of a phone, if no language and timezone are set for the “default config device” in , you will be forced to select a default language and timezone on the phone UI.
T19P | T19P E2 | T20P | T21P | T21P E2 | T22P | T26P | T28P | T32G | T38G | T40P | T41P | T42G | T46G | T48G | W52P | |
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Provisioning | Y | Y | Y | Y | Y | Y | Y | Y | NT [1] | Y | Y | Y | Y | Y | Y | Y |
H-A | Y | Y | Y | Y | Y | Y | Y | Y | N | N | Y | Y | Y | Y | Y | Y |
Directory XIVO | N | Y | N | N | Y | N | N | N | Y | Y | Y | Y | Y | N | Y | Y |
Funckeys | 0 | 0 | 2 | 2 | 2 | 3 | 13 | 16 | 3 | 16 | 3 | 15 | 15 | 27 | 27 | 0 |
Supported programmable keys | ||||||||||||||||
User with supervision function | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Group | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Queue | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Conference Room with supervision function | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
General Functions | ||||||||||||||||
Online call recording | N | N | N | N | N | N | N | N | NYT | N | N | N | N | N | N | N |
Phone status | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Sound recording | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Call recording | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Incoming call filtering | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Do not disturb | N | N | Y | SK | SK | SK | SK | SK | NYT | SK | SK | SK | SK | SK | SK | N |
Group interception | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Listen to online calls | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Directory access | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Filtering Boss - Secretary | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Transfers Functions | ||||||||||||||||
Blind transfer | SK | SK | HK | HK | HK | HK | HK | HK | NYT | HK | SK | SK | SK | HK | HK | SK |
Indirect transfer | SK | SK | HK | HK | HK | HK | HK | HK | NYT | HK | SK | SK | SK | HK | HK | SK |
Forwards Functions | ||||||||||||||||
Disable all forwarding | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Enable/Disable forwarding on no answer | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Enable/Disable forwarding on busy | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Enable/Disable forwarding unconditional | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Voicemail Functions | ||||||||||||||||
Enable voicemail with supervision function | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Reach the voicemail | N | N | HK | HK | HK | HK | HK | HK | NYT | HK | HK | HK | HK | HK | HK | HK |
Delete messages from voicemail | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Agent Functions | ||||||||||||||||
Connect/Disconnect a static agent | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Connect a static agent | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Disconnect a static agent | N | N | Y | Y | Y | Y | Y | Y | NYT | Y | Y | Y | Y | Y | Y | N |
Parking Functions | ||||||||||||||||
Parking | N | N | Y | Y | Y | Y | Y | Y | NYT | N | Y | Y | Y | Y | Y | N |
Parking position | N | N | Y | Y | Y | Y | Y | Y | NYT | N | Y | Y | Y | Y | Y | N |
Paging Functions | ||||||||||||||||
Paging | N | N | Y | Y | Y | Y | Y | Y | NYT | N | Y | Y | Y | Y | Y | N |
See also the list of community supported Yealink models.
Regarding the W52P (DECT), there is firmware for both the base station and the handset. The base and the handset are probably going to work if they are not using the same firmware version, although this does not seem to be officially recommended. By default, a base station will try to upgrade the firmware of an handset over the air (OTA) if the following conditions are met:
- Handset with firmware 26.40.0.15 or later
- Base station with firmware 25.40.0.15 or later
- Handset with hardware 26.0.0.6 or later
Otherwise, you’ll have to manually upgrade the handset firmware via USB.
In all cases, you should consult the Yealink documentation on Upgrading W52x Handset Firmware.
Note
Some function keys are shared with line keys
Supported expansion modules:
- Yealink® EXP38 (for Yealink® T26P/T28P)
- Yealink® EXP39 (for Yealink® T26P/T28P)
- Yealink® EXP40 (for Yealink® T46G/T48G)
Caption :
[1] | (1, 2, 3, 4, 5, 6, 7) These devices are marked as Not Tested because other similar models using the same firmware have been tested instead.
If these devices ever present any bugs, they will be troubleshooted by the XiVO support team. |
The community supported devices are only supported by the community. In other words, maintenance, bug, corrections and features are developed by members of the XiVO community. XiVO does not officially endorse support for these devices.
xivo-provd
plugins for these devices can be installed from the
“community supported devices” repository.
6700i and 9000i series:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
6730i | No | 8 | Yes |
6753i | Yes | 6 | Yes |
6757i | Yes | 30 | Yes |
9143i | Yes | 7 | Yes |
9480i | No | 6 | Yes |
9480CT | No | 6 | Yes |
IP Touch series:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
4008 Extended Edition | Yes | 4 | No |
4018 Extended Edition | Yes | 4 | No |
Note that you must not download the firmware for these phones unless you agree to the fact it comes from a non-official source.
For the plugin to work fully, you need these additional packages:
apt-get install p7zip python-pexpect telnet
1200 series IP Deskphones (previously known as Nortel IP Phones):
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
1220 IP | Yes | 0 | No |
1230 IP | No | 0 | No |
Cisco Small Business SPA300 series:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
SPA301 | No | 1 | No |
SPA303 | No | 3 | No |
Note
Function keys are shared with line keys for all SPA phones
Cisco Small Business SPA500 series:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
SPA501G | Yes | 8 | No |
SPA502G | No | 1 | No |
SPA504G | Yes | 4 | No |
SPA508G | Yes | 8 | No |
SPA509G | No | 12 | No |
SPA512G | No | 1 | No |
SPA514G | No | 4 | No |
SPA525G | Yes | 5 | No |
SPA525G2 | No | 5 | No |
The SPA500 expansion module is supported.
Cisco Small Business IP Phones (previously known as Linksys IP Phones)
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
SPA901 | No | 1 | No |
SPA921 | No | 1 | No |
SPA922 | No | 1 | No |
SPA941 | No | 4 | No |
SPA942 | Yes | 4 | No |
SPA962 | Yes | 6 | No |
Note
You must install the firmware of each SPA9xx phones you are using since they reboot in loop when they can’t find their firmware.
The SPA932 expansion module is supported.
ATAs:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
PAP2 | No | 0 | No |
SPA2102 | No | 0 | No |
SPA8800 | No | 0 | No |
SPA112 | No | 0 | No |
For best results, activate DHCP Integration on your XiVO.
Note
These devices can be used to connect Faxes. For better success with faxes some parameters must be changed. You can read the Using analog gateways section.
Note
If you want to manually resynchronize the configuration from the ATA device you should use the following url:
http://ATA_IP/admin/resync?http://XIVO_IP:8667/CONF_FILE
where :
- ATA_IP is the IP address of the ATA,
- XIVO_IP is the IP address of your XiVO,
- CONF_FILE is one of
spa2102.cfg
,spa8000.cfg
Also known as Siemens.
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
C470 IP | No | 0 | No |
C475 IP | No | 0 | No |
C590 IP | No | 0 | No |
C595 IP | No | 0 | No |
C610 IP | No | 0 | No |
C610A IP | No | 0 | No |
S675 IP | No | 0 | No |
S685 IP | No | 0 | No |
N300 IP | No | 0 | No |
N300A IP | No | 0 | No |
N510 IP PRO | No | 0 | No |
Panasonic KX-HTXXX series:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
KX-HT113 | No | — | No |
KX-HT123 | No | — | No |
KX-HT133 | No | — | No |
KX-HT136 | No | — | No |
Note
This phone is for testing for the moment
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
SPIP320 | No | 0 | No |
SPIP321 | No | 0 | No |
SPIP330 | No | 0 | No |
SPIP430 | No | 0 | No |
SPIP501 | Yes | 0 | No |
SPIP600 | No | 0 | No |
SPIP601 | No | 0 | No |
SPIP670 | No | 47 | No |
SoundStation IP:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
SPIP4000 | No | 0 | No |
Others:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
VVX1500 | No | 0 | No |
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
300 | No | 6 | Yes |
320 | Yes | 12 | Yes |
360 | No | — | Yes |
820 | Yes | 4 | Yes |
MP | No | — | Yes |
PA1 | No | 0 | Yes |
Note
For some models, function keys are shared with line keys
Warning
If you are using Snom phones with HA, you should not assign multiple lines to the same device.
There’s a known issue with the provisioning of Snom phones in XiVO:
- After a factory reset of a phone, if no language and timezone are set for the “default config device” in , you will be forced to select a default language and timezone on the phone UI.
Previously known as Thomson:
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
ST2022 | No | — | — |
ST2030 | Yes | 10 | Yes |
Note
Function keys are shared with line keys
Model | Tested [1] | Fkeys [2] | XiVO HA [3] | Plugin |
---|---|---|---|---|
CP860 | No | 0 | — | xivo-yealink-v72 |
T23P | No | 3 | — | xivo-yealink-v80 |
T23G | Yes | 3 | Yes | xivo-yealink-v80 |
T27P | Yes | 21 | Yes | xivo-yealink-v80 |
T29G | No | 27 | — | xivo-yealink-v80 |
T49G | Yes | 29 | Yes | xivo-yealink-v80 |
Note
Some function keys are shared with line keys
Model | Tested [1] | Fkeys [2] | XiVO HA [3] |
---|---|---|---|
IP station | Yes | 1 | No |
[1] | (1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18) Tested means the device has been tested by the XiVO development team and that
the developers have access to this device. |
[2] | (1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18) Fkeys is the number of programmable function keys that you can configure from the
XiVO web interface. It is not necessarily the same as the number of physical function
keys the device has (for example, a 6757i has 12 physical keys but you can configure 30
function keys because of the page system). |
[3] | (1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18) XiVO HA means the device is confirmed to work with XiVO HA. |
[4] | These devices are marked as Not Tested because other similar models using the same firmware have been tested instead.
If these devices ever present any bugs, they will be troubleshooted by the XiVO support team. |
The officially supported devices will be supported across upgrades and phone features are guaranteed to be supported on the latest version.
The community supported devices are only supported by the community. In other words, maintenance, bug, corrections and features are developed by members of the XiVO community. XiVO does not officially endorse support for these devices.
The next topics lists the officially and community supported devices. For each vendor, a table shows the various features supported by XiVO. Here’s an example:
Model X | Model Y | Model Z | |
---|---|---|---|
Provisioning | Y | Y | Y |
H-A | Y | Y | Y |
Directory XIVO | N | Y | Y |
Funckeys | 0 | 2 | 8 |
Supported programmable keys | |||
User with supervision function | Y | Y | Y |
The rows have the following meaning:
- Provisioning
- Is the device supported by the auto-provisioning system ?
- H-A
- Is the device supported by the high availability system ?
- Directory XiVO
- Is the device supported by the remote directory ? In other word, is it possible to consult the XiVO’s remote directory from the device ?
- Funckeys
How many function keys can be configured on the device from the XiVO web interface ?
The number of function keys that can be configured on a device is not necessarily the same as the number of physical function keys the device has. For example, an Aastra 6757i has 12 physical keys but you can configure 30 function keys because of the page system.
Inside a table, the following legend is used:
- Y = Yes / Supported
- N = No / Not supported
- NT = Not tested
- NYT = Not yet tested
Each table also contains a section about the supported function keys. In that section, the following legend can also be used:
- FK = Funckey
- SK = SoftKey
- HK = HardKey
- MN = Menu
Function keys work using the extensions in
. It is important to enable the function keys you want to use. Also, the enable transfer option in the user configuration services tab must be enabled to use transfer function keys.Administration¶
Advanced Configuration¶
This section describes the advanced system configuration.
XiVO offers the possibility to configure the general settings via the
page.
Configure XiVO General Settings
Live reload configuration permit to reload its configuration on command received from WEBI (this option is enabled by default).
XiVO offers the possibility to create and manage X.509 certificates via the the
page.These certificates can be used for:
- enabling SIP TLS
- enabling encryption between the CTI server and the XiVO clients
For the certificate used for HTTPS, see HTTPS certificate.
You can add a certificate by clicking on the add button at the top right of the page. You’ll then be shown this page:
You should look at the examples if you don’t know which attributes to set when creating your certificates.
When removing a certificate, you should remove all the files related to that certificates.
Warning
If you remove a certificate that is used somewhere in XiVO, then you need to manually reconfigure that portion of XiVO.
For example, if you remove the certificate files used for SIP TLS, then you need to manually disable SIP TLS or asterisk will look for certificate file but it won’t be able to find them.
In the following examples, if a field is not specified than you should leave it at its default value.
You need to create both a CA certificate and a server certificate.
CA certificate:
- Name : phones-CA
- Certification authority (checkbox) : checked
- Autosigned : checked
- Valid end date : at least one month in the future
- Common name : the FQDN of your XiVO
- Organization : your organization’s name, or blank
- Email : your email or organization’s email
Server certificate:
- Name : phones
- Certification authority (select) : phones-CA
- Valid end date : at least one month in the future
- Common name : the FQDN of your XiVO
- Organization : your organization’s name, or blank
- Email : your email or organization’s email
- Name : xivo-ctid
- Autosigned : checked
- Valid end date : at least one month in the future
- Common name : the FQDN of your XiVO
- Organization : your organization’s name, or blank
- Email : your email or organization’s email
Warning
You must not set a password for the certificate. If the certificate is password protected, the CTI server will not be able to use it.
XiVO offers the possibility to integrate LDAP servers. Once configured properly, you’ll be able to search your LDAP servers from your XiVO client and from your phones (if they support this feature).
Note
This page describes how to add LDAP servers as sources of contacts. For other sources of contacts, see Directories.
You can add a LDAP server by clicking on the add button at the top right corner of the
page. You’ll then be shown this page:
Adding a LDAP server
Enter the following information:
- Name: the server’s display name
- Host: the hostname or IP address
- Port: the port number (default: 389)
- Security layer: select SSL if it is activated on your server and you want to use it (default: disabled)
- SSL means TLS/SSL (doesn’t mean StartTLS) and port 636 should then be used
- Protocol version: the LDAP protocol version (default: 3)
Warning
When editing an LDAP server, you’ll have to restart the CTI server for the changes to be taken into account.
If you are using SSL with an LDAP server that is using a CA certificate from an
unknown certificate authority, you’ll have to put the certificate file as a
single file ending with .crt
into /usr/local/share/ca-certificates
and run update-ca-certificates
.
You also need to make sure that the /etc/ldap/ldap.conf
file contains a
line TLS_CACERT /etc/ssl/certs/ca-certificates.crt
.
After that, restart spawn-fcgi with service spawn-fcgi restart
.
Also, make sure to use the FQDN of the server in the host field when using SSL. The host field must match exactly what’s in the CN attribute of the server certificate.
Next thing to do after adding a LDAP server is to create a LDAP filter via the
page.You can add a LDAP filter by clicking on the add button at the top right of the page. You’ll then be shown this page:

Adding a LDAP Filter
Enter the following information:
- Name: the filter’s display name
- LDAP server: the LDAP server this filter applies to
- User: the
dn
of the user used to do search requests - Password: the password of the given user
- Base DN: the base
dn
of search requests - Filter: if specified, it replace the default filter
In some cases, you might have to use a custom filter for your search requests instead of the default filter.
In custom filters, occurrence of the pattern %Q
is replaced by what the user entered
on its phone.
Here’s some examples of custom filters:
cn=*%Q*
&(cn=*%Q*)(mail=*@example.org)
|(cn=*%Q*)(displayName=*%Q*)
The next step is to add a directory defintion for the LDAP filter you just created. See the directories section for more information.
Here’s an example of an LDAP directory definition:

If a custom filter is defined in the LDAP filter configuration, the fields in direct match will be added to that filter using an &. To only use the filter field of your LDAP filter configuration, do not add any direct match fields in your directory definition.
Example:
- Given an LDAP filter with filter
st=Canada
- Given a directory definition with a direct match
cn,o
- Then the resulting filter when doing a search will be
&(st=Canada)(|(cn=*%Q*)(o=*%Q*))
Boss Secretary Filter¶
The boss secretary filter allow to set a secretary or a boss role to a user. Filters can then be created to filter calls directed to a boss using different strategies.
- In order to be able to use the boss secretary filter you have to :
- Select a boss role for one the users
- Select a secretary role for one ot the users
- Create a filter to set a strategy for this boss secretary filter
- Add a function key for the user boss and the user secretary
The secretary or boss role can be set in the user’s configuration page under the service tab. To use this feature, at least one boss and one secretary must be defined.
The filter is used to associate a boss to one or many secretaries and to set a ring strategy. The call filter is added in the
page.- Different ringing strategies can be applied :
- Boss rings first then all secretaries one by one
- Boss rings first then secretaries are all ringing simultaneously
- Secretaries ring one by one
- Secretaries are all ringing simultaneously
- Boss and secretaries are ringing simultaneously
- Change the caller id if the secretary wants to know which boss was initialy called
When one of serial strategies is used, the first secretary called is the last in the list. The order can be modified by drag and drop in the list.
The call filter function can be activated and deactivated by the boss or the secretary using the *37 extension. The extension is defined in
.The call filter has to be activated for each secretary if more than one is defined for a given boss.
The extension to use is *37<callfilter member id>
.
In this example, you would set 2 Func Keys
*373
and *374
on the Boss.
On the secretary Jina LaPlante
you would set *373
.
On the secretary Ptit Nouveau
you would set *374
.
A more convenient way to active the boss secretary filter is to assign a function key on the boss’ phone
or the secretary’s phone. In the user’s configuration under Func Keys
. A function key can be added
for each secretaries of a boss.
If supervision is activated, the key will light up when filter is activated for this secretary. If a secretary also has a function key on the same boss/secretary combination the function key’s BLF will be in sync between each phones.
Warning
With SCCP phones, you must configure a custom Func Keys
.
Call Completion¶
The call completion feature (or CCSS, for Call Completion Supplementary Services) in XiVO allows for a caller to be automatically called back when a called party has become available.
- To illustrate, let’s say Alice attempts to call Bob.
- Bob is currently on a phone call with Carol, though, so Bob rejects the call from Alice
- Alice then dials *40 to request call completion.
- Once Bob has finished his phone call, Alice will be automatically called back by the system.
- When she answers, Bob will be called on her behalf.
This feature has been introduced in XiVO in version 14.17.
Call completion can be used in two scenarios:
- when the called party is busy (Call Completion on Busy Subscriber)
- when the called party doesn’t answer (Call Completion on No Response)
We have already discussed the busy scenario in the introduction section.
Let’s now illustrate the no answer scenario:
- Alice attempts to call Bob.
- Bob doesn’t answer the phone. Alternatively, Alice hangs up before Bob has the time to answer the call.
- Alice then dial *40 to request call completion.
- When Bob’s phone becomes busy and then is no longer busy, Alice is automatically called back.
- When she answers, Bob will be called on her behalf.
The important thing to note here is step #4. Bob’s phone needs to become busy and then no longer busy for Alice to be called back. This means that if Bob was away when Alice called him, but when he came back he did not received nor placed any call, then Alice will not be called back.
In fact, in all scenarios, after call completion has been requested by the caller, the called phone needs to transition from busy to no longer busy for the caller to be called back. This means that in the following scenario:
- Alice attempts to call Bob.
- Bob is currently on a phone call, so he doesn’t answer the call from Alice.
- Bob finish his call a few seconds later.
- Alice then dials *40 to request call completion (Bob is not busy anymore).
Then, for Alice to be called back, Bob needs to become busy and then not busy.
If Alice is busy when Bob becomes not busy, then the call completion callback will only happen after both Alice and Bob are not busy.
When call completion is active, it can be cancelled by dialing the *40 extension.
Some timers governs the use of call completion. These are:
- offer timer: the time the caller has to request call completion. Defaults to 30 (seconds).
- busy available timer: when call completion on busy subscriber is requested, if this timer expires before the called party becomes available, then the call completion attempt will be cancelled. Defaults to 900 (seconds).
- no response available timer: similar to the “busy available timer”, but when call completion on no response is requested. Defaults to 900 (seconds).
- recall timer: when the caller who requested call completion is called back, how long the original caller’s phone rings before giving up. Defaults to 30 (seconds).
It’s currently impossible to modify the value of these timers in XiVO.
There are four special scenarios:
- the call completion will not activate
- the call completion will activate and call back for the original called party
- the call completion will activate and call back for the rerouted called party
- the call completion will activate and call back for the original called party but fail to join him
It is not possible to activate call completion in the following two scenarios.
First scenario: Alice tries to call Bob, but Bob has currently reached its “simultaneous calls” limit. When activating call completion, Alice hears that the call completion can not be activated.
Note
The “simultaneous calls” option is configured per user via the XiVO web interface.
Second scenario: Alice tries to call Bob, but the call is redirected to Charlie.
This occurs when Bob redirects/rejects the call with any of the following:
- Unconditional call forwarding towards Charlie
- Closed schedule towards Charlie
- Call permission forbidding Alice to call Bob
- Preprocess subroutine forwarding the call towards Charlie
Scenario: Alice tries to call Bob, but the call is redirected to Charlie. When activating call completion, Alice hears that the call completion is activated and eventually Alice is called back to speak with Bob.
This occurs when Bob redirects/rejects the call with any of the following:
- No-answer call forwarding towards Charlie
- Busy call forwarding towards Charlie
Scenario: Alice tries to call Bob, but the call is redirected to Charlie. When activating call completion, Alice hears that the call completion is activated and eventually Alice is called back to speak with Charlie.
This occurs when Bob redirects the call with any of the following:
- Boss-Secretary filter to the secretary Charlie
Scenario: Alice tries to call Bob, but the call is redirected to Charlie. When activating call completion, Alice hears that the call completion is activated and eventually Alice is called back to speak with Bob. But when Alice answers, Bob is not called. If Alice activates call completion again, she will hear that the call completion was cancelled.
This occurs when Bob redirects/rejects the call with any of the following:
- Do Not Disturb mode
- a new call forwarding rule that was applied after Alice activated call completion:
- Unconditional call forwarding towards Charlie
- Closed schedule towards Charlie
- Call permission forbidding Alice to call Bob
- Preprocess subroutine forwarding the call towards Charlie
- Call completion can only be used with SIP lines. It can’t be used with SCCP lines.
- It can’t be used with outgoing calls and incoming calls, except if these calls are passing through a customized trunk of type Local.
- It can’t be used with groups or queues.
- The call completion feature can’t be enabled only for a few users; either all users have access to it, or none.
The call completion extension is enabled via the General tab.
page, in the
Call Completion Extension
If your XiVO has been installed in version 14.16 or earlier, then this extension is by default disabled. Otherwise, this extension is by default enabled.
Call Permissions¶
You can manage call permissions via the
page.Call permissions can be used for:
- denying a user from calling a specific extension
- denying a user of a group from calling a specific extension
- denying a specific extension on a specific outgoing call from being called
- denying an incoming call coming from a specific extension from calling you
More than one extension can match a given call permission, either by specifying more than one extension for that permission or by using extension patterns.
You can also create permissions that allow a specific extension to be called instead of being denied. This make it possible to create a general “deny all” permission and then an “allow for some” one.
Finally, instead of unconditionally denying calling a specific extension, call permissions can instead challenge the user for a password to be able to call that extension.
As you can see, you can do a lot of things with XiVO’s call permissions. They can be used to create fairly complex rules. That said, it is probably not a good idea to so because it’s pretty sure you’ll get it somehow wrong.
Note that when creating or editing a call permission, you must at least:
- fill the Name field
- have one extension / extension pattern in the Extensions field
- Add the extension in the extensions list
- In the Users tab, select the user
Note
User’s Rightcall Code ( under Services tab) overwrite all password call permissions for the user.
Warning
The extension can be anything but it will only work if it’s the extension of a user or an extension that pass through an outgoing call. It does not work, for example, if the extension is the number of a conference room.
First, you must create a group and add the user to this group. Note that groups aren’t required to have a number.
Then,
- Add the extension in the extensions list
- In the Groups tab, select the group
- Add the extension in the extensions list
- In the Outgoing calls tab, select the outgoing call
Note that selecting both a user and an outgoing call for the same call permission doesn’t mean the call permission applies only to that user. In fact, it means that the user can’t call that extension and that the extension can’t be called on the specific outgoing call. This in redundant and you will get the same result by not selecting the user.
Call permissions on incoming calls are semantically different from the other scenarios since the extension that you add to the permission will match the extension of the caller (i.e. the caller number) and not the extension that the caller dialed (i.e. the callee number).
- Add the extension in the extensions list.
- In the Incoming calls tab, select the incoming call
Call Logs¶
Call logs are pre-generated from CEL entries. The generation is done automatically by xivo-call-logd. xivo-call-logs is also run nightly to generate call logs from CEL that were missed by xivo-call-logd.
Note
The oldest call logs are periodically removed. See Purge Logs for more details.
Call logs can be accessed using the menu
page.
Calls Records Dashboard
Specifying no start date returns all available call logs. Specifying a start date and no end date returns all call logs from start date until now.
Call logs are presented in a CSV format. Here’s an example:
Call Date,Caller,Called,Period,user Field
2015-01-02T00:00:00,Alice (1001),1002,2,userfield
The CSV format has the following specifications:
- field names are listed on the first line
- fields are separated by commas:
,
- if there is a comma in a field value, the value is surrounded by double quotes:
"
- the UTF-8 character encoding is used
Call logs are also available from xivo-confd REST API.
Call logs can also be generated manually. To do so, log on to the target XiVO server and run:
xivo-call-logs
To avoid running for too long in one time, the call logs generation is limited to the N last
unprocessed CEL entries (default 20,000). This means that successive calls to xivo-call-logs
will process N more CELs, making about N/10 more calls available in call logs, going further back in
history, while processing new calls as well.
You can specify the number of CEL entries to consider. For example, to generate calls using the 100,000 last unprocessed CEL entries:
xivo-call-logs -c 100000
CLI Tools¶
XiVO comes with a collection of console (CLI) tools to help administer the server.
xivo-dist is the xivo repository sources manager. It is used to switch between distributions (production, development, release candidate, archived version). Example use cases :
- switch to production repository :
xivo-dist xivo-five
- switch to development repository :
xivo-dist xivo-dev
- switch to release candidate repository :
xivo-dist xivo-rc
- switch to an archived version’s repository (here 14.18) :
xivo-dist xivo-14.18
Conference Room¶
In this example, we’ll add a conference room with number 1010.
First, you need to define a conference room number range for the ‘’default’’ context via the ‘’Services / IPBX / IPBX configuration / Contexts’’ page.
You can then create a conference room via the ‘’Services / IPBX / IPBX settings / Conference rooms’’ page.
In this example, we have only filled the ‘’Name’’ and ‘’Number’’ fields, the others have been left to their default value.
As you can see, there’s quite a few options when adding / editing a conference room. Here’s a description of the most common one:
- General / PIN code
- Protects your conference room with a PIN number. People trying to join the room will be asked for the PIN code.
- General / Don’t play announce for first participant
- Don’t play the “you are currently the only person in this conference” for the first participant.
- General / Max participants
- Limits the number of participants in the conference room. A value of 0 means unlimited.
CTI Server¶
The CTI server configuration options can be found in the web-interface under the services tab.
The general options allow the administrator to manage network connections between the CTI server and the clients.
The section named STARTTLS options
allows the administrator to enable
encrypted communications between the clients and xivo-ctid and specify the
certificate and private keys to use.
If no certificate and private key is configured, xivo-ctid will use the ones
located in /usr/share/xivo-certs
.
Parting options are used to isolate XiVO users from each other. These options should be used when using the same XiVO for different enterprises.
Context separation is based on the user’s line context. A user with no line is not the member of any context and will not be able to do anything with the CTI client.
Note
xivo-dird must be restarted to take into account this parameter.
xivo-ctid uses xivo-auth to authenticate users. The default authentication backend is xivo_user. To change the authentication backend, add a configuration file in /etc/xivo-ctid/conf.d with the following content:
auth:
backend: backend_name
where backend name is the name of an enabled xivo-auth Backends Plugins.
In the Status menu, under Presences, you can edit presences group. The default presence group is francais. When editing a group, you will see a list of presences and their descriptions.
To use another presence group, you can edit the CTI profile you are using and select the appropriate presence group for that profile.
- Presence name is the name of the presence
- Display name is the human readable representation of this presence
- Color status is the color associated to this presence
- Other reachable statuses is the list of presence that can be switched from this presence state
- Actions are post selection actions that are triggered by selecting this presence
action | param |
---|---|
Enable DND | {‘true’,’false’} |
Pause agent in all queues | |
Unpause agent in all queues | |
Agent logoff |
To enable encryption of CTI communications between server and clients, you have to enable STARTTLS in
Custom certificates can be added in
and used inIn your XiVO Client, in the menu
, click on the lock icon.Note
A client which chooses to use encryption will not be able to connect to a server that does not have STARTTLS enabled.
Warning
For now, there is no mechanism for strong authentication of the server. The connection is encrypted, but the identity of the server is not verified.
The CTI profiles define which features are made available to a user. You can configure which profile will be used by a user in the menu
:
You can also customize the default profiles or add new profiles in the menu
:
To choose which features are available to users using a profile, you have to select which Xlets will be available.
The Xlets are detailed in Xlets.
The Position attribute determines how the Xlets will be laid out:
- dock will display a Xlet in its own frame. This frame can have some options:
- Floating means that the frame can be detached from the main window of the CTI Client.
- Closable means that the Xlet can be hidden
- Movable means that the Xlet can be moved (either inside the main window or outside)
- Scroll means that the Xlet will display a scroll bar if the Xlet is too large.
- grid will display a Xlet inside the main window, and it will not be movable. Multiple grid Xlets will be laid out vertically (the second below the first).
- tab will display a Xlet inside a tab of the Xlet Tabber. Thus the Xlet Tabber is required and can’t be in a tab position.
The Number attribute gives the order of the Xlets, beginning with 0. The order applies only to Xlets having the same Position attribute.
Display customer informations¶
Sheets can be defined under
in the web interface. Once a sheet is defined, it has to be assigned to an event in the menu.- Model
- The model contains the content of the displayed sheet.
- Event
- Events are actions that trigger the defined sheet. A sheet can be assigned to many events. In that case, the sheet will be raised for each event.
You must give a name to your sheet to be able to select it later.
The Focus
checkbox makes the XiVO Client pop up when the sheet is displayed, if the XiVO Client
was hidden.
There are two different ways to configure the contents of the sheet:
- creating a custom sheet from the Qt designer. This gives you a total control on the layout of the information and allows you to save and process data entered during or after a call.
- listing the different fields and their content. The information will be automatically laid out in a linear fashion and will be read-only.

The Qt interface
field is the path to the UI file created by the Qt Designer. The path can
either be a local file on your XiVO starting with file://
, or a HTTP URL.
You must add a field with type form
and display value qtui
for the form to be displayed.
The Qt Designer is part of the Qt development kit and is also available in the Qt Creator. They are available on the Qt project website.
Here is an example of a small form created with Qt Designer.
The Qt Designer screenshot.
Warning
In Qt Designer, one must set ‘vertical layout’ on the top widget (right click on the top widget > Lay out > Vertical layout).
You can download the file generated by this example from Qt Designer:
example-form.ui
Text fields (QLineEdit, QLabel, QPlainTextEdit) can contain variables that will be substituted. See the variable list for more information.
Default XiVO sheet example :
Example showing all kinds of fields:
Each field is represented by the following parameters :
- Field title : name of your line used as label on the sheet.
- Field type : define the type of field displayed on the sheet. Supported field types :
- title : to create a title on your sheet
- text : show a text
- url : a simple url link, open your default browser.
- urlx : an url button
- phone : create a tel: link, you can click to call on your sheet.
- form : show the form from an ui predefined. It’s an xml ui. You need to define qtui in display format.
- Default value : if given, this value will be used when all substitutions in the display value field fail.
- Display value : you can define text, variables or both. See the variable list for more information.
Three kinds of variables are available :
- xivo- prefix is reserved and set inside the CTI server:
- xivo-where for sheet events, event triggering the sheet
- xivo-origin place from where the lookup is requested (did, internal, forcelookup)
- xivo-direction incoming or internal
- xivo-did DID number
- xivo-calleridnum
- xivo-calleridname
- xivo-calleridrdnis contains information whether there was a transfer
- xivo-calleridton Type Of Network (national, international)
- xivo-calledidnum
- xivo-calledidname
- xivo-ipbxid (xivo-astid in 1.1)
- xivo-directory : for directory requests, it is the directory database the item has been found
- xivo-queuename queue called
- xivo-agentnumber agent number called
- xivo-date formatted date string
- xivo-time formatted time string, when the sheet was triggered
- xivo-channel asterisk channel value (for advanced users)
- xivo-uniqueid asterisk uniqueid value (for advanced users)
- db- prefixed variables are defined when the reverse lookup returns a result.
For example if you want to access to the reverse lookup full name, you need to define a field
fullname
in the directory definition, mapping to the full name field in your directory. The{db-fullname}
will be replaced by the caller full name. Every field of the directory is accessible this way.
- dp- prefixed ones are the variables set through the dialplan (through UserEvent application)
For example if you want to access from the dialplan to a variable dp-test you need to add in your dialplan this line (in a subroutine):
UserEvent(dialplan2cti,UNIQUEID: ${UNIQUEID},CHANNEL: ${CHANNEL},VARIABLE: test,VALUE: "Salut")
The {dp-test}
displays Salut.
After showing a sheet, the XiVO Client can also send back information to XiVO for post-processing or archiving.
Here are the requirements:
- The sheet must contain a button named
save
to submit information - Supported widgets:
- QCalendarWidget
- QCheckBox
- QComboBox
- QDateEdit
- QDateTime
- QDateTimeEdit
- QDoubleSpinBox
- QLabel
- QLineEdit
- QList
- QPlainTextEdit
- QRadioButton
- QSpinBox
- QTimeEdit
- Fields must have their name starting with
XIVOFORM_
If you want to send information that is not visible, you can make the widget invisible on the sheet:
change the maximumWidth or maximumHeight property to 0
edit the
.ui
file and add the following property to the widget:<property name="visible"> <bool>false</bool> </property>
When a CTI client submits a custom sheet, a call_form_result event is published on the event bus.
Mostly the same syntax as the sheet with less field types available (title, body). A Systray popup will display a single title (the last one added to the list of fields) and zero, one or more fields of type ‘body’.
Warning
The popup message on MacOSX works with Growl http://growl.info. We could get simple sheet popup to work using the free Growl Fork http://www.macupdate.com/app/mac/41038/growl-fork Note that this is not officially supported.
The action is for the xivo client, so if you configure an action, please be sure you understand it’s executed by the client. You need to allow this action in the client configuration too (menu XiVO Client -> Configure, tab Functions, tick option Customer Info and in sub-tab Customer Info tick the option Allow the Automatic Opening of URL).
The field in this tab receives the URL that will be displayed in your browser. You can also use variable substitution in this field.
http://example.org/foo
opens the URL on the default browserhttp://example.org/{xivo-did}
opens the URL on the default browser, after substituting the{xivo-did}
variable. If the substitution fails, the URL will remainhttp://example.org/{xivo-did}
, i.e. the curly brackets will still be present.http://example.org/{xivo-did}?origin={xivo-origin}
opens the URL on the default browser, after substituting the variables. If at least one of the substitution is successful, the failing substitutions will be replaced by an empty string. For example, if{xivo-origin}
is replaced by ‘outcall’ but{xivo-did}
is not substituted, the resulting URL will behttp://example.org/?origin=outcall
tcp://x.y.z.co.fr:4545/?var1=a1&var2=a2
connects to TCP port 4545 on x.y.z.co.fr, sends the stringvar1=a1&var2=a2
, then closesudp://x.y.z.co.fr:4545/?var1=a1&var2=a2
connects to UDP port 4545 on x.y.z.co.fr, sends the stringvar1=a1&var2=a2
, then closes
Note
any string that would not be understood as an URL will be handled like and URL it is a process to launch and will be executed as it is written
For tcp:// and udp://, it is a requirement that the string between / and ? is empty. An extension could be to define other serialization methods, if needed.
You can configure a sheet when a specific event is called. For example if you want to receive a sheet when an agent answers to a call, you can choose a sheet model for the Link event.
The following events are available :
- Dial: When the member’s phone starts ringing for calls on a group or queue or when the user receives a call
- Link: When a user or agent answers a call
- Unlink: When a user or agent hangup a call received from a queue
- Incoming DID: Received a call in a DID
- Hangup: Hangup the call
The informations about a call are displayed via the XiVO Client on forms called sheets.
The first step is to assign the URL to a dialplan variable. Go in the
setsheeturl.conf
. In this file, put the following:
[setsheeturl]
exten = s,1,NoOp(Starting Set Sheet URL)
same = n,Set(SHEET_URL_CTI=http://documentation.xivo.solutions)
same = n,UserEvent(dialplan2cti,UNIQUEID: ${UNIQUEID},CHANNEL: ${CHANNEL},VARIABLE: mysheeturl,VALUE: ${SHEET_URL_CTI})
same = n,Return()
You can replace documentation.xivo.solutions
by the URL you want.
The second step is to set the URL when the call is queued. To do that, we will
use a preprocessing subroutine. This is configured in the queue configuration :
go to Preprocessing subroutine
to setsheeturl
(the same
as above).
The third step is to configure the sheet to open the wanted URL. Go to
{dp-mysheeturl}
(the same as above).
The fourth and final step is to trigger the sheet when the agent answers the
queued call. Go to Agent linked
to the sheet you just created.
That’s it, you can assign agents to your queue, log the agents and make them answer calls with the XiVO Client opened, and your browser should open the specified URL.
Devices¶
First you have to display the list of devices.

Click on the synchronize button for a device.
- You will see a pop-up to confirm synchronization
- Click on the <ok> button.
You must wait until the full synchronization process has completed to determine the state returned back from the device. This can take several seconds. It is important to wait and do nothing during this time.
If synchronization is successful, a green information balloon notifies you of success.
If synchronization fails, a red information balloon warns you of failure.
Warning
When using multiple synchronization, the individual return states will not be displayed.
Select the devices you want to synchronize by checking the boxes.
A pop-up will appear requesting confirmation.
If mass synchronization was successfully sent to the devices, a green information balloon notifies you of success.
Directories¶
This page documents how to add and configure directories from custom sources. Directories added from custom sources can be used for lookup via the XiVO Client, directory feature of phones or for reverse lookup on incoming calls.
An example of adding a source and configuring source access is made for each type of source:
This type of directory is used to query the users of a XiVO. On a fresh install, the local XiVO is already configured. The URI field for this type of directory should be the base URL of a xivo-confd server.
This directory type matches the xivo backend in xivo-dird.
- id
- agent_id
- line_id
- firstname
- lastname
- exten
- context
- mobile_phone_number
- userfield
- description
- voicemail_number
The source file of the directory must be in CSV format. You will be able to choose the headers and the separator in the next steps. For example, the file will look like:
title|firstname|lastname|displayname|society|mobilenumber|email
mr|Emmett|Brown|Brown Emmett|DMC|5555551234|emmet.brown@dmc.example.com
This directory type matches the csv backend in xivo-dird.
For file directories, the Direct match and the Match reverse directories must be filled with the name of the column used to match entries.
Available fields are the one’s contained in the CSV file.
The data returned by the Web service must have the same format than the file directory. In the same way, you will be able to choose the headers and the separator in the next step.
This directory type matches the CSV web service backend in xivo-dird.
For web service directories, the Direct match and the Match reverse directories must be filled with the name of the HTTP query parameter that will be used when doing the HTTP requests.
Note that the CSV returned by the Web service is not further processed.
Manual configuration needs to be done to use a secure (SSL) connection. See CSV web service for more details.
Available fields are the ones contained in the CSV result.
http://example.org:8000/ws-phonebook return csv:
title|firstname|lastname|displayname|society|phone|email
mr|Emmett|Brown|Brown Emmett|DMC|5555551234|emmet.brown@dmc.example.com
ms|Alice|Wonderland|Wonderland Alice|DMC|5555551235|alice.wonderland@dmc.example.com

Given you have the following directory definition:
- Direct match :
search
- Match reverse directories :
phone
When a direct lookup for “Alice” is performed, then the following HTTP request:
GET /ws-phonebook?search=Alice HTTP/1.1
is emitted. When a reverse lookup for “5555551234” is performed, then the following HTTP request:
GET /ws-phonebook?phone=5555551234 HTTP/1.1
is emitted. On the reverse lookup, a filtering is performed on the result. In this example, it should have
phone
as column.

This type of directory source is the internal phonebook of a XiVO. The URI field is the one used to query the phonebook.
This directory type matches the phonebook backend in xivo-dird.
These fields are set in the General tab of the phone book.
- phonebook.description
- phonebook.displayname
- phonebook.email
- phonebook.firstname
- phonebook.fullname (this value is automatically generated as “<firstname> <lastname>”, e.g. “John Doe”)
- phonebook.lastname
- phonebook.society
- phonebook.title
- phonebook.url
These are the different phone numbers that are available
- phonebooknumber.fax.number
- phonebooknumber.home.number
- phonebooknumber.mobile.number
- phonebooknumber.office.number
- phonebooknumber.other.number
This type of directory source is the internal phonebook of XiVO dird. The URI field is used to connect to the xivo-dird database.
This directory type matches the dird_phonebook backend in xivo-dird.

URI : The URI to connect to the xivo-dird database
Tenant : Name of the tenant, the entity is used in the default configuration
Phonebook : Name of the phonebook to use

Name : Name of this source
Direct match : Fields to match when doing a lookup
Match reverse directories : Fields to match when doing a reverse lookup
Mapped fields : Add fields to be compatible with a configured display

Note
Phone IP should be in the authorized subnet to access the directories. See Remote directory.
Note
See LDAP for adding this source.
You can add new data sources via the
page.Directory name: the name of the directory
Type: there are 4 types of directory:
URI: the data source
Description: (optional) a description of the directory
Go in
and add a new directory definition.- Name: the name of the directory definition
- URI: the data source
- Delimiter: (optional) the field delimiter in the data source
- Direct match: the list used to match entries for direct lookup (comma separated)
- Match reverse directories: (optional) the list used to match entries for reverse lookup (comma separated)
- Mapped fields: used to add or modify columns in this directory source
- Fieldname: the identifier for this new field
- Value: a python format string that can be used to modify the data returned from a data source
It’s possible to do reverse lookups on incoming calls to show a better caller ID name when the caller is in one of our directories.
Reverse lookup will only be tried if at least one of the following conditions is true:
- The caller ID name is the same as the caller ID number
- The caller ID name is “unknown”
Also, reverse lookup is performed after caller ID number normalization (since XiVO 13.11).
To enable reverse lookup, you need to add an entry in Mapped fields:
- Fieldname:
reverse
- Value: the header of your data source that you want to see as the caller ID on your phone on incoming calls
- Match reverse directories:
phonebooknumber.office.number,phonebooknumber.mobile.number,phonebooknumber.home.number
- Fieldname:
reverse
- Value:
phonebook.society
This configuration will show the contact’s company name on the caller ID name, when the incoming call will match office, mobile or home number.

Phone directory takes 2 Fieldname by default:
display_name
: the displayed name on the phonephone
: the number to call
You will find below some useful configurations of Mapped fields.
Given a configuration where the directory source returns results with fields firstname and lastname . To add a name column to a directory, the administrator would add the following Mapped fields:
- Fieldname:
name
- Value:
{firstname} {lastname}
Given a directory source that need a prefix to be called, a new field can be created from an exising one. To add a prefix 9 to the numbers returned from a source, the administrator would add the following Mapped fields:
- Fieldname:
number
- Value:
9{number}
Sometimes, it can be useful to add a field to the search results. A string can be added without any formatting. To add a directory field to the xivodir directory, the administrator would add the following Mapped fields:
- Fieldname:
directory
- Value:
XiVO internal directory
Edit the default display filter or create your own in
.
Each line in the display filter will result in a header in your XiVO Client.
- Field title: text displayed in the header.
- Field type: type of the column, this information is used by the XiVO Client. (see type description)
- Default value: value that will be used if this field is empty for one of the configured sources.
- Field name: name of the field in the directory definitions. The specified names should be available in the configured sources. To add new column name to a directory definition see above.
The only way to configure display phone directory is through XiVO dird configuration.
To include a directory in direct directory definition:
- Go to .
- Edit your context.
- Select your display filter.
- Add the directories in the Directories section.
To include a directory in reverse directory definition:
- Go to .
- Add the directories to include to reverse lookups in the Related directories section.
To reload the directory configuration for XiVO Client, phone lookups and reverse lookups, use one of these methods:
- console
service xivo-dird restart
Directed Pickup¶
Directed pickup allows a user to intercept calls made to another user.
For example, if a user with number 1001 is ringing, you can dial *81001 from your phone and it will intercept (i.e. pickup) the call to this user.
The extension prefix used to pickup calls can be changed via the
page.There is a case where directed pickup does not work, which is the following:
Given you have a user U with a line of type "customized"
Given this custom line is using DAHDI technology
Given this user is a member of group G
When a call is made to group G
Then you won't be able to intercept the call made to U by pressing *8<line number of U>
If you find yourself in this situation, you’ll need to write a bit of dialplan.
For example, if you have the following:
- a user with a custom line with number 1001 in context default
- a custom line with interface
DAHDI/g1/5551234
Then add the following, or similar:
[custom_lines]
exten = line1001,1,NoOp()
same = n,Set(__PICKUPMARK=1001%default)
same = n,Dial(DAHDI/g1/5551234)
same = n,Hangup()
And do a dialplan reload
in the asterisk CLI.
Then, edit the line of the user and change the interface value to
Local/line1001@custom_lines
Note that you’ll need to update your dialplan if you update the number of the line or the context.
Entities¶
In some cases, as the telephony provider, you want different independent organisations to have their telephony served by your XiVO, e.g. different departments using the same telephony infrastructure, but you do not want each organisation to see or edit the configuration of other organisations.
In
, you can create entities, one for each independant organisation.In
, you can select an entity for each administrator.Note
Once an entity is linked with an administrator, it can not be deleted. You have to unlink the entity from all administrator to be able to delete it.
For the new entity to be useful, you need to create contexts in this entity. You may need:
- an Internal context for users, groups, queues, etc.
- an Incall context for incoming calls
- an Outcall context for outgoing calls, which should be included in the Internal context for the users to be able to call external numbers
Some fields are globally unique and will collide when the same value is used in different entities:
- User CTI login
- Agent number
- Queue name
- Context name
An error message will appear when creating resources with colliding parameters, saying the resource already exists, even if the entity-linked administrator can not see them.
Only the following lists may be filtered by entity:
- Lines
- Users
- Devices
- Groups
- Voicemails
- Conference Rooms
- Incoming calls
- Call filters
- Call pickups
- Schedules
- Agents
- Queues
For the devices:
- The filtering only applies to the devices associated with a line.
- The devices in autoprov mode or not configured mode are visible by every administrator.
The REST API does not have the notion of entity. When creating a resource without context via REST API, the resource will be associated to an arbitrary entity. Affected resources are:
- Contexts
- Call filters
- Group pickups
- Schedules
- Users
Fax¶
It’s possible to send faxes from XiVO using the fax Xlet in the XiVO client.

The fax Xlet in the XiVO Client
The file to send must be in PDF format.
If you want to receive faxes from XiVO, you need to add incoming calls definition with the Application destination and the FaxToMail application for every DID you want to receive faxes from.
This applies even if you want the action to be different from sending an email, like putting it on a FTP server. You’ll still need to enter an email address in these cases even though it won’t be used.
Note that, as usual when adding incoming call definitions, you must first define the incoming call range in the used context.

You can change the body of the email sent upon fax reception by editing /etc/xivo/mail.txt
.
The following variable can be included in the mail body:
%(dstnum)s
: the DID that received the fax
If you want to include a regular percent character, i.e. %
, you must write it as %%
in
mail.txt
or an error will occur when trying to do the variables substitution.
The agid
service must be restarted to apply changes:
service xivo-agid restart
You can change the subject of the email sent upon fax reception by editing
/etc/xivo/asterisk/xivo_fax.conf
.
Look for the [mail]
section, and in this section, modify the value of the subject
option.
The available variable substitution are the same as for the email body.
The agid
service must be restarted to apply changes:
service xivo-agid restart
You can change the from of the email sent upon fax reception by editing
/etc/xivo/asterisk/xivo_fax.conf
.
Look for the [mail]
section, and in this section, modify the value of the email_from
option.
The agid
service must be restarted to apply changes:
service xivo-agid restart
You can change the realname of the email sent upon fax reception by editing
/etc/xivo/asterisk/xivo_fax.conf
.
Look for the [mail]
section, and in this section, modify the value of the email_realname
option.
The agid
service must be restarted to apply changes:
service xivo-agid restart
The following features are only available via the /etc/xivo/asterisk/xivo_fax.conf
configuration file. They are not available from the web-interface.
The way it works is the following:
- you first declare some backends, i.e. actions to be taken when a fax is received. A backend name
looks like
mail
,ftp_example_org
orprinter_office
. - once your backends are defined, you can use them in your destination numbers. For example, when
someone calls the DID 100, you might want the
ftp_example_org
andmail
backend to be run, but otherwise, you only want themail
backend to be run.
Here’s an example of a valid /etc/xivo/asterisk/xivo_fax.conf
configuration file:
[general]
tiff2pdf = /usr/bin/tiff2pdf
mutt = /usr/bin/mutt
lp = /usr/bin/lp
[mail]
subject = FAX reception to %(dstnum)s
content_file = /etc/xivo/mail.txt
email_from = no-reply+fax@xivo.solutions
email_realname = Service Fax
[ftp_example_org]
host = example.org
username = foo
password = bar
directory = /foobar
[dstnum_default]
dest = mail
[dstnum_100]
dest = mail, ftp_example_org
The section named dstnum_default
will be used only if no DID-specific actions are defined.
After editing /etc/xivo/asterisk/xivo_fax.conf
, you need to restart the agid server
for the changes to be applied:
service xivo-agid restart
The FTP backend is used to send a PDF version of the received fax to an FTP server.
An FTP backend is always defined in a section beginning with the ftp
prefix. Here’s an example
for a backend named ftp_example_org
:
[ftp_example_org]
host = example.org
port = 2121
username = foo
password = bar
directory = /foobar
convert_to_pdf = 0
The port
option is optional and defaults to 21.
The directory
option is optional and if not specified, the document will be put in the user’s
root directory.
The convert_to_pdf
option is optional and defaults to 1. If it is set to 0, the TIFF file will
not be converted to PDF before being sent to the FTP server.
The uploaded file are named like ${XIVO_SRCNUM}-${EPOCH}.pdf
.
To use the printer backend, you must have the cups-client
package installed on your XiVO:
$ apt-get install cups-client
The printer backend uses the lp
command to print faxes.
A printer backend is always defined in a section beginning with the printer
prefix.
Here’s an example for a backend named printer_office
:
[printer_office]
name = office
convert_to_pdf = 1
When a fax will be received, the system command lp -d office <faxfile>
will be executed.
The convert_to_pdf
option is optional and defaults to 1. If it is set to 0, the TIFF file will
not be converted to PDF before being printed.
Warning
You need a CUPS server set up somewhere on your network.
By default, a mail backend named mail
is defined. You can define more mail backends if you
want. Just look what the default mail backend looks like.
XiVO does not currently support Fax Detection. A workaround is described in the Fax detection section.
XiVO is able to provision Cisco SPA122 and Linksys SPA2102, SPA3102 and SPA8000 analog gateways which can be used to connect fax equipments. This section describes the creation of custom template for SPA3102 which modifies several parameters.
Note
With SPA ATA plugins >= v0.8, you should not need to follow this section anymore since all of these parameters are now set in the base templates of all, except for Echo_Canc_Adapt_Enable, Echo_Supp_Enable, Echo_Canc_Enable.
Note
Be aware that most of the parameters are or could be country specific, i.e. :
- Preferred Codec,
- FAX Passthru Codec,
- RTP Packet Size,
- RTP-Start-Loopback Codec,
- Ring Waveform,
- Ring Frequency,
- Ring Voltage,
- FXS Port Impedance
Create a custom template for the SPA3102 base template:
cd /var/lib/xivo-provd/plugins/xivo-cisco-spa3102-5.1.10/var/templates/ cp ../../templates/base.tpl .
Add the following content before the
</flat-profile>
tag:<!-- CUSTOM TPL - for faxes - START --> {% for line_no, line in sip_lines.iteritems() %} <!-- Dial Plan: L{{ line_no }} --> <Dial_Plan_{{ line_no }}_ ua="na">([x*#].)</Dial_Plan_{{ line_no }}_> <Call_Waiting_Serv_{{ line_no }}_ ua="na">No</Call_Waiting_Serv_{{ line_no }}_> <Three_Way_Call_Serv_{{ line_no }}_ ua="na">No</Three_Way_Call_Serv_{{ line_no }}_> <Preferred_Codec_{{ line_no }}_ ua="na">G711a</Preferred_Codec_{{ line_no }}_> <Silence_Supp_Enable_{{ line_no }}_ ua="na">No</Silence_Supp_Enable_{{ line_no }}_> <Echo_Canc_Adapt_Enable_{{ line_no }}_ ua="na">No</Echo_Canc_Adapt_Enable_{{ line_no }}_> <Echo_Supp_Enable_{{ line_no }}_ ua="na">No</Echo_Supp_Enable_{{ line_no }}_> <Echo_Canc_Enable_{{ line_no }}_ ua="na">No</Echo_Canc_Enable_{{ line_no }}_> <Use_Pref_Codec_Only_{{ line_no }}_ ua="na">yes</Use_Pref_Codec_Only_{{ line_no }}_> <DTMF_Tx_Mode_{{ line_no }}_ ua="na">Normal</DTMF_Tx_Mode_{{ line_no }}_> <FAX_Enable_T38_{{ line_no }}_ ua="na">Yes</FAX_Enable_T38_{{ line_no }}_> <FAX_T38_Redundancy_{{ line_no }}_ ua="na">1</FAX_T38_Redundancy_{{ line_no }}_> <FAX_Passthru_Method_{{ line_no }}_ ua="na">ReINVITE</FAX_Passthru_Method_{{ line_no }}_> <FAX_Passthru_Codec_{{ line_no }}_ ua="na">G711a</FAX_Passthru_Codec_{{ line_no }}_> <FAX_Disable_ECAN_{{ line_no }}_ ua="na">yes</FAX_Disable_ECAN_{{ line_no }}_> <FAX_Tone_Detect_Mode_{{ line_no }}_ ua="na">caller or callee</FAX_Tone_Detect_Mode_{{ line_no }}_> <Network_Jitter_Level_{{ line_no }}_ ua="na">very high</Network_Jitter_Level_{{ line_no }}_> <Jitter_Buffer_Adjustment_{{ line_no }}_ ua="na">disable</Jitter_Buffer_Adjustment_{{ line_no }}_> {% endfor %} <!-- SIP Parameters --> <RTP_Packet_Size ua="na">0.020</RTP_Packet_Size> <RTP-Start-Loopback_Codec ua="na">G711a</RTP-Start-Loopback_Codec> <!-- Regional parameters --> <Ring_Waveform ua="rw">Sinusoid</Ring_Waveform> <!-- options: Sinusoid/Trapezoid --> <Ring_Frequency ua="rw">50</Ring_Frequency> <Ring_Voltage ua="rw">85</Ring_Voltage> <FXS_Port_Impedance ua="na">600+2.16uF</FXS_Port_Impedance> <Caller_ID_Method ua="na">Bellcore(N.Amer,China)</Caller_ID_Method> <Caller_ID_FSK_Standard ua="na">bell 202</Caller_ID_FSK_Standard> <!-- CUSTOM TPL - for faxes - END -->
Reconfigure the devices with:
xivo-provd-cli -c 'devices.using_plugin("xivo-cisco-spa3102-5.1.10").reconfigure()'
Then reboot the devices:
xivo-provd-cli -c 'devices.using_plugin("xivo-cisco-spa3102-5.1.10").synchronize()'
Most of this template can be copy/pasted for a SPA2102 or SPA8000.
Fax transmission, to be successful, MUST use G.711 codec. Fax streams cannot be encoded with lossy compression codecs (like G.729a).
That said, you may want to establish a SIP trunk using G.729a for all other communications to save bandwith. Here’s a way to be able to receive a fax in this configuration.
Note
There are some prerequisites:
- your SIP Trunk must offer both G.729a and G.711 codecs
- your fax users must have a customized outgoing calleridnum (for the codec change is based on this variable)
We assume that outgoing call rules and fax users with their DID are created
Create the file
/etc/asterisk/extensions_extra.d/fax.conf
with the following content:;; For faxes : ; The following subroutine forces inbound and outbound codec to alaw. ; For outbound codec selection we must set the variable with inheritance. ; Must be set on each Fax DID [pre-incall-fax] exten = s,1,NoOp(### Force alaw codec on both inbound (operator side) and outbound (analog gw side) when calling a Fax ###) exten = s,n,Set(SIP_CODEC_INBOUND=alaw) exten = s,n,Set(__SIP_CODEC_OUTBOUND=alaw) exten = s,n,Return() ; The following subroutine forces outbound codec to alaw based on outgoing callerid number ; For outbound codec selection we must set the variable with inheritance. ; Must be set on each outgoing call rule [pre-outcall-fax] exten = s,1,NoOp(### Force alaw codec if caller is a Fax ###) exten = s,n,GotoIf($["${CALLERID(num)}" = "0112697845"]?alaw:) exten = s,n,GotoIf($["${CALLERID(num)}" = "0112697846"]?alaw:end) exten = s,n(alaw),Set(__SIP_CODEC_OUTBOUND=alaw) exten = s,n(end),Return()
For each Fax users’ DID add the following string in the
Preprocess subroutine
field:pre-incall-fax
For each Outgoing call rule add the the following string in the
Preprocess subroutine
field:pre-outcall-fax
Graphics¶
The Services/Graphics section gives a historical overview of a XiVO system’s activity based on snapshots recorded every 5 minutes. Graphics are available for the following resources :
- CPU
- Entropy
- Interruptions
- IRQ Stats
- System Load
- Memory Usage
- Open Files
- Open Inodes
- Swap Usage
Each section is presented as a series of 4 graphics : daily, weekly, monthly and yearly history. Each graphic can be clicked on to zoom. All information presented is read only.
Groups¶
Groups are used to be able to call a set or users.
Group name cannot be general
reserved in asterisk configuration.
Group Pickup¶
Pickup groups allow users to intercept calls directed towards other users of the group. This is done either by dialing a special extension or by pressing a function key.
- In order to be able to use group pickup you have to:
- Create a pickup group
- Enable an extension to intercept calls
- Add a function key to interceptors
Pickup groups can be created in the
page.In the general tab, you can define a name and a description for the pickup group. In the Interceptors tab, you can define a list of users, groups or queues that can intercept calls. In the Intercepted tab, you can define a list of users, groups or queues that can be intercepted.
The pickup extension can be defined in the
page.The extension used by group pickup is called Group interception it’s default value is *8.
Warning
The extension must be enabled even if a function key is used.
To assign a function to an interceptor, go to
, edit an interceptor and go to the Func Keys tab.Add a new function key of type Group Interception and save.
Server/Hardware¶
This section describes how to configure the telephony hardware on a XiVO server.
Note
Currently XiVO supports only Digium Telephony Interface cards
The configuration process is the following :
For your Digium card to work properly you must load the appropriate DAHDI kernel module.
This is done via the file /etc/dahdi/modules
and this page will guide you through its configuration.
You can see which cards are detected by issuing the dahdi_hardware
command:
dahdi_hardware
pci:0000:05:0d.0 wcb4xxp- d161:b410 Digium Wildcard B410P
pci:0000:05:0e.0 wct4xxp- d161:0205 Wildcard TE205P (4th Gen)
This command gives the card name detected and, more importantly, the DAHDI kernel module needed for this card. In the above example you can see that two cards are detected in the system:
- a Digium B410P which needs the
wcb4xxp
module - and a Digium TE205P which needs the
wct4xxp
module
Now that we know the modules we need, we can create our configuration file:
Create the file
/etc/dahdi/modules
:touch /etc/dahdi/modules
Fill it with the modules name you found with the
dahdi_hardware
command (one module name per line). In our example, your/etc/dahdi/modules
file should contain the following lines:wcb4xxp wct4xxp
Note
In the /usr/share/dahdi/modules.sample
file you can find all the modules supported in your
XiVO version.
Now that you have loaded the correct module for your card you must:
- check if you need to follow one of the Specific configuration sections below,
- and continue with the next configuration step which is to configure the echo canceller.
This section lists some specific configuration. You should not follow them unless you have a specific need.
With E1/T1 cards you must select the correct line mode between:
- E1 : the European standard,
- and T1 : North American standard
For old generation cards (TE12x, TE20x, TE40x series) the line mode is selected via a physical jumper.
For new generation cards like TE13x, TE23x, TE43x series the line mode is selected by configuration.
If you’re configuring one of these TE13x, T23x, T43x cards then you MUST create a configuration file to set the line mode to E1:
Create the file
/etc/modprobe.d/xivo-wcte-linemode.conf
:touch /etc/modprobe.d/xivo-wcte-linemode.conf
Fill it with the following lines replacing
DAHDI_MODULE_NAME
by the correct module name (wcte13xp
,wcte43x
...):# set the card in E1/T1 mode options DAHDI_MODULE_NAME default_linemode=e1
Then, restart the services:
xivo-service restart
It is recommended to use telephony cards with an hardware echo-canceller module.
Warning
with TE13x, TE23x and TE43x cards, you MUST install the echo-canceller firmware. Otherwise the card won’t work properly.
If you have an hardware echo-canceller module you have to install its firmware.
You first need to know which firmware you have to install. The simplest way is to restart dahdi and then to lookup in the dmesg which firmware does DAHDI request at startup:
xivo-service restart
dmesg |grep firmware
[5461540.738209] wct4xxp 0000:01:0e.0: firmware: agent aborted loading dahdi-fw-oct6114-064.bin (not found?)
[5461540.738310] wct4xxp 0000:01:0e.0: VPM450: firmware dahdi-fw-oct6114-064.bin not available from userspace
In the example above you can see that the module wct4xxp
requested the dahdi-fw-oct6114-064.bin
firmware file but did not found it.
But you now know that you need the dahdi-fw-oct6114-064.bin
firmware.
When you know which firmware you need you can install it with xivo-fetchfw
utility.
Use
xivo-fetchfw
to find the name of the package. You can search fordigium
occurrences in the available packages:xivo-fetchfw search digium
Find the package name which matches the firmware file you need. In our example, we need the
dahdi-fw-oct6114-064.bin
file which is supplied by the package nameddigium-oct6114-064
:xivo-fetchfw install digium-oct6114-064
Now that you installed hardware echo-canceller firmware you must activate it
in /etc/asterisk/chan_dahdi.conf
file:
echocancel = 1
Now that you have loaded the correct module for your card you must:
- check if you need to follow one of the Specific configuration sections below,
- and continue with the next configuration step which is to configure your card according to the operator links.
This section describes some specific configuration. You should not follow them unless you have a specific need.
If you have an hardware echo-canceller you may want to use it to detect the DTMF signal (instead of asterisk).
Create the file
/etc/modprobe.d/xivo-hwec-dtmf.conf
:touch /etc/modprobe.d/xivo-hwec-dtmf.conf
Fill it with the following lines replacing
DAHDI_MODULE_NAME
by the correct module name (wcte13xp
,wct4xxp
...):options DAHDI_MODULE_NAME vpmdtmfsupport=1
Then, restart the services:
xivo-service restart
Now that you have loaded the correct DAHDI modules and configured the echo canceller you can proceed with the card configuration. Follow one of the appropriate link below :
Verify that the wcb4xxp
module is uncommented in /etc/dahdi/modules
.
If it wasn’t, do again the step Load the correct DAHDI modules.
Issue the command:
dahdi_genconf
Warning
it will erase all existing configuration in /etc/dahdi/system.conf
and /etc/asterisk/dahdi-channels.conf
files !
First step is to check /etc/dahdi/system.conf
file:
- check the span numbering,
- if needed change the clock source,
See detailed explanations of this file in the /etc/dahdi/system.conf section.
Below is an example for a typical french BRI line span:
# Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) RED
span=1,1,0,ccs,ami
# termtype: te
bchan=1-2
hardhdlc=3
echocanceller=mg2,1-2
Then you have to modify the /etc/asterisk/dahdi-channels.conf
file:
remove the unused lines like:
context = default group = 63
change the
context
lines if needed,the
signalling
should be one of:bri_net
bri_cpe
bri_net_ptmp
bri_cpe_ptmp
See some explanations of this file in the /etc/asterisk/dahdi-channels.conf section.
Below is an example for a typical french BRI line span:
; Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) RED
group = 0,11 ; belongs to group 0 and 11
context = from-extern ; incoming call to this span will be sent in 'from-extern' context
switchtype = euroisdn
signalling = bri_cpe ; use 'bri_cpe' signalling
channel => 1-2 ; the above configuration applies to channels 1 and 2
Now that you have configured your BRI card:
- you must check if you need to follow one of the Specific configuration sections below,
- then, if you have another type of card to configure, you can go back to the configure your card section,
- if you have configured all your card you have to configure the DAHDI interconnections in the web interface.
You will find below 3 configurations that we recommend for BRI lines. These configurations were tested on different type of french BRI lines with success.
Note
The pre-requisites are:
- XiVO >= 14.12,
- Use per-port dahdi interconnection (see the DAHDI interconnections section)
If you don’t know which one to configure we recommend that you try each one after the other in this order:
- PTMP without layer1/layer2 persistence
- PTMP with layer1/layer2 persistence
- PTP with layer1/layer2 persistence
In this mode we will configure asterisk and DAHDI:
- to use Point-to-Multipoint (PTMP) signalling,
- and to leave Layer1 and Layer2 DOWN
Follow theses steps to configure:
Before the line
#include dahdi-channels.conf
add, in file/etc/asterisk/chan_dahdi.conf
, the following lines:layer1_presence = ignore layer2_persistence = leave_down
In the file
/etc/asterisk/dahdi-channels.conf
usebri_cpe_ptmp
signalling:signalling = bri_cpe_ptmp
Create the file
/etc/modprobe.d/xivo-wcb4xxp.conf
to deactivate the layer1 persistence:touch /etc/modprobe.d/xivo-wcb4xxp.conf
Fill it with the following content:
options wcb4xxp persistentlayer1=0
Then, apply the configuration by restarting the services:
xivo-service restart
Note
Expected behavior:
- The dahdi show status command should show the BRI spans in RED status if there is no call,
- For outgoing calls the layer1/layer2 should be brought back up by the XiVO (i.e. asterisk/chan_dahdi),
- For incoming calls the layer1/layer2 should be brought back up by the operator,
- You can consider that there is a problem only if incoming or outgoing calls are rejected.
In this mode we will configure asterisk and DAHDI:
- to use Point-to-Multipoint (PTMP) signalling,
- and to keep Layer1 and Layer2 UP
Follow theses steps to configure:
Before the line
#include dahdi-channels.conf
add, in file/etc/asterisk/chan_dahdi.conf
, the following lines:layer1_presence = required layer2_persistence = keep_up
In the file
/etc/asterisk/dahdi-channels.conf
usebri_cpe_ptmp
signalling:signalling = bri_cpe_ptmp
If it exists, delete the file
/etc/modprobe.d/xivo-wcb4xxp.conf
:rm /etc/modprobe.d/xivo-wcb4xxp.conf
Then, apply the configuration by restarting the services:
xivo-service restart
Note
Expected behavior:
- The dahdi show status command should show the BRI spans in OK status even if there is no call,
- In asterisk CLI you may see the spans going Up/Down/Up : it is a problem only if incoming or outgoing calls are rejected.
In this mode we will configure asterisk and DAHDI:
- to use Point-to-Point (PTP) signalling,
- and use default behavior for Layer1 and Layer2.
Follow theses steps to configure:
In file
/etc/asterisk/chan_dahdi.conf
remove all occurrences oflayer1_presence
andlayer2_persistence
options.In the file
/etc/asterisk/dahdi-channels.conf
usebri_cpe
signalling:signalling = bri_cpe
If it exists, delete the file
/etc/modprobe.d/xivo-wcb4xxp.conf
:rm /etc/modprobe.d/xivo-wcb4xxp.conf
Then, apply the configuration by restarting the services:
xivo-service restart
Note
Expected behavior:
- The dahdi show status command should show the BRI spans in OK status even if there is no call,
- In asterisk CLI you should not see the spans going Up and Down : if it happens, it is a problem only if incoming or outgoing calls are rejected.
Verify that the correct module is configured in /etc/dahdi/modules
depending on the card you installed in your server.
If it wasn’t, do again the step Load the correct DAHDI modules
Warning
TE13x, TE23x, TE43x cards :
- these cards need a specific dahdi module configuration. See TE13x, TE23x, TE43x: E1/T1 selection paragraph,
- you MUST install the correct echo-canceller firmware to be able to use these cards. See Hardware Echo-cancellation paragraph.
Issue the command:
dahdi_genconf
Warning
it will erase all existing configuration in /etc/dahdi/system.conf
and /etc/asterisk/dahdi-channels.conf
files !
First step is to check /etc/dahdi/system.conf
file:
- check the span numbering,
- if needed change the clock source,
- usually (at least in France) you should remove the
crc4
See detailed explanations of this file in the /etc/dahdi/system.conf section.
Below is an example for a typical french PRI line span:
# Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" CCS/HDB3/CRC4 RED
span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
Then you have to modify the /etc/asterisk/dahdi-channels.conf
file:
remove the unused lines like:
context = default group = 63
change the
context
lines if needed,the
signalling
should be one of:pri_net
pri_cpe
Below is an example for a typical french PRI line span:
; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" CCS/HDB3/CRC4 RED
group = 0,11 ; belongs to group 0 and 11
context = from-extern ; incoming call to this span will be sent in 'from-extern' context
switchtype = euroisdn
signalling = pri_cpe ; use 'pri_cpe' signalling
channel => 1-15,17-31 ; the above configuration applies to channels 1 to 15 and 17 to 31
Now that you have configured your PRI card:
- you must check if you need to follow one of the Specific configuration sections below,
- then, if you have another type of card to configure, you can go back to the configure your card section,
- if you have configured all your card you have to configure the DAHDI interconnections in the web interface.
If you have several PRI cards in your server you should link them with a synchronization cable to share the exact same clock.
To do this, you need to:
- use the coding wheel on the Digium cards to give them an order of recognition in DAHDI/Asterisk (see Digium_telephony_cards_support),
- daisy-chain the cards with a sync cable (see Digium_telephony_cards_support),
- load the DAHDI module with the
timingcable=1
option.
Create /etc/modprobe.d/xivo-timingcable.conf
file and insert the line:
options DAHDI_MODULE_NAME timingcable=1
Where DAHDI_MODULE_NAME
is the DAHDI module name of your card (e.g. wct4xxp for a TE205P).
- XiVO does not support hardware echocanceller on the TDM400 card. Users of TDM400 card willing to setup an echocanceller will have to use a software echocanceller like OSLEC.
Verify that one of the {wctdm,wctdm24xxp}
module is uncommented in /etc/dahdi/modules
depending on the card you installed in your server.
If it wasn’t, do again the step Load the correct DAHDI modules
Note
Analog cards work with card module. You must add the appropriate card module to your analog card. Either:
- an FXS module (for analog equipment - phones, ...),
- an FXO module (for analog line)
Issue the command:
dahdi_genconf
Warning
it will erase all existing configuration in /etc/dahdi/system.conf
and /etc/asterisk/dahdi-channels.conf
files !
First step is to check /etc/dahdi/system.conf
file:
- check the span numbering,
See detailed explanations of this file in the /etc/dahdi/system.conf section.
Below is an example for a typical FXS analog line span:
# Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5"
fxoks=32
echocanceller=mg2,32
Then you have to modify the /etc/asterisk/dahdi-channels.conf
file:
remove the unused lines like:
context = default group = 63
change the
context
andcallerid
lines if needed,the
signalling
should be one of:fxo_ks
for FXS lines -yes it is the reversefxs_ks
for FXO lines - yes it is the reverse
Below is an example for a typical french PRI line span:
; Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5"
signalling=fxo_ks
callerid="Channel 32" <4032>
mailbox=4032
group=5
context=default
channel => 32
Now that you have configured your PRI card:
- you must check if you need to follow one of the Specific configuration sections below,
- then, if you have another type of card to configure, you can go back to the configure your card section,
- if you have configured all your card you have to configure the DAHDI interconnections in the web interface.
If you use FXS modules you should create the file /etc/modprobe.d/xivo-tdm
and insert the line:
options DAHDI_MODULE_NAME fastringer=1 boostringer=1
Where DAHDI_MODULE_NAME is the DAHDI module name of your card (e.g. wctdm for a TDM400P).
If you use FXO modules you should create file /etc/modprobe.d/xivo-tdm
:
options DAHDI_MODULE_NAME opermode=FRANCE
Where DAHDI_MODULE_NAME is the DAHDI module name of your card (e.g. wctdm for a TDM400P).
Verify that the wctc4xxp
module is uncommented in /etc/dahdi/modules
.
If it wasn’t, do again the step Load the correct DAHDI modules.
To configure the card you have to:
Install the card firmware:
xivo-fetchfw install digium-tc400m
Comment out the following line in
/etc/asterisk/modules.conf
:noload = codec_dahdi.so
Restart asterisk:
service asterisk restart
Now that you have configured your Voice Compression card:
- you must check if you need to follow one of the Specific configuration sections below,
- then, if you have another type of card to configure, you can go back to the configure your card section.
The Digium TC400 card can be used to transcode:
- 120 G.729a channels,
- 92 G.723.1 channels,
- or 92 G.729a/G.723.1 channels.
Depending on the codec you want to transcode, you can modify the mode
parameter which can take the following value:
- mode = mixed : this the default value which activates transcoding for 92 channels in G.729a or G.723.1 (5.3 Kbit and 6.3 Kbit)
- mode = g729 : this option activates transcoding for 120 channels in G.729a
- mode = g723 : this option activates transcoding for 92 channels in G.723.1 (5.3 Kbit et 6.3 Kbit)
Create the file
/etc/modprobe.d/xivo-transcode.conf
:touch /etc/modprobe.d/xivo-transcode.conf
And insert the following lines:
options wctc4xxp mode=g729
Apply the configuration by restarting the services:
xivo-service restart
Verify that the card is correctly seen by asterisk with the
transcoder show
CLI command - this command should show the encoders/decoders registered by the TC400 card:*CLI> transcoder show 0/0 encoders/decoders of 120 channels are in use.
If you didn’t do it already, you have to restart the services to apply the configuration:
xivo-service restart
At the end of this page you will also find some general notes and DAHDI.
A span is created for each card port. Below is an example of a standard E1 port:
span=1,1,0,ccs,hdb3
dchan=16
bchan=1-15,17-31
echocanceller=mg2,1-15,17-31
Each span has to be declared with the following information:
span=<spannum>,<timing>,<LBO>,<framing>,<coding>[,crc4]
spannum
: corresponds to the span number. It starts to 1 and has to be incremented by 1 at each new span. This number MUST be unique.timing
: describes the how this span will be considered regarding the synchronization :- 0 : do not use this span as a synchronization source,
- 1 : use this span as the primary synchronization source,
- 2 : use this span as the secondary synchronization source etc.
LBO
: 0 (not used)framing
: correct values areccs
orcas
. For ISDN lines,ccs
is used.coding
: correct values arehdb3
orami
. For example,hdb3
is used for an E1 (PRI) link, whereasami
is used for T0 (french BRI) link.crc4
: this is a framing option for PRI lines. For example it is rarely use in France.
Note that the dahdi_genconf
command should usually give you the correct parameters (if you correctly set the cards
jumper). All these information should be checked with your operator.
This file contains the general parameters of the DAHDI channel.
It is not generated via the dahdi_genconf
command.
This file contains the parameters of each channel.
It is generated via the dahdi_genconf
command.
Below is an example of span definition:
group=0,11
context=from-extern
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
Note that parameters are read from top to bottom in a last match fashion and are applied to the
given channels when it reads a line channel =>
.
Here the channels 1 to 15 and 17 to 31 (it is a typical E1) are set:
- in groups 0 and 11 (see DAHDI interconnections)
- in context
from-extern
: all calls received on these channels will be sent in the contextfrom-extern
- and configured with switchtype
euroisdn
and signallingpri_cpe
It’s always useful to verify if there isn’t any missed IRQ problem with the cards.
Check:
cat /proc/dahdi/<span number>
If the IRQ misses counter increments, it’s not good:
cat /proc/dahdi/1
Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
IRQ misses: 1762187
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXOKS (In use)
4 WCTDM/0/3 FXOKS (In use)
Digium gives some hints in their Knowledge Base here : http://kb.digium.com/entry/1/63/
PRI Digium cards needs 1000 interruption per seconds. If the system cannot supply them, it increment the IRQ missed counter.
As indicated in Digium KB you should avoid shared IRQ with other equipments (like HD or NIC interfaces).
Incall¶
Interconnections¶

Situation diagram
Interconnecting two XiVO will allow you to send and receive calls between the users configured on both sides.
The steps to configure the interconnections are:
- Establish the trunk between the two XiVO, that is the SIP connection between the two servers
- Configure outgoing calls on the server(s) used to emit calls
- Configure incoming calls on the server(s) used to receive calls
For now, only SIP interconnections have been tested.
The settings below allow a trunk to be used in both directions, so it doesn’t matter which server is A and which is B.
Consider XiVO A wants to establish a trunk with XiVO B.
On XiVO B, go on page
, and create a SIP trunk:Name : xivo-trunk
Username: xivo-trunk
Password: pass
Connection type: Friend
IP addressing type: Dynamic
Context: <see below>
Note
For the moment, Name and Username need to be the same string.
The Context
field will determine which extensions will be reachable by the
other side of the trunk:
- If
Context
is set todefault
, then every user, group, conf room, queue, etc. that have an extension if thedefault
context will be reachable directly by the other end of the trunk. This setting can ease configuration if you manage both ends of the trunk. - If you are establishing a trunk with a provider, you probably don’t want
everything to be available to everyone else, so you can set the
Context
field toIncalls
. By default, there is no extension available in this context, so we will be able to configure which extension are reachable by the other end. This is the role of the incoming calls: making bridges from theIncalls
context to other contexts.
On XiVO A, create the other end of the SIP trunk on the
:Name: xivo-trunk
Username: xivo-trunk
Password: pass
Identified by: Friend
Connection type: Static
Address: <XiVO B IP address or hostname>
Context: Incalls
Register tab:
Register: checked
Transport: udp
Username: xivo-trunk
Password: pass
Remote server: <XiVO B IP address or hostname>
On both XiVO, activate some codecs, Signaling
:
Enabled codecs: at least GSM (audio)
At that point, the Asterisk command sip show registry
on XiVO B should print
a line showing that XiVO A is registered, meaning your trunk is established.
The outgoing calls configuration will allow XiVO to know which extensions will be called through the trunk.
On the call emitting server(s), go on the page
and add an outgoing call.Tab General:
Trunks: xivo-trunk
Tab Exten:
Exten: **99. (note the period at the end)
Stripnum: 4
This will tell XiVO: if any extension begins with **99
, then try to dial it
on the trunk xivo-trunk
, after removing the 4 first characters (the **99
prefix).
The most useful special characters to match extensions are:
. (period): will match one or more characters
X: will match only one character
You can find more details about pattern matching in Asterisk (hence in XiVO) on the Asterisk wiki.
Now that we have calls going out from a XiVO, we need to route incoming calls on the XiVO destination.
Note
This step is only necessary if the trunk is linked to an Incoming calls context.
To route an incoming call to the right destination in the right context, we will create an incoming call in
.Tab General:
DID: 101
Context: Incalls
Destination: User
Redirect to: someone
This will tell XiVO: if you receive an incoming call to the extension 101
in
the context Incalls
, then route it to the user someone
. The destination
context will be found automatically, depending on the context of the line of the
given user.
So, with the outgoing call set earlier on XiVO A, and with the incoming call
above set on XiVO B, a user on XiVO A will dial **99101
, and the user
someone
will ring on XiVO B.
When you want to send and receive calls to the global telephony network, one option is to subscribe to a VoIP provider. To receive calls, your XiVO needs to tell your provider that it is ready and to which IP the calls must be sent. To send calls, your XiVO needs to authenticate itself, so that the provider knows that your XiVO is authorized to send calls and whose account must be credited with the call fare.
The steps to configure the interconnections are:
- Establish the trunk between the two XiVO, that is the SIP connection between the two servers
- Configure outgoing calls on the server(s) used to emit calls
- Configure incoming calls on the server(s) used to receive calls
You need the following information from your provider:
- a username
- a password
- the name of the provider VoIP server
- a public phone number
On your XiVO, go on page
, and create a SIP/IAX trunk:Name : provider_username
Username: provider_username
Password: provider_password
Connection type: Peer
IP addressing type: voip.provider.example.com
Context: Incalls (or another incoming call context)
Register tab:
Register: checked
Transport: udp
Name: provider_username
Username: provider_username
Password: provider_password
Remote server: voip.provider.example.com
Note
For the moment, Name and Username need to be the same value.
If your XiVO is behind a NAT device or a firewall, you should set the following:
Monitoring: Yes
This option will make Asterisk send a signal to the VoIP provider server every 60 seconds (default settings), so that NATs and firewall know the connection is still alive. If you want to change the value of this cycle period, you have to select the appropriate value of the following parameter:
Qualify Frequency:
At that point, the Asterisk command sip show registry
should print a line
showing that you are registered, meaning your trunk is established.
The outgoing calls configuration will allow XiVO to know which extensions will be called through the trunk.
Go on the page
and add an outgoing call.Tab General:
Trunks: provider_username
Tab Exten:
Exten: 418. (note the period at the end)
This will tell XiVO: if an internal user dials a number beginning with 418
,
then try to dial it on the trunk provider_username
.
The most useful special characters to match extensions are:
. (period): will match one or more characters
X: will match only one character
You can find more details about pattern matching in Asterisk (hence in XiVO) on the Asterisk wiki.
Now that we have calls going out, we need to route incoming calls.
To route an incoming call to the right destination in the right context, we will create an incoming call in
.Tab General:
DID: your_public_phone_number
Context: Incalls (the same than configured in the trunk)
Destination: User
Redirect to: the_front_desk_guy
This will tell XiVO: if you receive an incoming call to the public phone number
in the context Incalls
, then route it to the user
the_front_desk_guy
. The destination context will be found automatically,
depending on the context of the line of the given user.
The goal of this architecture can be one of:
- start a smooth migration between an old telephony system towards IP telephony with XiVO
- bring new features to the PBX like voicemail, conference, IVR etc.
First, XiVO is to be integrated transparently between the operator and the PBX. Then users or features are to be migrated from the PBX to the XiVO.
Warning
It requires a special call routing configuration on both the XiVO and the PBX.
You must have an ISDN card able to support both the provider and PBX ISDN links.
Example : If you have two provider links towards the PBX, XiVO should have a 4 spans card : two towards the provider, and two towards the PBX.
If you use two cards, you have to :
- Use a cable for clock synchronization between the cards
- Configure the wheel to define the cards order in the system.
Please refer to the section Sync cable
You have now to configure two files :
/etc/dahdi/system.conf
/etc/asterisk/dahdi-channels.conf
You mainly need to configure the timing
parameter on each span. As a general rule :
- Provider span - XiVO will get the clock from the provider :
the
timing
value is to be different from 0 (see /etc/dahdi/system.conf section) - PBX span - XiVO will provide the clock to the PBX :
the
timing
value is to be set to 0 (see /etc/dahdi/system.conf section)
Below is an example with two provider links and two PBX links:
# Span 1: TE4/0/1 "TE4XXP (PCI) Card 0 Span 1" (MASTER)
span=1,1,0,ccs,hdb3 # Span towards Provider
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Span 2: TE4/0/2 "TE4XXP (PCI) Card 0 Span 2"
span=2,2,0,ccs,hdb3 # Span towards Provider
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Span 3: TE4/0/3 "TE4XXP (PCI) Card 0 Span 3"
span=3,0,0,ccs,hdb3 # Span towards PBX
bchan=63-77,79-93
dchan=78
echocanceller=mg2,63-77,79-93
# Span 4: TE4/0/4 "TE4XXP (PCI) Card 0 Span 4"
span=4,0,0,ccs,hdb3 # Span towards PBX
bchan=94-108,110-124
dchan=109
echocanceller=mg2,94-108,110-124
In the file /etc/asterisk/dahdi-channels.conf
you need to adjust, for each span :
group
: the group number (e.g.0
for provider links,2
for PBX links),context
: the context (e.g.from-extern
for provider links,from-pabx
for PBX links)signalling
:pri_cpe
for provider links,pri_net
for PBX side
Warning
most of the PBX uses overlap dialing for some destination (digits are sent one by one
instead of by block). In this case, the overlapdial
parameter has to be activated on the PBX
spans:
overlapdial = incoming
Below an example of /etc/asterisk/dahdi-channels.conf
:
; Span 1: TE4/0/1 "TE4XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-extern
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
; Span 2: TE4/0/2 "TE4XXP (PCI) Card 0 Span 2"
group=0,12
context=from-extern
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
; PBX link #1
; Span 3: TE4/0/3 "TE2XXP (PCI) Card 0 Span 3"
group=2,13
context=from-pabx ; special context for PBX incoming calls
overlapdial=incoming ; overlapdial activation
switchtype = euroisdn
signalling = pri_net ; behave as the NET termination
channel => 63-77,79-93
; PBX link #2
; Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
group=2,14
context=from-pabx ; special context for PBX incoming calls
overlapdial=incoming ; overlapdial activation
switchtype = euroisdn
signalling = pri_net ; behave as the NET termination
channel => 94-108,110-124
We first need to create a route for calls coming from the PBX
# Create a file named pbx.conf
in the directory /etc/asterisk/extensions_extra.d/
,
# Add the following lines in the file:
[from-pabx]
exten = _X.,1,NoOp(### Call from PBX ${CARLLERID(num)} towards ${EXTEN} ###)
exten = _X.,n,Goto(default,${EXTEN},1)
This dialplan routes incoming calls from the PBX in the default
context of XiVO.
It enables call from the PBX :
* towards a SIP phone (in default
context)
* towards outgoing destniation (via the to-extern
context included in default
context)
In the webi, create a context named to-pabx
:
- Name : to-pabx
- Display Name : TO PBX
- Context type : Outcall
- Include sub-contexts : No context inclusion
This context will permit to route incoming calls from the XiVO to the PBX.
In our example, incoming calls on spans 1 and 2 (spans pluged to the provider) are routed by from-extern context. We are going to create a default route to redirect incoming calls to the PBX.
Create an incoming call as below :
- DID : XXXX (according to the number of digits sent by the provider)
- Context : Incoming calls
- Destination : Customized
- Command : Goto(to-pabx,${XIVO_DSTNUM},1)
You have to create two interconnections :
- provider side : dahdi/g0
- PBX side : dahdi/g2
In the menu
page :- Name : t2-operateur
- Interface : dahdi/g0
- Context : to-extern
The second interconnection :
- Name : t2-pabx
- Interface : dahdi/g2
- Context : to-pabx
You must create two rules of outgoing calls in the menu
page :- Redirect calls to the PBX :
- Name : fsc-pabx
- Context : to-pabx
- Trunks : choose the t2-pabx interconnection
In the extensions tab :
- Exten : XXXX
- Create a rule “fsc-operateur”:
- Name : fsc-operateur
- Context : to-extern
- Trunks : choose the “t2-operateur” interconnection
In the extensions tab:
exten = X.
The following configuration is based on the example found here
- username:
GV18005551212
- password:
password
- exten:
18005551212
- host:
gvgw.simonics.com
Under
.- General
- Name:
GV18005551212
- Authentication username:
GV18005551212
- Password:
password
- Caller ID:
18005551212
- Connection type:
Friend
- IP Addressing type:
static
gvgw.simonics.com
- Context:
<your incoming call context>
- Name:
- Register
- Register:
checked
- Transport:
UDP
- Name:
GV18005551212
- Password:
password
- Remote Server:
GV18005551212
- Contact:
18005551212
- Register:
- Signaling
- Monitoring:
Yes
- Monitoring:
See the Set the outgoing calls section.
See the Set the incoming calls section.
There are three types of interconnections :
- Customized
- SIP
- IAX
Customized interconnections are mainly used for interconnections using DAHDI or Local channels:
Name : it is the name which will appear in the outcall interconnections list,
Interface : this is the channel name (for DAHDI see DAHDI interconnections)
Interface suffix (optional) : a suffix added after the dialed number (in fact the Dial command will dial:
<Interface>/<EXTEN><Interface suffix>
Context : currently not relevant
To use your DAHDI links you must create a customized interconnection.
Name : the name of the interconnection like e1_span1 or bri_port1
Interface : must be of the form dahdi/[group order][group number]
where :
group order
is one of :g
: pick the first available channel in group, searching from lowest to highest,G
: pick the first available channel in group, searching from highest to lowest,r
: pick the first available channel in group, going in round-robin fashion (and remembering where it last left off), searching from lowest to highest,R
: pick the first available channel in group, going in round-robin fashion (and remembering where it last left off), searching from highest to lowest.
group number
is the group number to which belongs the span as defined in the /etc/asterisk/dahdi-channels.conf.
Warning
if you use a BRI card you MUST use per-port dahdi groups. You should not use a group like g0 which spans over several spans.
For example, add an interconnection to the menu
Name : interconnection name
Interface : dahdi/g0
When setting up an interconnection with the public network or another PBX, it is possible to set a caller ID in different places. Each way to configure a caller ID has it’s own use case.
The format for a caller ID is the following "My Name" <9999>
If you don’t set the number part of
the caller ID, the dialplan’s number will be used instead. This might not be a good option in most
cases.
When you create an outgoing call, it’s possible to set the it to internal, using the check box in the outgoing call configuration menu. When this option is activated, the caller’s caller ID will be forwarded to the trunk. This option is use full when the other side of the trunk can reach the user with it’s caller ID number.
When the caller’s caller ID is not usable to the called party, the outgoing call’s caller id can be fixed to a given value that is more use full to the outside world. Giving the public number here might be a good idea.
A user can also have a forced caller ID for outgoing calls. This can be use full for someone who has his own public number. This option can be set in the user’s configuration page. The Outgoing Caller ID id option must be set to Customize. The user can also set his outgoing caller ID to anonymous.
The order of precedence when setting the caller ID in multiple place is the following.
- Internal
- User’s outgoing caller ID
- Outgoing call
- Default caller ID
Interactive Voice Response¶
Interactive voice response (IVR) is a technology that allows a computer to interact with humans through the use of voice and DTMF tones input via keypad. In telecommunications, IVR allows customers to interact with a company’s host system via a telephone keypad or by speech recognition, after which they can service their own inquiries by following the IVR dialogue.
—Wikipedia
The IVR function is not yet available in graphic mode in XiVO. This functionality is currently supported using scripts, also named dialplan.
First step, you need to create a configuration file, that contain an asterisk context and your IVR dialpan. In our example, both (file and context) are named dp-ivr-example.

Copy all these lines in the newly created configuration file (in our case, dp-ivr-example) :
[dp-ivr-example]
exten = s,1,NoOp(### dp-ivr-example.conf ###)
same = n,NoOp(Set the context containing your ivr destinations.)
same = n,Set(IVR_DESTINATION_CONTEXT=my-ivr-destination-context)
same = n,NoOp(Set the directory containing your ivr sounds.)
same = n,Set(GV_DIRECTORY_SOUNDS=/var/lib/xivo/sounds/ivr-sounds)
same = n,NoOp(the system answers the call and waits for 1 second before continuing)
same = n,Answer(1000)
same = n,NoOp(the system plays the first part of the audio file "welcome to ...")
same = n(first),Playback(${GV_DIRECTORY_SOUNDS}/ivr-example-welcome-sound)
same = n,NoOp(variable "counter" is set to 0)
same = n(beginning),Set(counter=0)
same = n,NoOp(variable "counter" is incremented and the label "start" is defined)
same = n(start),Set(counter=$[${counter} + 1])
same = n,NoOp(counter variable is now = ${counter})
same = n,NoOp(waiting for 1 second before reading the message that indicate all choices)
same = n,Wait(1)
same = n,NoOp(play the message ivr-example-choices that contain all choices)
same = n,Background(${GV_DIRECTORY_SOUNDS}/ivr-example-choices)
same = n,NoOp(waiting for DTMF during 5s)
same = n,Waitexten(5)
;##### CHOICE 1 #####
exten = 1,1,NoOp(pressed digit is 1, redirect to 8000 in ${IVR_DESTINATION_CONTEXT} context)
exten = 1,n,Goto(${IVR_DESTINATION_CONTEXT},8000,1)
;##### CHOICE 2 #####
exten = 2,1,NoOp(pressed digit is 2, redirect to 8833 in ${IVR_DESTINATION_CONTEXT} context)
exten = 2,n,Goto(${IVR_DESTINATION_CONTEXT},8833,1)
;##### CHOICE 3 #####
exten = 3,1,NoOp(pressed digit is 3, redirect to 8547 in ${IVR_DESTINATION_CONTEXT} context)
exten = 3,n,Goto(${IVR_DESTINATION_CONTEXT},8547,1)
;##### CHOICE 4 #####
exten = 4,1,NoOp(pressed digit is 4, redirect to start label in this context)
exten = 4,n,Goto(s,start)
;##### TIMEOUT #####
exten = t,1,NoOp(no digit pressed for 5s, process it like an error)
exten = t,n,Goto(i,1)
;##### INVALID CHOICE #####
exten = i,1,NoOp(if counter variable is 3 or more, then goto label "error")
exten = i,n,GotoIf($[${counter}>=3]?error)
exten = i,n,NoOp(pressed digit is invalid and less than 3 errors: the guide ivr-exemple-invalid-choice is now played)
exten = i,n,Playback(${GV_DIRECTORY_SOUNDS}/ivr-example-invalid-choice)
exten = i,n,Goto(s,start)
exten = i,n(error),Playback(${GV_DIRECTORY_SOUNDS}/ivr-example-error)
exten = i,n,Hangup()
To call the script dp-ivr-example from an external phone, you must create an incoming call and redirect the call to the script dp-ivr-example with the command :
Goto(dp-ivr-example,s,1)

To call the script dp-ivr-example from an internal phone you must create an entry in the default
context (xivo-extrafeatures
is included in default
). The best way is to add the extension in
the file xivo-extrafeatures.conf
.

exten => 8899,1,Goto(dp-ivr-example,s,1)
In many cases, you need to associate your IVR to a schedule to indicate when your company is closed.

First step, create your schedule (1) from the menu
. In the General tab, give a name (3) to your schedule and configure the open hours (4) and select the sound which is played when the company is closed.In the Closed hours tab (6), configure all special closed days (7) and select the sound that indicate to the caller that the company is exceptionally closed.
The IVR script is now only available during workdays.

Return editing your Incall (
) and assign the newly created schedule in the “Schedules” tab
Monitoring¶
The Monitoring section gives an overview of a XiVO system’s status and of all monitored processes. It is divided into 6 sections :
Displays generic information about the operating system, network addresses, uptime and load average. Read only.
Displays free/used space on physical storage partitions. Read only.
Monitors the CPU usage. Read only.
Displays network interfaces and corresponding network traffic. Read only.
Displays Physical and swap memory usage. Read only.
Lists XiVO related processes (most of which are daemons) with their corresponding status, uptime, resource usage and controls to Restart service (blue button), stop service (red button) and stop monitoring service (grey button).
Music on Hold¶
The menu
leads to the list of available on-hold musics.Available categories are:
files: play sound files. Formats supported:
Format Name Filename Extension G.719 .g719 G.723 .g723 .g723sf G.726 .g726-40 .g726-32 .g726-24 .g726-16 G.729 .g729 GSM .gsm iLBC .ilbc Ogg Vorbis .ogg (only mono files sampled at 8000 Hz) G.711 A-law .alaw .al .alw G.711 μ-law .pcm .ulaw .ul .mu .ulw G.722 .g722 Au .au Siren7 .siren7 Siren14 .siren14 SLN .raw .sln .sln12 .sln16 .sln24 .sln32 .sln44 .sln48 .sln96 .sln192 VOX .vox WAV .wav .wav16 WAV GSM .WAV .wav49 Only 1 audio channel must be present per file, i.e. files must be in mono.
If your music on hold files don’t seem to work, you should look for errors in the asterisk logs.
The on-hold music will always play from the start.
mp3: play MP3 files.
The on-hold music will play from an arbitrary position on the track, it will not play from the start.
custom: do not play sound files. Instead, run an external process. That process must send on stdout the same binary format than WAV files.
Example process:
/usr/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://streaming.example.com/stream.mp3
Note
Processes run by custom categories are started as soon as the category is created and will only stop when the category is deleted. This means that on-hold music fed from online streaming will constantly be receiving network traffic, even when there are no calls.
Paging¶
With XiVO, you can define paging (i.e. intercom) extensions to page a group of users. When calling a paging extension, the phones of the specified users will auto-answer, if they support it.
You can manage your paging extensions via the
page.
When adding a new paging extension, the number can be any numeric value; to call it,
you just need to prefix the paging number with *11
.
Parking¶
With XiVO it is possible to park calls, the same way you may park your car in a car parking. If you define supervised keys on a phone set for all the users of a system, when a call is parked, all the users are able to see that some one is waiting for an answer, push the phone key and get the call back to the phone.
There is a default parking number, 700, which is already configured when you install XiVO, but you may change the default configuration by editing the parking extension in menu
Using this extension, you may define the parking number used to park call, the parking lots, wether the sytem is rotating over the parking lots to park the calls, enable parking hint if you want to be able to supervise the parking using phone keys and other system default parameters.
You have two options in case of parking timeout :
Callback the peer that parked this call
In this case the call is sent back to the user who parked the call.
Send park call to the dialplan
In case you don’t want to call back the user who parked the call, you have the option to send the call to any other extension or application. If the parking times out, the call is sent back to the dialplan in context
[parkedcallstimeout]
. You can define this context in a dialplan configuration file where you may define this context with dialplan commands.Example:
[parkedcallstimeout] exten = s,1,Noop('park call time out') same = n,Playback(hello-world) same = n,Hangup()
It is also usual to define supervised phone keys to be able to park and unpark calls as in the example below.
Phonebook¶
A global phone book can be defined in
. The phone book can be used from the XiVO Client, from the phones directory look key if the phone is compatible and are used to set the Caller ID for incoming calls.You can add entries one by one or you can mass-import from a CSV file.
Note
To configure phonebook, see Directories.
Go in the
section and move your mouse cursor on the + button in the upper right corner. Select Import a file.The file to be imported must be a CSV file, with a pipe character | as field delimiter. The file must be encoded in UTF-8 (without an initial BOM).
Mandatory headers are :
- title (possible values : “mr”, “mrs”, “ms”)
- displayname
Optional headers are :
- firstname
- lastname
- society
- mobilenumber [1]
- url
- description
- officenumber [1]
- faxnumber [1]
- officeaddress1
- officeaddress2
- officecity
- officestate
- officezipcode
- officecountry [2]
- homenumber [1]
- homeaddress1
- homeaddress2
- homecity
- homestate
- homezipcode
- homecountry [2]
- othernumber [1]
- otheraddress1
- otheraddress2
- othercity
- otherstate
- otherzipcode
- othercountry [2]
[1] | (1, 2, 3, 4, 5) These fields must contain only numeric characters, no space, point, etc. |
[2] | (1, 2, 3) These fields must contain ISO country codes. The complete list is described here. |
Provisioning¶
XiVO supports the auto-provisioning of a large number of telephony Devices, including SIP phones, SIP ATAs, and even softphones.
The auto-provisioning feature found in XiVO make it possible to provision, i.e. configure, a lots of telephony devices in an efficient and effortless way.
Here’s a simplified view of how auto-provisioning is supported on a typical SIP hardphone:
- The phone is powered on
- During its boot process, the phone sends a DHCP request to obtain its network configuration
- A DHCP server replies with the phone network configuration + an HTTP URL
- The phone use the provided URL to retrieve a common configuration file, a MAC-specific configuration file, a firmware image and some language files.
Building on this, configuring one of the supported phone on XiVO is as simple as:
- Configuring the DHCP Server
- Installing the required provd plugin
- Powering on the phone
- Dialing the user’s provisioning code from the phone
And voila, once the phone has rebooted, your user is ready to make and receive calls. No manual editing of configuration files nor fiddling in the phone’s web interface.
- Device synchronisation does not work in the situation where multiple devices are connected from behind a NAPT network equipment. The devices must be resynchronised manually.
You have two options to get your phone to be provisioned:
- Set up a DHCP server
- Tell manually each phone where to get the provisioning informations
You may want to manually configure the phones if you are only trying XiVO or if your network configuration does not allow the phones to access the XiVO DHCP server.
You may want to set up a DHCP server if you have a significant number of phones to connect, as no manual intervention will be required on each phone.
XiVO includes a DHCP server that facilitate the auto-provisioning of telephony devices. It is not activated by default.
There’s a few things to know about the peculiarities of the included DHCP server:
- it only answers to DHCP requests from supported devices.
- it only answers to DHCP requests coming from the VoIP subnet (see network configuration).
This means that if your phones are on the same broadcast domain than your computers, and you would like the DHCP server on your XiVO to handle both your phones and your computers, that won’t do it.
The DHCP server is configured via the
page:- Active
- Activate/desactivate the DHCP server.
- Pool start
- The lower IP address which will be assigned dynamically. This address should
be in the VoIP subnet. Example:
10.0.0.10
. - Pool end
- The higher IP address which will be assigned dynamically. This address should
be in the VoIP subnet. Example:
10.0.0.99
. - Extra network interfaces
A list of space-separated network interface name. Example:
eth0
.Useful if you have done some custom configuration in the
/etc/dhcp/dhcpd_extra.conf
file. You need to explicitly specify the additional interfaces the DHCP server should listen on.
After saving your modifications, you need to click on Apply system configuration for them to be applied.
provd
Plugins¶The installation and management of provd
plugins is done via the
page:
The page shows the list of both the installed and installable plugins. You can see if a plugin is installed or not by looking at the Action column.
Here’s the list of other things that can be done from this page:
- update the list of installable plugins, by clicking on the top right icon. On a fresh XiVO installation, this is the first thing to do.
- install a new plugin
- upgrade an installed plugin
- uninstall an installed plugin
- edit an installed plugin, i.e. install/uninstall optional files that are specific to each plugin, like firmware or language files
After installing a new plugin, you are automatically redirected to its edit page. You can then download and install optional files specific to the plugin. You are strongly advised to install firmware and language files for the phones you’ll use although it’s often not a strict requirement for the phones to work correctly.
Warning
If you uninstall a plugin that is used by some of your devices, they will be left in an unconfigured state and won’t be associated to another plugin automatically.
The search box at the top comes in handy when you want to find which plugin to install for your device. For example, if you have a Cisco SPA508G, enter “508” in the search box and you should see there’s 1 plugin compatible with it.
Note
If your device has a number in its model name, you should use only the number as the search keyword since this is what usually gives the best results.
It’s possible there will be more than 1 plugin compatible with a given device. In these cases, the difference between the two plugins is usually just the firmware version the plugins target. If you are unsure about which version you should install, you should look for more information on the vendor website.
It’s good practice to only install the plugins you need and no more.
By default, the list of plugins available for installation are the stable plugins for the officially supported devices.
This can be changed in the URL field to one of the following value:
page, by setting thehttp://provd.xivo.solutions/plugins/1/stable/
– officially supported devices “stable” repository (default)http://provd.xivo.solutions/plugins/1/testing/
– officially supported devices “testing” repositoryhttp://provd.xivo.solutions/plugins/1/archive/
– officially supported devices “archive” repositoryhttp://provd.xivo.solutions/plugins/1/addons/stable/
– community supported devices “stable” repositoryhttp://provd.xivo.solutions/plugins/1/addons/testing/
– community supported devices “testing” repository
The difference between the stable and testing repositories is that the latter might contain plugins that are not working properly or are still in developement.
The archive repository contains plugins that were once in the stable repository.
After setting a new URL, you must refresh the list of installable plugins by clicking the update icon of the
page.If you have set up a DHCP server on XiVO and the phones can access it, you can skip this section.
The according provisioning plugins must be installed.
On the phone, go to
and enter the following settings:- Server type: HTTP
- Server address:
http://<XiVO IP address>:8667/000000000000.cfg
Then save and reboot the phone.
On the web interface of your phone, go to
, and enter the following settings:- Server URL:
http://<XiVO IP address>:8667

Save the changes by clicking on the Confirm button and then click on the Autoprovision Now button.
Once you have installed the proper provd plugins for your devices and setup correctly your DHCP server, you can then connect your devices to your network.
But first, go to
page. You will then see that no devices are currently known by your XiVO:You can then power on your devices on your LAN. For example, after you power on an Aastra 6731i and give it the time to boot and maybe upgrade its firmware, you should then see the phone having its first line configured as ‘autoprov’, and if you refresh the devices page, you should see that your XiVO now knows about your 6731i:
You can then dial from your Aastra 6731i the provisioning code associated to a line of one of your user. You will hear a prompt thanking you and your device should then reboot in the next few seconds. Once the device has rebooted, it will then be properly configured for your user to use it. And also, if you update the device page, you’ll see that the icon next to your device has now passed to green:
To remove a phone from XiVO or enable a device to be used for another user there are two different possibilities :
- click on the
reset to autoprov
button on the web interface

The phone will restarts and display autoprov, ready to be used for another user.
Edit the user associated to the device and put the device field to null.
- click on the
Save
button on the web interface
The phone doesn’t restart and the phone is in autoprov mode in the device list.
You can synchronize the device to reboot it.
Edit the primary user associated to the terminal (one with the line 1) and put the device field to null.
- click on the
Save
button on the web interface
The primary line of the phone has been removed, so the device will lose its funckeys associated to primary user but there others lines associated to the device will stay provisionned.
The phone doesn’t restart and the phone is in autoprov mode in the device list.
You can synchronize the device for reboot it.
- Dial *guest (*48378) on the phone dialpad followed by xivo (9486) as a password
The phone restarts and display autoprov, ready to be used for another user.
If your phones are getting their network configuration from your XiVO’s DHCP server, it’s possible to activate the DHCP integration on the
page.What DHCP integration does is that, on every DHCP request made by one of your
phones, the DHCP server sends information about the request to provd
, which
can then use this information to update its device database.
This feature is useful for phones which lack information in their TFTP/HTTP requests. For example, without DHCP integration, it’s impossible to extract model information for phones from the Cisco 7900 series. Without the model information extracted, there’s chance your device won’t be automatically associated to the best plugin.
This feature can also be useful if your phones are not always getting the same IP addresses, for one reason or another. Again, this is useful only for some phones, like the Cisco 7900; it has no effect for Aastra 6700.
Custom templates comes in handy when you have some really specific configuration to make on your telephony devices.
Templates are handled on a per plugin basis. It’s not possible for a template to be
shared by more than one plugin since it’s a design limitation of the plugin system
of provd
.
Note
When you install a new plugin, templates are not migrated automatically, so you must manually copy them from the old plugin directory to the new one. This does not apply for a plugin upgrade.
Let’s suppose we have installed the xivo-aastra-3.3.1-SP2
plugin and
want to write some custom templates for it.
First thing to do is to go into the directory where the plugin is installed:
cd /var/lib/xivo-provd/plugins/xivo-aastra-3.3.1-SP2
Once you are there, you can see there’s quite a few files and directories:
tree
.
+-- common.py
+-- entry.py
+-- pkgs
| +-- pkgs.db
+-- plugin-info
+-- README
+-- templates
| +-- 6730i.tpl
| +-- 6731i.tpl
| +-- 6739i.tpl
| +-- 6753i.tpl
| +-- 6755i.tpl
| +-- 6757i.tpl
| +-- 9143i.tpl
| +-- 9480i.tpl
| +-- base.tpl
+-- var
+-- cache
+-- installed
+-- templates
+-- tftpboot
+-- Aastra
+-- aastra.cfg
The interesting directories are:
- templates
- This is where the original templates lies. You should not edit these files directly but instead copy the one you want to modify in the var/templates directory.
- var/templates
- This is the directory where you put and edit your custom templates.
- var/tftpboot
- This is where the configuration files lies once they have been generated from the templates. You should look at them to confirm that your custom templates are giving you the result you are expecting.
Warning
When you uninstall a plugin, the plugin directory is removed altogether, including all the custom templates.
A few things to know before writing your first custom template:
- templates use the Jinja2 template engine.
- when doing an
include
or anextend
from a template, the file is first looked up in thevar/templates
directory and then in thetemplates
directory. - device in autoprov mode are affected by templates, because from the point of view
of
provd
, there’s no difference between a device in autoprov mode or fully configured. This means there’s usually no need to modify static files invar/tftpboot
. And this is a bad idea since a plugin upgrade will override these files.
cp templates/base.tpl var/templates
vi var/templates/base.tpl
xivo-provd-cli -c 'devices.using_plugin("xivo-aastra-3.3.1-SP2").reconfigure()'
Once this is done, if you want to synchronize all the affected devices, use the following command:
xivo-provd-cli -c 'devices.using_plugin("xivo-aastra-3.3.1-SP2").synchronize()'
Let’s supose we want to customize the template for our 6739i:
cp templates/6739i.tpl var/templates
vi var/templates/6739i.tpl
xivo-provd-cli -c 'devices.using_plugin("xivo-aastra-3.3.1-SP2").reconfigure()'
To create a custom template for a specific device you have to create a device-specific template
named <device_specific_file_with_extension>.tpl
in the var/templates/
directory :
- for an Aastra phone, if you want to customize the file
00085D2EECFB.cfg
you will have to create a template file named00085D2EECFB.cfg.tpl
, - for a Snom phone, if you want to customize the file
000413470411.xml
you will have to create a template file named000413470411.xml.tpl
, - for a Polycom phone, if you want to customize the file
0004f2211c8b-user.cfg
you will have to create a template file named0004f2211c8b-user.cfg.tpl
, - and so on.
Here, we want to customize the content of a device-specific file named 00085D2EECFB.cfg
,
we need to create a template named 00085D2EECFB.cfg.tpl
:
cp templates/6739i.tpl var/templates/00085D2EECFB.cfg.tpl
vi var/templates/00085D2EECFB.cfg.tpl
xivo-provd-cli -c 'devices.using_mac("00085D2EECFB").reconfigure()'
Note
The choice to use this syntax comes from the fact that provd
supports devices that do not have MAC addresses,
namely softphones.
Also, some devices have more than one file (like Snom), so this way make it possible to customize more than 1 file.
The template to use as the base for a device specific template will vary depending on the need. Typically, the model template will be a good choice, but it might not always be the case.
From time to time, new firmwares are released by the devices manufacturer. This sometimes translate to a new plugin being available for these devices.
When this happens, it almost always means the new plugin obsoletes the older one. The older plugin is then considered “end-of-life”, and won’t receive any new updates nor be available for new installation.
Let’s suppose we have the old xivo-aastra-3.2.2.1136
plugin installed on our
xivo and want to use the newer xivo-aastra-3.3.1-SP2
plugin.
Both these plugins can be installed at the same time, and you can manually change the plugin used by a phone by editing it via the
page.If you are using custom templates in your old plugin, you should copy them to the new plugin and make sure that they are still compatible.
Once you take the decision to migrate all your phones to the new plugin, you can use the following command:
xivo-provd-cli -c 'helpers.mass_update_devices_plugin("xivo-aastra-3.2.2.1136", "xivo-aastra-3.3.1-SP2")'
Or, if you also want to synchronize (i.e. reboot) them at the same time:
xivo-provd-cli -c 'helpers.mass_update_devices_plugin("xivo-aastra-3.2.2.1136", "xivo-aastra-3.3.1-SP2", synchronize=True)'
You can check that all went well by looking at the
page.The provisioning server has partial support for environment where the telephony devices are behind a NAT equipment.
By default, each time the provisioning server receives an HTTP/TFTP request from a device, it makes sure that only one device has the source IP address of the request. This is not a desirable behaviour when the provisioning server is used in a NAT environment, since in this case, it’s normal that more than 1 devices have the same source IP address (from the point of view of the server).
If all your devices used on your XiVO are behind a NAT, you should disable this behaviour by
setting the NAT
option to 1 via the
page.
Enabling the NAT option will also improve the performance of the provisioning server in this scenario.
If you have many devices behind a NAT equipment, you should also check the security section to make sure the IP address of your NAT equipment doesn’t get banned unintentionally.
- You must only have phones of the following brands:
- Aastra
- Cisco SPA
- Yealink
- All your devices must be behind a NAT equipment (the devices may be grouped behind different NAT equipments, not necessarily the same one)
- You must provision the devices via the Web interface, i.e. associate the devices from the user form. Using the 6-digit provisioning code on the phone will produce unexpected results (i.e. the wrong device will be provisioned)
For technical information about why other devices are not supported, you can look at this issue on the XiVO bug tracker.
By design, the auto-provisioning process is vulnerable to:
- Leakage of sensitive information: some files that are served by the provisioning server contains sensitive information, e.g. SIP credentials that are used by SIP phones to make calls. Depending on your network configuration and the amount of information an attacker has on your telephony ecosystem (phone vendor, MAC address, etc.), he could retrieve the content of some files containing sensitive information.
- Denial-of-service attack: in its default configuration, each time the provisioning server identify a request coming from a new device, it creates a new device object in its database. An attacker could spoof requests to the provisioning server to create a huge amount of devices, creating a denial-of-service condition.
That said, starting from XiVO 16.08, XiVO adds Fail2ban support to the
provisioning server to drastically lower the likelihood of such attacks. Every time a request for a
file potentially containing sensitive information is requested, a log line is appended to the
/var/log/xivo-provd-fail2ban.log
file, which is monitored by fail2ban. The same thing
happens when a new device is automatically created by the provisioning server.
The fail2ban configuration for the provisioning server is located at
/etc/fail2ban/jail.d/xivo.conf
. You may want to adjust the findtime
/ maxretry
value
if you have special requirements. In particular, if you have many phones behind a NAT equipment,
you’ll probably have to adjust these values, since every request coming from your phones behind your
NAT will appear to the provisioning server as coming from the same source IP address, and this IP
address will then be more likely to get banned promptly if you, for example, reboot all your phones
at the same time. Another solution would be to add your IP address to the list of ignored IP address
of fail2ban. See the fail2ban(1) man page for more information.
XiVO 16.08 or later is required. You also need to use compatible xivo-provd plugins. Here’s the list of official plugins which are compatible:
Plugin family | Version |
---|---|
xivo-aastra | >= 1.6 |
xivo-cisco-sccp | >= 1.1 |
xivo-cisco-spa | >= 1.0 |
xivo-digium | >= 1.0 |
xivo-polycom | >= 1.7 |
xivo-snom | >= 1.6 |
xivo-yealink | >= 1.26 |
If you have a phone provisioned with XiVO and its one of the supported ones, you’ll be able to search in your XiVO directory and place call directly from your phone.
See the list of supported devices to know if a model supports the XiVO directory or not.
For the remote directory to work on your phones, the first thing to do is to go to the
page.You then have to add the range of IP addresses that will be allowed to access the directory. So if you know that your phone’s IP addresses are all in the 192.168.1.0/24 subnet, just click on the small “+” icon and enter “192.168.1.0/24”, then save.
Once this is done, on your phone, just click on the “remote directory” function key and you’ll be able to do a search in the XiVO directory from it.
Jitsi (http://jitsi.org/) is an opensource softphone (previously SIP Communicator).
XiVO now support Jitsi sofphones provisioning. Here are the steps to follow :
Open XiVO Web interface, and go to Configuration tab, Then chose
, Install the Jitsi plugin you want to use : e.g.:xivo-jitsi-1
You can now launch your Jitsi softphone
Launch Jitsi,
If you don’t have any accounts configured Jitsi will launch a windows and you can click
Use online provisioning. Otherwise go to Tools -> Options -> Advanced -> Provisioing, Click on Enable provisioning
Select Manually specify a provisioning URI,
Enter the folowing URI where <provd_ip> is the VoIP interface IP address of your XiVO and <provd_port> is the provd port (default : 8667)
http://<provd_ip>:<provd_port>/jitsi?uuid=${uuid}
When done, quit Jitsi,
Launch Jitsi again,
- You should now be connected with in autoprov mode,
- You could see a new device in the devices list,
- You can now provision the phones by typing the provisioning code (you get it in the Lines list),
- Quit Jitsi again (configuration syncing is not available with the Jitsi plugin)
- And launch Jitsi again : you should now be connected with you phone account
SCCP Configuration¶
To be able to provision SCCP phones you should :
- activate the DHCP Server,
- activate the DHCP Integration,
- Then install a plugin for SCCP Phone:

Installing xivo cisco-sccp plugin
At this point you should have a fully functional DHCP server that provides IP address to your phones. Depending on what type of CISCO phone you have, you need to install the plugin sccp-legacy, sccp-9.4 or both.
Note
Please refer to the Provisioning page for more information on how to install CISCO firmwares.
Once your plugin is installed, you’ll be able to edit which firmwares and locales you need. If you are unsure, you can choose all without any problem.

Editing the xivo-cisco-sccp-legacy plugin
Now if you connect your first SCCP phone, you should be able to see it in the device list.
- Listing the detected devices:

Device list
When connecting a second SCCP phone, the device will be automatically detected as well.

Device list
The last step is to create a user with a SCCP line.
- Creating a user with a SCCP line:

Add a new user

Edit user informations
Before saving the newly configured user, you need to select the Lines menu and add a SCCP line. Now, you can save your new user.

Add a line to a user
Congratulations ! Your SCCP phone is now ready to be called !
With SCCP phones, the only function keys that can be configured are:
- Key: Only the order is important, not the number
- Type:
Customized
; Any other type doesn’t work - Destination: Any valid extension
- Label: Any label
- Supervision:
Enabled
orDisabled
- SCCP Phones support directmedia (direct RTP). In order for SCCP phones to use directmedia, one must enable the directmedia option in SCCP general settings:
Features | Supported |
---|---|
Receive call | Yes |
Initiate call | Yes |
Hangup call | Yes |
Transfer call | Yes |
Congestion Signal | Yes |
Autoanswer (custom dialplan) | Yes |
Call forward | Yes |
Multi-instance per line | Yes |
Message waiting indication | Yes |
Music on hold | Yes |
Context per line | Yes |
Paging | Yes |
Direct RTP | Yes |
Redial | Yes |
Speed dial | Yes |
BLF (Supervision) | Yes |
Resync device configuration | Yes |
Do not disturb (DND) | Yes |
Group listen | Yes |
Caller ID | Yes |
Connected line ID | Yes |
Group pickup | Yes |
Auto-provisioning | Not yet |
Multi line | Not yet |
Codec selection | Yes |
NAT traversal | Not yet |
Type of Service (TOS) | Manual |
Device type | Supported | Firmware version | Timezone aware |
---|---|---|---|
7905 | Yes | 8.0.3 | No |
7906 | Yes | SCCP11.9-4-2SR1-1 | Yes |
7911 | Yes | SCCP11.9-4-2SR1-1 | Yes |
7912 | Yes | 8.0.4(080108A) | No |
7920 | Yes | 3.0.2 | No |
7921 | Yes | 1.4.5.3 | Yes |
7931 | Yes | SCCP31.9-4-2SR1-1 | Yes |
7937 | Testing | ||
7940 | Yes | 8.1(SR.2) | No |
7941 | Yes | SCCP41.9-4-2SR1-1 | Yes |
7941GE | Yes | SCCP41.9-4-2SR1-1 | Yes |
7942 | Yes | SCCP42.9-4-2SR1-1 | Yes |
7945 | Testing | ||
7960 | Yes | 8.1(SR.2) | No |
7961 | Yes | SCCP41.9-4-2SR1-1 | Yes |
7962 | Yes | SCCP42.9-4-2SR1-1 | Yes |
7965 | Testing | ||
7970 | Testing | ||
7975 | Testing | ||
CIPC | Yes | 2.1.2 | Yes |
Models not listed in the table above won’t be able to connect to Asterisk at all. Models listed as “Testing” are not yet officially supported in XiVO: use them at your own risk.
The “Timezone aware” column indicates if the device supports the timezone tag in its configuration file, i.e. in the file that the device request to the provisioning server when it boots. If you have devices that don’t support the timezone tag and these devices are in a different timezone than the one of the XiVO, you can look at the issue #5161 for a potential solution.
Schedules¶
Schedules are specific time frames that can be defined to open or close a service. Within schedules you may specify opening days and hours or close days and hours.
A default destination as user, group ... can be defined when the schedule is in closed state.
Schedules can be applied to :
- Users
- Groups
- Inbound calls
- Outbound calls
- Queues
A schedule is composed of a name, a timezone, one or more opening hours or days that you may setup using a calendar widget, a destination to be used when the schedule state is closed.
With the calendar widget you may select months, days of month, days of week and opening time.
You may also optionaly select closed hours and destination to be applied when period is inside the main schedule. For example, your main schedule is opened between 08h00 and 18h00, but you are closed between 12h00 and 14h00.
When you have a schedule associated to a user, if this user is called during a closed period, the caller will first hear a prompt saying the call is being transferred before being actually redirected to the closed action of the schedule.
If you don’t want this prompt to be played, you can change the behaviour by:
- editing the
/etc/xivo/asterisk/xivo_globals.conf
file and setting theXIVO_FWD_SCHEDULE_OUT_ISDA
to1
- reloading the asterisk dialplan with an
asterisk -rx "dialplan reload"
.
Sound Files¶
On a fresh install, only en_US and fr_FR sounds are installed. Canadian French and German are available too.
To install Canadian French sounds you have to execute the following command:
apt-get install asterisk-sounds-wav-fr-ca xivo-sounds-fr-ca
To install German sounds you have to execute the following command:
apt-get install asterisk-sounds-wav-de-de xivo-sounds-de-de
Now you may select the newly installed language for your users.
Asterisk will read natively WAV files encoded in wav 8kHz, 16 bits, mono.
The following command will return the encoding format of the <file>
$ file <file>
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
The following command will re-encode the <input file> with the correct parameters for asterisk and write into the <output file>:
sox <input file> -b 16 -c 1 -t wav <output file> rate -I 8000
Switchboard¶
This page describes the configuration needed to have a switchboard on your XiVO.
Switchboard functionality is available in the XiVO client. The goal of this page is to explain how to configure your switchboard and how to use it.
The switchboard xlet and profile allow an operator to view incoming calls, answer them, put calls on hold, view the calls on hold and pick up the calls on hold.
Note
The shortcut keys of the switchboard do not work on the Mac version of the XiVO client.
Note
The enter shortcut to answer a call will not work if the focus is currently on a widget that will consume the key press. ie: a text field, a drop down
Note
Attended transfers to the switchboard cannot be managed with the switchboard xlets depending on the moment at which the call was completed.
Be sure to read the limitations before configuring a switchboard.
In order to configure a switchboard on your XiVO, you need to:
- Create a queue for your switchboard
- Create a queue for your switchboard’s calls on hold
- Create the users that will be operators
- Activate the switchboard option for your phone
- Create an agent for your user
- Assign the incoming calls to the switchboard queue
- For each operator, add a function key for logging in or logging out from the switchboard queue.
- Set “no answer” destinations on the switchboard queue
The supported phones for the switchboard are:
Brand | Model | XiVO version | Plugin version |
---|---|---|---|
Aastra | 6755i | >= 14.07 | >= xivo-aastra-3.3.1-SP2, v1.0 |
Aastra | 6757i | >= 14.07 | >= xivo-aastra-3.3.1-SP2, v1.0 |
Aastra | 6735i | >= 14.07 | >= xivo-aastra-3.3.1-SP2, v1.2 |
Aastra | 6737i | >= 14.07 | >= xivo-aastra-3.3.1-SP2, v1.2 |
Polycom | VVX 400 | >= 15.11 | >= xivo-polycom-5.3.0, v1.3 |
Polycom | VVX 410 | >= 15.11 | >= xivo-polycom-5.3.0, v1.3 |
Snom | 720 | >= 14.14 | >= xivo-snom-8.7.3.25.5, v1.0 |
Snom | D725 | >= 14.14 | >= xivo-snom-8.7.5.17, v1.4 |
Yealink | T46G | >= 15.01 | >= xivo-yealink-72.0, v1.22.1 |
All calls to the switchboard will first be distributed to a switchboard queue.
To create this queue, go to
and click the add button.
The following configuration is mandatory
- The field has to be __switchboard
- The field has to be Ring all
- The field has to be xivo_subr_switchboard
- The option has to be enabled
- The option has to be enabled
- The option has to be disabled
- The option has to be 1 second
- The option has to be disabled
- The option has to be disabled
- The option has to be No
Other important fields
- The field is the name displayed in the XiVO client xlets and in the statistics
- The field is the number that will be used to reach the switchboard internally (typically 9)
The switchboard uses a queue to track its calls on hold.
To create this queue, go to
and click the add button.The following configuration is mandatory
- The field has to be __switchboard_hold
- The field has to be a valid number in a context reachable by the switchboard
Other important fields
- The field is the name displayed in the XiVO client xlets and in the statistics
Warning
This queue MUST have NO members
Each operator needs to have a user configured with a line. The XiVO client profile has to be set to Switchboard.
The following configuration is mandatory for switchboard users
- The field has to be set
- The option has to be enabled
- The field has to be set
- The field has to be set
- The field has to be set to Switchboard
- The field has to have a valid extension
- The supported device field has to be a
- The option has to be enabled
- The option has to be enabled

The switchboard option must be activated on the phone. It’s possible to activate this option only on supported phones and plugins.
- Edit device associated to your user in
- Check the switchboard checkbox and save
- Synchronize your phone to apply the changes

To be able to use a Polycom phone for the switchboard, the XiVO must be able to do HTTP requests to the phone. This might be problematic if there’s a NAT between your XiVO and your phone.
It’s possible to configure the Polycom switchboard via the configuration files of xivo-ctid. The following options are available:
switchboard_polycom:
username: xivo_switchboard
password: xivo_switchboard
answer_delay: 0.5
You will also need to change the XML API username/password by creating a custom template for your phone.
When using a Snom switchboard, you must not configure a function key on position 1.
To be able to use a Snom phone for the switchboard, the XiVO must be able to do HTTP requests to
the phone. This might be problematic if there’s a NAT between your XiVO and your phone. The
following command should work from your XiVO’s bash command line wget http://guest:guest@<phone IP
address>/command.htm?key=SPEAKER
. If this command does not activate the phone’s speaker, your
network configuration will have to be fixed before you can use the Snom switchboard.
It’s possible to configure the Snom switchboard via the configuration files of xivo-ctid. The following options are available:
switchboard_snom:
username: guest
password: guest
answer_delay: 0.5
You have to change the username and password option if you have changed the administrator username or administrator password for your phone in
.When using a Yealink switchboard, you must not configure a function key on position 1.
Each operator needs to have an associated agent.
Warning
Each agent MUST ONLY be a member of the Switchboard queue
To create an agent:
- Go to
- Click on the group default
- Click on the Add button

- Associate the user to the agent in the Users tab

- Assign the Agent to the Switchboard Queue (and ONLY to the Switchboard queue)

Incoming calls must be sent to the Switchboard queue to be distributed to the operators. To do this, we have to change the destination of our incoming call for the switchboard queue.
In this example, we associate our incoming call (DID 444) to our Switchboard queue:

When there are no operators available to answer a call, “No Answer” destinations should be used to redirect calls towards another destination.
You also need to set the timeout of the Switchboard queue to know when calls will be redirected.

The reachability timeout must not be disabled nor be too short.
The time before retrying a call to a member should be as low as possible (1 second).

In this example we redirect “No Answer”, “Busy” and “Congestion” calls to the everyone group and “Fail” calls to the guardian user.
You can also choose to redirect all the calls to another user or a voice mail.

The transfer destination is chosen in the Directory xlet. You must follow the Directory Xlet section to be able to use it.
The above documentation can be used for multiple switchboards on the same XiVO by replacing the __switchboard and __switchboard_hold queues name and configuring the operators XiVO client accordingly in the
window.
All switchboard queues should be added to the xivo-ctid configuration. New
queues can be added by adding a file in /etc/xivo-ctid/conf.d
. For
example, the following content should be used for a new switchboard queue
names __switchboard_two and an hold queue names __switchboard_hold_two.
{"switchboard_queues": {"__switchboard_two": true},
"switchboard_hold_queues": {"__switchboard_hold_two": true}}
Warning
The switchboard configuration must be completed before using the switchboard. This includes :
- Device, User, Agent and Queues configuration (see above),
- Directory xlet configuration (see Directory Xlet)
If it’s not the case, the user must disconnect his XiVO client and reconnect.
Be sure to read the limitations before using the switchboard.
When the user connects with his XiVO Client, he gets the Switchboard profile.

- Current Call frame
- Answer button
- Call button
- Blind transfer button
- Attended transfer button
- Hold button
- Hangup button
- Incoming Calls list
- Waiting Calls list
- Directory Xlet
- Dial Xlet
Note
If you don’t see the Switchboard Xlet, right-click on the grey bar at the right of the Help menu and check Switchboard:

The operator can login his agent using a function key or an extension to start receiving calls.
When the switchboard receives a call, the new call is added to the Incoming Calls list on the left and the phone starts ringing. The user can answer this call by:
- clicking on any call in the list
- clicking the Answer button
- pressing the Enter key
Note
The XiVO Client must be the active window for the keyboard shortcuts to be handled
The operator can select which call to answer by:
- clicking directly on the incoming call
- pressing F6 to select the incoming calls frame and pressing the up and down arrow keys
Selecting a call to answer while talking will not answer the call.
Once the call has been answered, it is removed from the incoming calls list and displayed in the Current Call frame.
The switchboard operator can do the following operations:
- Press the Call button or press F3
- Search for the call destination in the directory xlet
- Press to confirm the selection and start the call
The switchboard operator can hang up its current call by either:
- Clicking the Hangup button
- Pressing the F8 key
If the operator has placed a new call via the Directory or Dial xlet and that call has not yet been answered, he can cancel it in the same way.
Once the call has been answered and placed in the current call frame, the operator has 3 choices:
- transfer the call to another user
- using the Blind transfer button or the F4 key.
- using the Attended transfer button or the F5 key
- put the call on hold using the Hold button or the F7 key
- end the call using the Hangup button or the F8 key.
Transfer buttons allow the operator to select towards which destination he wishes to transfer the call. This is made through the Directory xlet. For defails about the xlet Directory usage and configuration see Directory Xlet.
Once the destination name has been entered, press Enter. If multiple destinations are displayed, you can choose by:
- double-clicking on the destination
- using Up/Down arrows then:
- pressing Enter
- pressing the transfer button again
Blind transfers are straightforward: once the call is transferred, the operator is free to manage other calls.
Attended transfers are a bit more complicated: the operator needs to wait for the transfer destination to answer before completing the transfer.
In this example, the operator is currently asking Bernard Marx if he can transfer Alice Wonderland to him.

- Complete transfer button
- Cancel transfer button
- Transfer destination filtering field (xlet Directory)
- Transfer destination list (xlet Directory)
Once the destination has answered, you can:
- cancel the transfer with F8 key
- complete the transfer with F5 key
Note
The operator can not complete an attended transfer while the transfer destination is ringing. In this case, the operator must cancel the attended transfer and use the Blind transfer action.
If the user places the call on hold, it will be removed from the Current call frame and displayed in the Waiting calls list. The time counter shows how long the call has been waiting, thus it will be reset each time the call returns in the Waiting calls list. The calls are ordered from the oldest to the newest.
Once a call has been placed on hold, the operator will most certainly want to retrieve that call later to distribute it to another destination.
To retrieve a call on hold:
- click the desired call in the Waiting calls list
- with the keyboard:
- move the focus to the Waiting calls list (F9 key)
- choose the desired call with the arrow keys
- press the Enter key.
Once a call has been retrieved from the Waiting calls list, it is moved back into the Current Call frame, ready to be distributed.
Note
Statistics are produced by xivo-ctid. If a call is received when xivo-ctid is stopped, no statistics will be produced for that call.
Note
Statistics are only generated for calls answered in XiVO ≥ 16.03.
Note
Statistics are only available for existing switchboard queues, i.e. deleting a queue will also delete the associated statistics.
Switchboard statistics can be retrieved in CSV format via the web interface in
.
- Start date: when empty, the result will contain statistics from the beginning
- End date: when empty, the result will contain statistics until the current time
Note
Switchboard statistics older than a year are automatically removed. See Purge Logs for more details.
The generated CSV report includes the following columns:
- date: The date at which the calls were received
- entered: The number of calls to the switchboard for the given date excluding calls when the switchboard was closed (e.g. with a schedule)
- answered: The number of calls that have been answered by the operator and then transferred or completed by the operator
- transferred: The number of calls that have been transferred by the switchboard operator to another destination
- abandoned: The number of calls that have been abandoned in the switchboard queue or while waiting in the hold queue
- forwarded: The number of calls that have been forwarded to another destination:
- a call reaching a full queue
- a call waiting until the max ring time is reached
- a call forwarded because of a diversion rule
- a call forwarded because of a leave empty condition
- waiting_time_average: The average time spent in the switchboard and hold queue for all calls that entered the switchboard
Switchboard statistic events are published on the bus to be consumed by collectd.
In order to process these events, you need:
collectd installed on your XiVO:
apt-get install collectd
In
/etc/collectd/collectd.conf.d/amqp.conf
, configure collectd to read events from the bus (RabbitMQ):LoadPlugin amqp <Plugin "amqp"> <Subscribe "xivo"> Host "127.0.0.1" Port "5672" VHost "/" User "guest" Password "guest" Exchange "collectd" ExchangeType "topic" RoutingKey "collectd.#" </Subscribe> </Plugin>
another service receiving events from collectd, e.g. logstash, graphite, another collectd.
The collectd events have the following attributes:
- host: the UUID of the XiVO.
- plugin:
switchboard
- plugin_instance: the name (not the display name) of the queue for incoming calls of the switchboard.
- type:
counter
orgauge
. - type_instance: the following values.
- entered
- This event is produced when a call enters the switchboard on an open schedule. Calls that did not enter the queue, if the queue was full for example, will also generate an entered event.
- abandoned
- This event is produced when the called hangs up while waiting in the incoming queue or in the hold queue.
- transferred
- This event is produced when a call is transferred from the switchboard by the operator. For attended transfers, the event is sent when the transfer is completed.
- forwarded
This event is produced when a call is redirected to another destination under certain conditions. This include:
- When the queue is full
- When the queue timeout is reached
- When no agent are logged with a join empty configuration
- When a divertion occured
- completed
- This event is produced when a call was answered by the operator without being transferred to another destination.
- wait_time
- This event is produced when a call is completed, its value is the sum of all times spent in the hold queue and the time spent in the incoming queue before being answered.
Users¶
Users Configuration.
Users can be imported and associated to other resources by use of a CSV file. CSV Importation can be used in situations where you need to modify many users at the same in an efficient manner, or for migrating users from one system to another. A CSV file can be created and edited by spreadsheet tools such as Excel, LibreOffice/OpenOffice Calc, etc.
The first line of a CSV file contains a list of field names (also sometimes called “columns”). Each new line afterwards are users to import. CSV data must respect the following conditions:
- Files must be encoded in UTF-8
- Fields must be separated with a
,
- Fields can be optionally quoted with a
"
- Double-quotes can be escaped by writing them twice (e.g.
Robert ""Bob"" Jenkins
) - Empty fields or headers that are not defined will be considered null.
- Fields of type bool must be either
0
for false, or1
for true. - Fields of type int must be a positive number
In the following tables, columns have been grouped according to their resource. Each resource is created and associated to its user when all required fields for that resource are present.
Field | Type | Required | Values | Description |
---|---|---|---|---|
entity_id | int | Yes | Entity ID (Defined in menu | )|
firstname | string | Yes | User’s firstname | |
lastname | string | User’s lastname | ||
string | User’s email | |||
language | string | de_DE, en_US, es_ES, fr_FR, fr_CA | User’s language | |
mobile_phone_number | string | Mobile phone number | ||
outgoing_caller_id | string | Customize outgoing caller id for this user | ||
enabled | bool | Enable/Disable the user | ||
supervision_enabled | bool | Enable/Disable supervision | ||
call_transfer_enabled | bool | Enable/Disable call transfers by DTMF | ||
dtmf_hangup_enabled | bool | Enable/Disable hangup by DTMF | ||
simultaneous_calls | int | Number of calls a user can have on his phone simultaneously | ||
ring_seconds | int | Must be a multiple of 5 | Number of seconds a call will ring before ending | |
call_permission_password | string | Overwrite all passwords set in call permissions associated to the user |
Field | Type | Required | Values | Description |
---|---|---|---|---|
cti_profile_enabled | bool | No | Activates the XiVO Client account for this user | |
username | string | Yes, if profile enabled | XiVO Client username | |
password | string | Yes, if profile enabled | XiVO Client password | |
cti_profile_name | string | Yes, if profile enabled | XiVO Client profile (Defined in menu | )
Field | Type | Required | Values | Description |
---|---|---|---|---|
exten | string | Yes | Number for calling the user. The number must be inside the range of acceptable numbers defined for the context | |
context | string | Yes | Context | |
line_protocol | string | Yes | sip, sccp | Line protocol |
sip_username | string | SIP username | ||
sip_secret | string | SIP secret |
Field | Type | Required | Values | Description |
---|---|---|---|---|
incall_exten | string | Yes | Number for calling the user from an incoming call (i.e outside of XiVO). The number must be inside the range of acceptable numbers defined for the context. | |
incall_context | string | Yes | context used for calls coming from outside of XiVO | |
incall_ring_seconds | int | Number of seconds a call will ring before ending |
Field | Type | Required | Values | Description |
---|---|---|---|---|
voicemail_name | string | Yes | Voicemail name | |
voicemail_number | string | Yes | Voicemail number | |
voicemail_context | string | Yes | Voicemail context | |
voicemail_password | string | A sequence of digits or # | Voicemail password | |
voicemail_email | string | Email for sending notifications of new messages | ||
voicemail_attach_audio | bool | Enable/Disable attaching audio files to email message | ||
voicemail_delete_messages | bool | Enable/Disable deleting message after notification is sent | ||
voicemail_ask_password | bool | Enable/Disable password checking |
Field | Type | Required | Values | Description |
---|---|---|---|---|
call_permissions | string | list separated by semicolons (; ) |
Names of the call permissions to assign to the user |
Once your file is ready, you can import it via
. At the top of the page there is a plus button. A submenu will appear when the mouse is on top. Click on Import a file.The following example defines 3 users who each have a phone number. The first 2 users have a SIP line, where as the last one uses SCCP:
entity_id,firstname,lastname,exten,context,line_protocol
1,John,Doe,1000,default,sip
1,George,Clinton,1001,default,sip
1,Bill,Bush,1002,default,sccp
The following example imports a user with a phone number and a voicemail:
entity_id,firstname,lastname,exten,context,line_protocol,voicemail_name,voicemail_number,voicemail_context
1,John,Doe,1000,default,sip,Voicemail for John Doe,1000,default
The following exmple imports a user with both an internal and external phone number (e.g. incoming call):
entity_id,firstname,lastname,exten,context,line_protocol,incall_exten,incall_context
1,John,Doe,1000,default,sip,2050,from-extern
The field list for an update is the same as for an import with the addition of the column uuid, which is mandatory. For each line in the CSV file, the updater goes through the following steps:
- Find the user, using the uuid
- For each resource (line, voicemail, extension, etc) find out if it already exists.
- If an existing resource was found, associate it with the user. Otherwise, create it.
- Update all remaining fields
The following restrictions must also be respected during update:
- Columns that are not included in the CSV header will not be updated.
- A field that is empty (i.e, “”) will be converted to NULL, which will unset the value.
- A line’s protocol cannot be changed (i.e you cannot go from “sip” to “sccp” or vice-versa).
- An incall cannot be updated if the user has more than one incall associated.
Updating is done through the same menu as importing (
). A submenu will appear when the mouse is on top. Click on Update from file in the submenu.CSV exports can be used as a scaffold for updating users, or as a means of importing users into another system. An export will generate a CSV file with the same list of columns as an import, with the addition of uuid and provisioning_code.
Exports are done through the same menu as importing (
). Click on Export to CSV in the submenu. You will be asked to download a file.Function keys can be configured to customize the user’s phone keys. Key types are pre-defined and can be browsed through the Type drop-down list. The Supervision field allows the key to be supervised. A supervised key will light up when enabled. In most cases, a user cannot add multiple times exactly the same function key (example : two user function keys pointing to the same user). Adding the same function key multiple times can lead to undefined behavior and generally will delete one of the two function keys.
Warning
SCCP device only supports type “Customized”.

For User keys, start to key in the user name in destination, XiVO will try to complete with the corresponding user.
If the forward unconditional function key is used with no destination the user will be prompted when the user presses the function key and the BLF will monitor ALL unconditional forward for this user.
To enable online call recording, you must check the “Enable online call recording” box in the user form.

Users Services
When this option is activated, the user can press *3
during a conversation to start/stop online
call recording. The recorded file will be available in the monitor
directory of the
menu.
You can enable/disable the recording of all calls for a user in 2 different way:
- By checking the “Call recording” box of the user form.

Users Services
- By using the extension *26 from your phone (the “call recording” option must be activated in ).
When this option is activated, all calls made to or made by the user will be recorded in the monitor
directory of the menu.
Voicemail¶
Voicemail Configuration.
The global voicemail configuration is located under
.There are 2 ways to add a voicemail:
New voicemails can be added using the +
button.
Once your voicemail is configured, you have to edit the user configuration and search the voicemail previously created and then associate it to your user.
The other way is to add the voicemail from user’s configuration in the ‘voicemail’ tab by
- Clicking the
+
button - Filling the voicemail form
- Saving
Note
The user’s language must be set in the general tab
You can disable a user’s voicemail by un-checking the ‘Enable voicemail’ option on the Voicemail tab from user’s configuration.
Delete voicemail is done on
or from the user’s voicemail tab.Note
- Deleting a voicemail is irreversible. It deletes all messages associated with that voicemail.
- If the voicemail contains messages, the message waiting indication on the phone will not be deactivated until the next phone reboot.
Unchecking the option Ask password
allows you to skip password checking for the voicemail only
when it is consulted from an internal context.
- when calling the voicemail with *98
- when calling the voicemail with *99<voicemail number>
Warning
If the the *99 extension is enabled and a user does not have a password on its voicemail, anyone from the same context will be able to listen to its messages, change its password and greeting messages.
However, the password will be asked when the voicemail is consulted through an incoming call. For instance, let’s consider the following incoming call:
With such a configuration, when calling this incoming call from the outside, we will be asked for:
- the voicemail number we want to consult
- the voicemail password, even if the “Disable password checking option” is activated
And then, we will be granted access to the voicemail.
Take note that the second “context” field contains the context of the voicemail. Voicemails of other contexts will not be accessible through this incoming call.
Warning
For security reasons, such an incoming call should be avoided if a voicemail in the given context has no password.
If xivo-confd is on a remote host, xivo-confd-client configuration will be required to be able to change the voicemail passwords using a phone.
This configuration should be done:
mkdir -p /etc/systemd/system/asterisk.service.d
cat >/etc/systemd/system/asterisk.service.d/remote-confd-voicemail.conf <<EOF
[Service]
Environment=CONFD_HOST=localhost
Environment=CONFD_PORT=9486
Environment=CONFD_HTTPS=true
Environment=CONFD_USERNAME=<username>
Environment=CONFD_PASSWORD=<password>
EOF
systemctl daemon-reload
WebRTC¶
Note
added in version 2016.04
From version 2016.04 one can use WebRTC with XiVO PBX and XiVO CC in the following environment:
- LAN network (currently no support for WAN environment),
- with the:
- Web Assistant with Chrome browser version 55 (tested on 55.0.2883.87 m 64-bit),
- or Desktop Assistant
See WebRTC Requirements.
Note
Current WebRTC implementation requires to create user with one line configured for WebRTC. To have user with both SIP and WebRTC line is not supported.
- Create user
- Add line to user without any device
- Edit the line created and, in the Advanced tab, add webrtc=yes options:
For the records
Note
This is the manual way to configure a WebRTC line. It is here for the record. You should follow the Configuration of user with WebRTC line instead.
- Create user
- Optional: set codec to ulaw
- Add line to user without any device
- Configure Advanced Line options, so that it is usable with the softphone WebRTC
avpf = yes
call-limit = 1
dtlsenable = yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify = no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey = /etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup = actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
encryption = yes
force_avp = yes
icesupport = yes
transport = ws
Web Services Access¶
You may configure Web Services / REST API permissions in
.Web services access may have two different meanings:
- Who may access REST APIs of various XiVO daemons, and which resources in those REST APIs?
- Who may access PHP web services under
https://xivo.example.com/xivo/configuration/json.php/*
?
Those REST API interfaces are documented on http://api.xivo.io. They all require an authorization token, obtained by giving valid credentials to the REST API of xivo-auth. The relevant settings are:
- Login/Password: the xivo-auth credentials (for the xivo-auth backend
xivo_service
) - ACL: The list of authorized REST API resources. See REST API Permissions.
Unlike PHP web services, there is no host-based authorization, so the Host
setting is not
relevant.
A few REST API access are automatically generated during the installation of XiVO, so that XiVO services may authenticate each other.
You will probably only need to create such a REST API access when you want another non-XiVO service to communicate with XiVO via REST API.
Warning
DEPRECATED
Those web services are deprecated. There is no documentation about their usage, and the goal is to remove them.
They are still protected with HTTP authentication, requiring a login and password. The relevant settings are:
- Login/Password: the HTTP authentication credentials
- Host: the authorized hosts that are allowed to make HTTP requests:
- Empty value: HTTP authentication
- Non-empty value: no HTTP authentication, all requests coming from this host will be accepted. Valid hosts may be: a hostname, an IP address, a CIDR block.
There is no fine-grained permissions: either the user has access to every PHP web services, or none.
Contact Center¶
In XiVO, the contact center is implemented to fulfill the following objectives :
Call routing
Includes basic call distribution using call queues and skills-based routing
Agent and Supervisor workstation.
Provides the ability to execute contact center actions such as: agent login, agent logout and to receive real time statistics regarding contact center status
Statistics reporting
Provides contact center management reporting on contact center activities
Advanced functionalities
Call recording
Screen Pop-up
Agents¶
A call center agent is the person who handles incoming or outgoing customer calls for a business. A call center agent might handle account inquiries, customer complaints or support issues. Other names for a call center agent include customer service representative (CSR), telephone sales or service representative (TSR), attendant, associate, operator, account executive or team member.
—SearchCRM
In this respect, agents in XiVO have no fixed line and can login from any registered device.
- Create a user with a SIP line and a provisioned device.
- Create agents.
- Create a queue adding created agent as member of queue.
These settings are specific for a given agent.
These settings are specific for a given agent.
These settings are specific for a given agent.
These settings are specific for a given agent.
These settings are global for all agents.
Queues¶
Call queues are used to distribute calls to the agents subscribed to the queue. Queues are managed on the
page.A queue can be configured with the following options:
- Name: used as an unique id, cannot be
general
- Display name: Displayed on the supervisor screen
- On-Hold music: The music the caller will hear. The music is played when waiting and when the call is on hold.
A ring strategy defines how queue members are called when a call enters the queue. A queue can use one of the following ring strategies:
- Linear: For each call, in the same order, starting from the same member
- For agents: In login order
- For static members: In definition order
- Least recent: call the member who least recently hung up a call
- Fewest calls: call the member with the fewest completed calls
- Round robin memory: call the “next” member after the one who answered last
- Random: call a member at random
- Weight random: same as random, but taking the member penalty into account
- Ring all: call all members at the same time
Warning
When editing a queue, you can’t change the ring strategy to linear. This is due to an asterisk limitation. Unfortunately, if you want to change the ring strategy of a queue to linear, you’ll have to delete it first and then create a new queue with the right strategy.
Note
When an agent is a member of many queues the order of call distribution between multiple queues is nondeterministic and cannot be configured.
You may control how long a call will stay in a queue using different timers:
- Member reachabillity time out (Advanced tab): Maximum number of seconds a call will ring on an agent’s phone. If a call is not answered within this time, the call will be forwarded to another agent.
- Time before retrying a call to a member (Advanced tab): Used once a call has reached the “Member reachability time out”. The call will be put on hold for the number of seconds alloted before being redirected to another agent.
- Ringing time (Application tab): The total time the call will stay in the queue.
- Timeout priority (Application tab): Determines which timeout to use before ending a call. When set to “configuration”, the call will use the “Member reachability time out”. When set to “dialplan”, the call will use the “Ringing time”.
Calls can be diverted on no answer:
- No answer: The call reached the “Ringing time” in Application tab and no agent answered the call
- Congestion: The number of calls waiting has reached the “Maximum number of people allowed to wait” limit specified on the advanced tab
- Fail: No agent was available to answer the call when the call entered the queue (“Join an empty queue” condition on the advanced tab) or the call was queued and no agents were available to answer (“Remove callers if there are no agents” on the advanced tab)
Diversions can be used to redirect calls to another destination when a queue is very busy. Calls are redirected using one of the two following scenarios:
The diversion check is done only once per call, before the preprocess subroutine is executed and before the call enters the queue.
In the following sections, a waiting call is a call that has entered the queue but has not yet been answered by a queue member.
When this scenario is used, the administrator can set a destination for calls to be sent to when the estimated waiting time is over the threshold.
Note that if a new call arrives when there are no waiting calls in the queue, the call will always be allowed to enter the queue.
Note
- this estimated waiting time is computed from the actual hold time of all answered calls in the queue (since last asterisk restart) according to an exponential smoothing formula
- the estimated waiting time of a queue is updated only when a queue member answers a call.
When this scenario is used, the administrator can set a destination for calls to be sent to when the number of waiting calls per logged-in agent is over the threshold.
The number of waiting calls includes the call for which the check is currently being performed.
The number of logged-in agents is the sum of user members and currently logged-in agent members. An agent only needs to be logged in and a member of the queue to participate towards the count of logged-in agents, regardless of whether he is available, on call, on pause or on wrapup.
The maximum number of waiting calls per logged-in agent can have a fractional part.
Here are a few examples:
Maximum number of waiting calls per logged-in agent: 1
Current number of waiting calls: 2
Current number of logged-in agents: 2
Number of waiting calls per logged-in agent when a new call arrives: 3 / 2 = 1.5
Call will be redirected
Maximum number of waiting calls per logged-in agent: 0.5
Number of waiting calls: 5
Number of logged-in agents: 12
Number of waiting calls per logged-in agent when a new call arrives: 6 / 12 = 0.5
Call will not be redirected
Note that if a new call arrives when there are no waiting calls in the queue, the call will always be allowed to enter the queue. For example, in the following scenario:
Maximum number of waiting calls per logged-in agent: 0.5
Current number of waiting calls: 0
Current number of logged-in agents: 1
Number of waiting calls per logged-in agent when a new call arrives: 1 / 1 = 1
Even if the number of waiting calls per logged-in agent (1) is greater than the maximum (0.5), the call will still be accepted since there are currently no waiting calls.
Supervision¶
Allows a contact center supervisor to monitor contact center activities such as:
- Monitoring real time information from call queues
- Agent activities per call queues
- Agent detailed activities
A supervisor profile defined in
menu usually contains the following Xlets :- Identity
- Queues
- Queue members
- Queues (entries detail)
- Agents (list)
- Agents (detail)
Note: | You may also see the Agent Status Dashboard |
---|
- Clicking on a queue’s name in the queue list will display the agent list in the xlet Queue Members and show waiting calls in the Calls of a Queue xlet.
- Clicking on an agent’s name in the agent list will display information on the agent in the Agent Details xlet
- Clicking on the + icon in the Agent Details xlet will display information about the selected queue in the Calls of a Queue and Queue Members xlets.
General information
The queue list is a dashboard displaying queue statistics and real-time counters for each queue configured on the XiVO.
Real-time Columns
The data of following columns display real-time information.
- Queues
- queue name and number if configured to be displayed
- Waiting calls
- The number of calls currently waiting for an agent in this queue. The background color can change depending of the configured thresholds
- EWT
- Estimated waiting time
- Longest wait
- The longest waiting time for currently waiting calls. The background color can change depending of the configured thresholds
- Talking
- The number of agents currently in conversation in the queue. This column is set to 0 when the queue has just been created and no members have been added.
- Logged
- The number of logged agents in the queue. This column is set to “N/A” when the queue has just been created and no members have been added.
- Available
- The number of available agents ready to take a call in the queue. This column is set to N/A when the queue has just been created and no members have been added.
Last Period Columns
The data of following columns are based on statistics fetched from a fixed-width window of time, e.g. the last 60 minutes or the last 10 minutes. See below to configure the width of the window for each queue.
- Received
- The number of calls received in this queue during the configured statistical window
- Answered
- The number of calls answered in this queue during the configured statistical window
- Abandoned
- The number of calls abandoned in this queue during the configured statistical window
- Mean waiting time
- The mean wait time in the statistical time window, in mm:ss If no calls are received, “-” is displayed
- Max waiting time
- The longest wait time in the statistical time window, in mm:ss If no calls are received, “-” is displayed
- Efficiency
- Answered calls over received calls during the configured statistical window (unanswered calls that are still waiting are not taken into account). If no calls are received, “-” is displayed
- QOS
- Percentage of calls taken within X seconds over answered calls during the configured statistical window. If no calls are received, “-” is displayed
Counter availability
When the XiVO client is started, “na” is diplayed for counters that have not been initialized.
When the XiVO client is restarted, the counters are always displayed and calculated as if the application was not restarted. When the server is restarted, counters are reinitialized.
Enabling the xlet
The xlet can be added to any CTI profile from the web interface.
Configuration
Some values can be configured for the xlet. The statistic fetch timer can be set in the CTI profile preferences. This option is expressed in seconds and the default is 30 seconds.
The statistical period can be configured through the XiVO client once logged in by right-clicking on the Queue’s name in the Queues xlet. For each queue, you can configure the following information:
- Qos: maximum wait time for a call, in seconds.
- Window: period of time used for accumulating statistics, in seconds.
The data used to compute statistics on the XiVO server is only kept for a maximum of 3 hours. The window period cannot be configured to go beyond this limit.
Display options can also be set on the client side. A threshold can be configured to change the color of a column using the following parameters:
- Queue thresholds (waiting calls): number of waiting calls in the queue.
- Display queue’s longest wait: Add a column displaying the number of seconds the longest call has waited.
- Queue thresholds (longest wait): number of seconds for the longest waiting call in the queue.
- Display queue number: Add a column displaying the queue’s number.
Monitoring queues on high dimension screens
You may want to display the queue list on one big screen, visible by multiple people. However, the default font will not be large enough, so the information will not be readable.
You can change the font size of this Xlet by giving a configuration file when launching the XiVO Client:
> xivoclient -stylesheet big_fonts.qss # Windows and Mac
$ xivoclient -- -stylesheet big_fonts.qss # GNU/Linux
The big_fonts.qss
file should contain:
QueuesView {font-size: 40px;}
QueuesView QHeaderView {font-size: 40px;}
Units of size that can be used are described on the Qt documentation.
General information
The queue list is a dashboard displaying each agent configured on the XiVO.
Columns
- Number
- The agent’s number
- First name & Last name
- The agent’s first name and last name
- Listen
A clickable cell to listen to the agent’s current call.
Clicking on the cell will make your phone ring. When you’ll answer, you’ll hear the conversation the agent is having.
You’ll then be able to press the following digits on your phone to switch between the different “listen” modes:
- 4 - spy mode (default). No one hears you.
- 5 - whisper mode. Only the agent hears you.
- 6 - barge mode. Both the agent and the person he’s talking to hear you.
- Status since
Shows the agent’s status and the time spent in this status. An agent can have three statuses:
- Not in use when he is ready to answer an ACD call
- Out of queue when he called or answered a call not from the queue
- In use when he is either on call from a queue, on pause or on wrapup
- Logged
- A clickable cell to log or unlog the agent
- Joined queues
- The number of queues the agent will be receiving calls from
- Paused
- A clickable cell to pause or unpause the agent
- Paused queues
- The number of queues in which the agent is paused
General information
Display advanced informations of an agent and enable to login/logoff, add/remove to a queue, and pause/unpause.

Agent Details
- This is the status information of agent
- Button to login/logoff agent
- Supervision button of the Xlet “Calls of a queue”
- Add/Remove agent for given queue
- Pause/Unpause button for given queue
The queue members lists which agents or phones will receive calls from the selected queue and some of their attributes.

Columns
- Number
- The agent number or the phone number of the queue member.
- Firstname and Lastname
- First name and last name of the agent or the user to which the phone belongs.
- Logged
- Whether the agent is logged or not. Blank for a phone.
- Paused
- Whether the agent is paused or not. Blank for a phone.
- Answered calls
- Number of calls answered by the member since last login (for an agent), or restart or configuration reload.
- Last call
- Hangup time of the last answered calls.
- Penalty
- Penalty of the queue member.
You can configure XiVO to have the following scenario:
- The agent person leaves temporarily his office (lunch, break, ...)
- He sets his presence in the XiVO Client to the according state
- The agent will be automatically set in pause and his phone will not ring from queues
- He comes back to his office and set his presence to ‘Available’
- The pause will be automatically cancelled
You can configure the presence states of CTI profiles
and attach Actions
to them, such as Set in pause or Enable DND.
You can then attach an action Set in pause for multiple presence states and attach an action Cancel the pause for the presence state Available.
For now, the actions attached to the mandatory presence Disconnected will not be taken into account.
Agent Status Dashboard¶
The goal of the agent status dashboard xlet is to give contact center supervisors a better overview of agent status evolution in active queues.

The xlet is read-only and presents a list of queues. For each queue, the xlet displays a status box for each logged in agent. Each status box gives the following information:
- Agent name
- Agent status: Shows the agent’s status. An agent can have six statuses:
- Not in use when he is ready to answer an ACD call
- Int. Incoming when he answered an internal call not from a queue
- Int. Outgoing when he emitted an internal call not from a queue
- Ext. Incoming when he answered an external call not from a queue
- Ext. Outgoing when he emitted an external call not from a queue
- In use when he is either on call a from a queue, on pause or on wrapup
- Agent status since: Shows the time spent in the current status
- Background color:
- green if Not in use
- purple if Int. Incoming or Int. Outgoing
- pink if Ext. Incoming or Ext. Outgoing
- orange if In use
Note that the agent status will only change when the communication is established, not when phones are ringing.
If an agent emits a call via his XiVO Client, the status will change to Int. Outgoing or Ext. Outgoing when the destination phone rings, instead of when the destination answers.
- Given the agent is on an ACD call
- When the agent logs out
- When the agent hangs up the ACD call
- When the agent logs back in via CTI Client
- Then the agent may be seen as outgoing non-ACD communication, whether there is a non-ACD communication or not
To make the agent Not in use again, make a non-ACD call and hangup.
- Given the agent is on ACD call
- When the agent calls someone else (e.g. his supervisor)
- When the ACD call hangs up (while the agent talks to his supervisor)
- Then the agent is seen as available, instead of in outgoing non-ACD communication.
This applies to all kinds of non-ACD calls.
The disposition of the Xlet can be changed in two ways:
- Placement of queues
- Which queues are displayed
The disposition is saved whenever the XiVO Client is closed and restored when it is opened again.
The little windows containing each queue can be resized and moved around. That way, any layout can be achieved, according to the size and importance of each queue.
There is a little contextual menu when right-clicking on the title bar of every queue window. Checking/unchecking the lines of this menu shows/hides the associated queue.

- There is no profile containing this xlet. The profile must be created manually.
- There is no sorting on agents in a queue.
- An empty queue will display an empty box with no message specifying the queue has no logged agents.
No special configuration is necessary other than creating a CTI profile in which the Agent Status Dashboard is added.
Skills-Based Routing¶
Skills-based routing (SBR), or Skills-based call routing, is a call-assignment strategy used in call centres to assign incoming calls to the most suitable agent, instead of simply choosing the next available agent. It is an enhancement to the Automatic Call Distributor (ACD) systems found in most call centres. The need for skills-based routing has arisen, as call centres have become larger and dealt with a wider variety of call types.
—Wikipedia
In this respect, skills-based routing is also based on call distribution to agents through waiting queues, but one or many skills can be assigned to each agent, and call can be distributed to the most suitable agent.
In skills-based routing, you will have to find a way to be able to tag the call for a specific skill need. This can be done for example by entering the call distribution system using different incoming call numbers, using an IVR to let the caller do his own choice, or by requesting to the information system database the customer profile.

Skills-Based Routing
- Create the skills
- Apply the skills to the agents
- Create the skill rule sets
- Assign the skill rule sets using a configuration file
- Apply the skill rule sets to call qualification, i.e. incoming calls by using the preprocess subroutine field
Note that you shouldn’t use skill based routing on a queue with queue members of type user because the behaviour is not defined and might change in a future XiVO version.
Skills are created using the menu
. Each skill belongs to a category. First create the category, and in this category create different skills.Note that a skill name can’t contain upper case letters and must be globally unique (i.e. the same name can’t be used in two different categories).

Skills Creation
Once all the skills are created you may apply them to agents. Agents may have one or more skills from different categories.

Apply Skills to Agents
It is typical to use a value between 0 and 100 inclusively as the weight of a skill, although any integer is accepted.
Once skills are created, rule sets can be defined.
A rule set is a list of rules. Here’s an example of a rule set containing 2 rules:
- WT < 60, english > 50
- english > 0
The first rule of this rule set can be read as:
If the caller has been waiting for less than 60 seconds (WT < 60), only try to call agents which have the skill “english” set to a value higher than 50; otherwise, go to the next rule.
And the second rule can be read as:
Only try to call agents which have the skill “english” set to a value higher than 0.
Let’s examine some simple scenarios, because there’s actually some subtilities on how calls are distributed. We will suppose that we have a queue with the default settings and the following members:
- Agent A, with skill english set to 75
- Agent B, with skill english set to 25
Given:
- Agent A is logged and not in use
- Agent B is logged and not in use
- There is no call in the queue
When a new call enters the queue, then it is distributed to Agent A. As long as Agent A is available and doesn’t answer the call, the call will never be distributed to Agent B, even after 60 seconds of waiting time.
When another call enters the queue, then after 60 seconds of waiting time, this call will be distributed to Agent B (and the first call will still be distributed only to Agent A).
The reason is that there’s a difference between a call that is being distributed (i.e. that is making agents ring) and a call that is waiting for being distributed. When a call is being distributed to a set of members, no other rule is tried as long as there’s at least 1 of these members available.
Given:
- Agent A is not logged
- Agent B is logged and not in use
- There is no call in the queue
When a new call enters the queue, then it is immediately distributed to Agent B.
The reason is that when there’s no logged agent matching a rule, the next rule is immediately tried.
Each rule set is composed of rules, and each rule has two parts, separated by a comma:
- the first part (optional) is the “dynamic part”
- the second part is the “skill part”
Each part contains an expression composed of operators, variables and integer constants.
The following operators can be used inside rules:
Comparison operators:
- operand1 ! operand2 (is not equal)
- operand1 = operand2 (is equal)
- operand1 > operand2 (is greater than)
- operand1 < operand2 (is lesser than)
Logical operators:
- operand1 & operand2 (both are true)
- operand1 | operand2 (at least one of them are true)
‘!’ is the operator with the higher priority, and ‘|’ the one with the lower priority. You can use parentheses ‘()’ to change the priority of operations.
The dynamic part can reference the following variables:
- WT
- EWT
The waiting time (WT) is the elapsed time since the call entered the queue. The time the call pass in an IVR or another queue is not taken into account.
The estimated waiting time (EWT) has never fully worked. It is mentioned here only for historical reason. You should not use it. It might be removed in a future XiVO version.
Examples: |
---|
- WT < 60
The skill part can reference any skills name as variables.
You can also use meta-variables, starting with a ‘$’, to substitute them with data set on the Queue() call. For example, if you call Queue() with the skill rule set argument equal to:
select_lang(lang=german)
Then every $lang
occurrence will be replaced by ‘german’.

Create Skill Rule Sets
Examples: |
---|
- english > 50
- technic ! 0 & ($os > 29 & $lang > 39 | $os > 39 & $lang > 19)
Note that the expression:
english | french
is equivalent to:
english ! 0 | french ! 0
Sometimes, a rule references a skill which is not defined for every agent. For example, given the following rule:
english > 0 | english < 1
Then, for an agent which has the skill english defined, the result of this expression is always true. For an agent which does not have the skill english defined, the result of this expression is always false.
Said differently, an agent without a skill X is not the same as an agent with the skill X set to the value 0.
Technically, this is what is happening when evaluating the rule “english > 0” for an agent without the skill english:
english > 0
= <Substituing english with the agent value>
"undefined" > 0
= <A comparison with "undefined" in at least one operand yields undefined>
"undefined"
= <In a boolean context, "undefined" is equal to false>
false
This behaviour applies to every comparison operators.
Also, the syntax that is currently accepted for comparison is always of the form:
variable cmp_op constant
Where “variable” is a variable name, “cmp_op” is a comparison operator and “constant” is an integer constant. This means the following expressions are not accepted:
- 10 < english (but english > 10 is accepted)
- english < french (the second operand must be a constant)
- 10 < 11 (the first operand must be a variable name)
A skill rule set is attached to a call using a bit of dialplan. This dialplan is stored in a configuration file you may edit using menu
.In the figure above, 3 different languages are selected using three different subroutines.
Each of this different selections of subroutines can be applied to the call qualifying object. In the following example language selection is applied to incoming calls.
Example: |
---|
Configuration file for simple skill selection :
[simple_skill_english]
exten = s,1,Set(XIVO_QUEUESKILLRULESET=english_rule_set)
same = n,Return()
[simple_skill_french]
exten = s,1,Set(XIVO_QUEUESKILLRULESET=french_rule_set)
same = n,Return()
You may monitor your waiting calls with skills using the asterisk CLI and the
command queue show <queue_name>
:
xivo-jylebleu*CLI> queue show services
services has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime, 2s talktime), W:0, C:1, A:10, SL:0.0% within 0s
Members:
Agent/2000 (Not in use) (skills: agent-1) has taken no calls yet
Agent/2001 (Unavailable) (skills: agent-4) has taken no calls yet
Virtual queue english:
Virtual queue french:
1. SIP/jyl-dev-assur-00000017 (wait: 0:05, prio: 0)
Callers:
You may monitor your skills groups with the command queue show skills groups <agent_name>
:
xivo-jylebleu*CLI> queue show skills groups <PRESS TAB>
agent-2 agent-3 agent-4 agent-48 agent-7 agent-1
xivo-jylebleu*CLI> queue show skills groups agent-1
Skill group 'agent-1':
- bank : 50
- english : 100
You may monitor your skills rules with the command queue show skills rules <rule_name>
:
xivo-jylebleu*CLI> queue show skills rules <PRESS TAB>
english french select_lang
xivo-jylebleu*CLI> queue show skills rules english
Skill rules 'english':
=> english>90
Statistics¶
The statistics page is used to monitor the efficiency of queues and agents. Statistics are automatically generated every six hours. They can also be generated manually.
Note
The contact center statistics do not apply to switchboard queues. See Switchboard for more details.
Note
The oldest statistics are periodically removed. See Purge Logs for more details.
In order to display call center statistics, you must create at least one configuration profile.
The configuration profile is used to generate reports from the cache. The cache is generated independently from the configuration so adding a new configuration does not require a new cache generation.
Field | Values | Description |
---|---|---|
name | string | Configuration name, useful for remembering what the configuration is used for |
interval | enum [0-999] [day, week, month] | Default time interval used when displaying statistics. Examples: “-1 day”: show statistics for yesterday “-3 weeks”: show statistics for the last 3 weeks |
show on summary page | Display this configuration on the summary page | |
timezone | America/Montreal | Your time zone |
Period cache | Maximum and minimum dates that can be used for displaying statistics | |
start | YYYY-MM | Start date |
end | YYYY-MM | End date. If left to 0, use the servers’ current date |
Working Hours | Work hours for agents | |
start | hh:mm | Beginning of working hours. |
end | hh:mm | End of working hours |
Periods | Number of calls answered for a time period | |
Period 1 | number of seconds (Example: 20) | Show number of calls answered within 20 seconds in column “P1” |
Period n | number of seconds (Example: 20) | Show number of calls answered within 20 seconds in column “Pn” |
Note
Calls outside of working hours will not be in the cache. e.g. if working hours are from 8:00 AM to 16:00 PM, a call at 7:55 AM will not show up in the reports.
Note
Statistics are computed on the hour. e.g. If work hours are from 8:30 to 16:15, working hours should be set from 8:00 to 17:00.
Note
Period includes both bounds, if the same number is used for the higher bound and the lower bound of next period, some calls will be counted twice. i.e period 1 : 0-30 period 2 : 31-60 period 3 : 61
The cache must be generated before using reports. By default, the cache is automatically generated every six hours.
However, you can safely generate it manually. The script to generate the cache is xivo-stat fill_db. When this script is run, statistics will be regenerated for the last 8 hours starting from the previous hour. e.g. If you run xivo-stat on 2012-08-04 11:47:00, statistics will be regenerated from 2012-08-04 03:00:00 to 2012-08-04 11:47:00
Note
xivo-stat fill_db can be a long operation when used for the first time or after a xivo-stat clean_db
Warning
The current events have an end date of the launch date of the script xivo-stat as the end date.
If for some reason the cache generation fails or the cache becomes unusable, the administrator can safely clean the cache using xivo-stat clean_db and then regenerate it. This operation will only clear the cache and does not erase any other data.
Queue statistics can be viewed in
.The first table displays a list of queues with all the counters for the period choosen from the Dashboard panel
By clicking on a queue name you may display detailed queue statistics
Statistics can be displayed :
- Received: Total number of received calls
- Answered: Calls answered by an agent
- Abandoned: Calls that were hung up while waiting for an answer
- Dissuaded or Overflowed:
- Closed: Calls received when the queue was closed
- No answer (NA): Calls that reached the ring timeout delay
- Satured: Calls received when the queue was already full (“Maximum number of people allowed to wait:” limit of advanced tab) or when one of the diversion parameters were reached
- Blocking : Calls received when no agents were available or when there were no agents to take the call, join an empty queue condition or remove callers if there are no agents condition is reached (advanced queue parameter tab).
- Average waiting time (AWT): The average waiting time of call on wait
- Answered rate (HR): The ratio of answered calls over (received calls - closed calls)
- Quality of service (QoS): Percentage of calls answered in less than x seconds over the number of answered calls, where x is defined in the configuration
Agent performance statistics can be viewed in
.
Note
The agent performance counters do not take into account transfer between agents: if agent A processes a call and transfers it to agent B, only the counters of agent A will be updated. Ignoring any info after the call transfer.
- Answered: Number of calls answered by the agent
- Conversation: Total time spent for calls answered during a given period
- Login: Total login time of an agent.
- Wrapup: Total time spent in wrapup by an agent.
- Pause: Total pause time of an agent
Warning
Data generated before XiVO 12.19 might have erroneous results for the Login time counter
Note
The Pause time counter only supports PAUSEALL and UNPAUSEALL command from cticlient. The agent must also be a member of a least 1 queue.
Note
Wrapup time events were added to XiVO in version 12.21





Display by period defined in configuration, i.e. between 0 and 10s, 10s and 30s etc ... the number of handled calls and the number of abandonned calls

You may click on a queue name to get more information for this queue




Reporting¶
You may use your own reporting tools to be able to produce your own reports provided you do not use the XiVO server original tables, but copy the tables to your own data server. You may use the following procedure as a template :
- Allow remote database access on XiVO
- Create a postgresql account read only on asterisk database
- Create target tables in your database located on the data server
- Copy the statistic table content to your data server
- The queue_log table of the asterisk database is filled by events from Asterisk and by custom dialplan events
- xivo-stat fill_db is then used to read data from the queue_log table and generate the tables stat_call_on_queue and stat_queue_periodic
- The web interface generate tables and graphics from the stat_call_on_queue and stat_queue_periodic tables depending on the selected configuration
This table is used to store each call individually. Each call received on a queue generates a single entry in this table containing time related fields and a foreign key to the agent who answered the call and another on the queue on which the call was received.
It also contains the status of the call ie. answered, abandoned, full, etc.
Field | Values | Description |
---|---|---|
id | generated | |
callid | numeric value | This call id is also used in the CEL table and can be used to get call detail information |
time | Call time | |
ringtime | Ringing duration time in seconds | |
talktime | Talk time duration in seconds | |
waittime | Wait time duration in seconds | |
status | See status description below | |
queue_id | Id of the queue, the name of the queue can be found in table stat_queue , using this name
queue details can be found in table queuefeatures |
|
agent_id | Id of the agent, the agent name can be found in table stat_agent , using this name
agent details can be found in table agentfeatures using the number in the second part of the name
Exemple : Agent/1002 is agent with number 1002 in table agentfeatures |
Status | Description |
---|---|
full | Call was not queued because queue was full, happens when the number of calls is greater than the maximum number of calls allowed to wait |
closed | Closed due to the schedule applied to the queue |
joinempty | No agents were available in the queue to take the call (follows the join empty parameter of the queue) |
leaveempty | No agents available while the call was waiting in the qeuue |
divert_ca_ratio | Call diverted because the ratio number of agent number of calls waiting configured was exceeded |
divert_waittime | Call diverted because the maximum expected waiting time configured was exceeded |
answered | Call was answered |
abandoned | Call hangup by the caller |
timeout | Call stayed longer than the maximum time allowed in queue parameter |
This table is an aggregation of the queue_log table.
This table contains counters on each queue for each given period. The granularity at the time of this writing is an hour and is not configurable. This table is then used to compute statistics for a given range of hours, days, week, month or year.
Field | Description |
---|---|
id | Generated id |
time | time period, all counters are aggregated for an hour |
answered | Number of answered calls during the period |
abandoned | Number of abandoned calls during the period |
total | Total calls received during the period |
full | Number of calls received when queue was full |
closed | Number of calls received on close |
joinempty | Number of calls received no agents available |
leaveempty | Number of calls diverted agents not available during the wait |
divert_ca_ratio | Number of calls diverted due to the number of agent number versus calls waiting configured was exceeded |
divert_waittime | Number of calls diverted because the maximum expected waiting time configured was exceeded |
timeout | Number of calls diverted because the maximum time allowed in queue parameter was exceeded |
queue_id |
This table is used to match agents to an id that is different from the id in the agent configuration table. This is necessary to avoid loosing statistics on a deleted agent. This also means that if an agent changes number ie. Agent/1001 to Agent/1202, the supervisor will have to take this information into account when viewing the statistics. Affecting an old number to a another agent also means that the supervisor will have to ignore entries for this given agent for the period before the number assignment to the new agent.
This table is used to store queues in a table that is different from the queue configuration table. This is necessary to avoid losing statistics on a deleted queue. Renaming a queue is also not handled at this time.
High Availability (HA)¶
The HA solution in XiVO makes it possible to maintain basic telephony function whether your main XiVO server is running or not. When running a XiVO HA cluster, users are guaranteed to never experience a downtime of more than 5 minutes of their basic telephony service.
The HA solution in XiVO is based on a 2-nodes “master and slave” architecture. In the normal situation, both the master and slave nodes are running in parallel, the slave acting as a “hot standby”, and all the telephony services are provided by the master node. If the master fails or must be shutdown for maintenance, then the telephony devices automatically communicate with the slave node instead of the master one. Once the master is up again, the telephony devices failback to the master node. Both the failover and the failback operation are done automatically, i.e. without any user intervention, although an administrator might want to run some manual operations after failback as to, for example, make sure any voicemail messages that were left on the slave are copied back to the master.
Prerequisites¶
The HA in XiVO only works with telephony devices (i.e. phones) that support the notion of a primary and backup telephony server.
- Phones must be able to reach the master and the slave (take special care if master and slave are not in the same subnet)
- If firewalling, the master must be allowed to join the slave on ports 22 and 5432
- If firewalling, the slave must be allowed to join the master with an ICMP ping
- Trunk registration timeout (
expiry
) should be less than 300 seconds (5 minutes) - The slave must have no provisioning plugins installed.
The HA solution is guaranteed to work correctly with the following devices.
Quick Summary¶
- You need two configured XiVO (wizard passed)
- Configure one XiVO as a master -> setup the slave address (VoIP interface)
- Restart services (xivo-service restart) on master
- Configure the other XiVO as a slave -> setup the master address (VoIP interface)
- Configure file synchronization by runnning the script
xivo-sync -i
on the master - Start configuration synchronization by running the script
xivo-master-slave-db-replication <slave_ip>
on the master - Resynchronize all your devices
- Configure the XiVO Clients
That’s it, you now have a HA configuration, and every hour all the configuration done on the master will be reported to the slave.
Configuration Details¶
First thing to do is to install 2 XiVO.
Important
When you upgrade a node of your cluster, you must also upgrade the other so that they both are running the same version of XiVO. Otherwise, the replication might not work properly.
You must configure the HA in the Web interface (
page).You can configure the master and slave in whatever order you want.
You must also run xivo-sync -i
on the master to setup file synchronization. Running xivo-sync
-i
will create a passwordless SSH key on the master, stored under the /root/.ssh
directory,
and will add it to the /root/.ssh/authorized_keys
file on the slave. The following directories
will then be rsync’ed every hour:
- /etc/asterisk/extensions_extra.d
- /etc/xivo/asterisk
- /var/lib/asterisk/agi-bin
- /var/lib/asterisk/moh
- /var/lib/xivo/certificates
- /var/lib/xivo/sounds/acd
- /var/lib/xivo/sounds/playback
Warning
When the HA is configured, some changes will be automatically made to the configuration of XiVO.
SIP expiry value on master and slave will be automatically updated:
- min: 3 minutes
- max: 5 minutes
- default: 4 minutes

The provisioning server configuration will be automatically updated in order to allow phones to switch from XiVO power failure.

Warning
Do not change these values when the HA is configured, as this may cause problems. These values will be reset to blank when the HA is disabled.
Important
For the telephony devices to take the new proxy/registrar settings into account, you must resynchronize the devices or restart them manually.
Default status of High Availability is disabled:
Note
You can reset at any time by choosing a server mode (disabled)

HA Dashboard Disabled (default state)
Important
You have to restart services (xivo-service restart) once the master node is disabled.
In choosing the method Master
you must enter the IP address of the VoIP interface of the slave node.

HA Dashboard Master
Important
You have to restart all services (xivo-service restart) once the master node is configured.
In choosing the method Slave
you must enter the IP address of the VoIP interface of the master node.

HA Dashboard Slave
Once master slave configuration is completed, XiVO configuration is replicated from the master node to the slave every hour (:00).
Replication can be started manually by running the replication scripts on the master:
xivo-master-slave-db-replication <slave_ip>
xivo-sync
The replication does not copy the full XiVO configuration of the master. Notably, these are excluded:
- All the network configuration (i.e. everything under the section)
- All the support configuration (i.e. everything under the section)
- Call logs
- Call center statistics
- Certificates
- HA settings
- Provisioning configuration
- Voicemail messages
Less importantly, these are also excluded:
- Queue logs
- CELs
You have to enter the master and slave address in the Connection
tab of the
XiVO Client configuration :

The main server is the master node and the backup server is the slave node.
When connecting the XiVO Client with the main server down, the login screen will hang for 3 seconds before connecting to the backup server.
Internals¶
4 scripts are used to manage services and data replication.
- xivo-master-slave-db-replication <slave_ip> is used on the master to replicate the master’s data on the slave server. It runs on the master.
- xivo-manage-slave-services {start,stop} is used on the slave to start, stop monit and asterisk. The services won’t be restarted after an upgrade or restart.
- xivo-check-master-status <master_ip> is used to check the status of the master and enable or disable services accordingly.
- xivo-sync is used to sync directories from master to slave.
Limitations¶
When the master node is down, some features are not available and some behave a bit differently. This includes:
- Call history / call records are not recorded.
- Voicemail messages saved on the master node are not available.
- Custom voicemail greetings recorded on the master node are not available.
- Phone provisioning is disabled, i.e. a phone will always keep the same configuration, even after restarting it.
- Phone remote directory is not accessible, because provisioned IP address points to the master.
Note that, on failover and on failback:
- DND, call forwards, call filtering, ..., statuses may be lost if changed recently.
- If you are connected as an agent, then you might need to reconnect as an agent when the master goes down. Since it’s hard to know when the master goes down, if your CTI client disconnects and you can’t reconnect it, then it’s a sign the master might be down.
Additionally, only on failback:
- Voicemail messages are not copied from the slave to the master, i.e. if someone left a message on your voicemail when the master was down, you won’t be able to consult it once the master is up again.
- More generally, custom sounds are not copied back. This includes recordings.
Here’s the list of limitations that are more relevant on an administrator standpoint:
- The master status is up or down, there’s no middle status. This mean that if Asterisk is crashed the XiVO is still up and the failover will NOT happen.
Berofos Integration¶
XiVO offers the possibility to integrate a berofos failover switch within a HA cluster.
This is useful if you have one or more ISDN lines (i.e. T1/E1 or T0 lines) that you want to use whatever the state of your XiVO HA cluster. To use a berofos within your XiVO HA installation, you need to properly configure both your berofos and your XiVOs, then the berofos will automatically switch your ISDN lines from your master node to your slave node if your master goes down, and vice-versa when it comes back up.
You can also use a Berofos failover switch to secure the ISDN provider lines when installing a XiVO in front of an existing PBX. The goal of this configuration is to mitigate the consequences of an outage of the XiVO : with this equipment the ISDN provider links could be switched to the PBX directly if the XiVO goes down.
XiVO does not offer natively the possibility to configure Berofos in this failover mode. The Berofos Integration with PBX section describes a workaround.
There is nothing to be done on the master node.
First, install the bntools package:
apt-get install bntools
This will make the bnfos
command available.
You can then connect your berofos to your network and power it on. By default, the berofos will try to get an IP address via DHCP. If it is not able to get such address from a DHCP server, it will take the 192.168.0.2/24 IP address.
Note
The DHCP server on XiVO does not offer IP addresses to berofos devices by default.
Next step is to create the /etc/bnfos.conf
file via the following command:
bnfos --scan -x
If no berofos device is detected using this last command, you’ll have to explicitly specify the IP address of the berofos via the -h option:
bnfos --scan -x -h <berofos ip>
At this stage, your /etc/bnfos.conf
file should contains something like this:
[fos1]
mac = 00:19:32:00:12:1D
host = 10.34.1.50
#login = <user>:<password>
It is advised to configure your berofos with a static IP address. You first need to put your berofos into flash mode :
- press and hold the black button next to the power button,
- power on your berofos,
- release the black button when the red LEDs of port D start blinking.
Then, you can issue the following command, by first replacing the network configuration with your one:
bnfos --netconf -f fos1 -i 10.34.1.20 -n 255.255.255.0 -g 10.34.1.1 -d 0
Note
-i
is the IP address-n
is the netmask-g
is the gateway-d 0
is to disable DHCP
You can then update your berofos firmware to version 1.53:
wget http://www.beronet.com/downloads/berofos/bnfos_v153.bin
bnfos --flash bnfos_v153.bin -f fos1
Once this is done, you’ll have to reboot your berofos in operationnal mode (that is in normal mode).
Then you must rewrite the /etc/bnfos.conf
(mainly if you changed the IP address):
bnfos --scan -x -h <berofos ip>
Now that your berofos has proper network configuration and an up to date firmware, you might want to set a password on your berofos:
bnfos --set apwd=<password> -f fos1
bnfos --set pwd=1 -f fos1
You must then edit the /etc/bnfos.conf
and replace the login line to something like:
login = admin:<password>
Next, configure your berofos for it to work correctly with the XiVO HA:
bnfos --set wdog=0 -f fos1
bnfos --set wdogdef=0 -f fos1
bnfos --set scenario=0 -f fos1
bnfos --set mode=1 -f fos1
bnfos --set modedef=1 -f fos1
This, among other things, disable the watchdog. The switching from one relay mode to the other will be done by the XiVO slave node once it detects the master node is down, and vice-versa.
Finally, you can make sure everything works fine by running the xivo-berofos command:
xivo-berofos master
The green LEDs on your berofos should be lighted on ports A and B.
Here’s how to connect the ISDN lines between your berofos with:
- two XiVOs in high availability
In this configuration you can protect up two 4 ISDN lines. If more than 4 ISDN lines to protect, you must set up a Multiple berofos configuration.
Here’s an example with 4 ISDN lines coming from your telephony provider:
ISDN lines (provider)
| | | |
| | | |
+---------------------------------------------+
| A B C D |
| 1|2|3|4 1|2|3|4 1|2|3|4 1|2|3|4 |
+---------------------------------------------+
| | | | | | | |
| | | | | | | |
+--------+ +--------+
| xivo-1 | | xivo-2 |
+--------+ +--------+
Here’s how to connect your berofos with:
- two XiVOs in high availability,
- one PBX.
In this configuration you can protect up two 2 ISDN lines. If more than 2 ISDN lines to protect, you must set up a Multiple berofos configuration.
Logical view:
+--------+ +-----+
-- Provider ----| xivo-1 | -- ISDN Interconnection --| PBX | -- Phones
+--------+ +-----+
| xivo-2 |
+--------+
This example shows the case where there are 2 ISDN lines coming from your telephony provider:
ISDN lines (provider)
| |
| |
+------------------------------------------------------+
| A B C D |
| 1|2|3|4 1|2 3|4 1|2|3|4 1|2 3|4 |
+------------------------------------------------------+
| | CPE | | | | NET CPE | | | | NET
| | spans | | | | spans spans | | | | spans
| | +----------+ +------------+
| | | xivo-1 | | xivo-2 |
| | +----------+ +------------+
| |
| |
+------+
| PBX |
+------+
This case is not currently supported. You’ll find a workaround in the Berofos Integration with PBX section.
It’s possible to use more than 1 berofos with XiVO.
For each supplementary berofos you want to use, you must first configure it properly
like you did for the first one. The only difference is that you need to add a berofos
declaration to the /etc/bnfos.conf
file instead of creating/overwriting the
file. Here’s an example of a valid config file for 2 berofos:
[fos1]
mac = 00:19:32:00:12:1D
host = 10.100.0.201
login = admin:foobar
[fos2]
mac = 00:11:22:33:44:55
host = 10.100.0.202
login = admin:barfoo
Warning
berofos name must follow the pattern fosX
where X is a number starting with 1,
then 2, etc. The bnfos
tool won’t work properly if it’s not the case.
When your XiVO switch the relay mode of your berofos, it logs the event in the
/var/log/syslog
file.
Note that when the berofos is off, the A and D ports are connected together. This behavior is not customizable.
It is important to remove the /etc/bnfos.conf
file on the slave node when you don’t
want to use anymore your berofos with your XiVOs.
You can reset the berofos configuration :
- Power on the berofos,
- When red and green LEDs are still lit, press & hold the black button,
- Release it when the red LEDs of the D port start blinking fast
- Reboot the beronet, it should have lost its configuration.
Troubleshooting¶
When replicating the database between master and slave, if you encounter problems related to the system locale, see PostgreSQL localization errors.
Scalability and Distributed Systems¶
This section gathers configuration that are possible using XiVO to feature rich scalable communication systems.
Contact and Presence Sharing¶
XiVO allow the administrator to share presence and statuses between multiple installations. For example, an enterprise could have a XiVO in each office and still be able to search, contact and view the statuses of colleagues in other offices.
This page will describe the steps that are required to configure such use case.

- All XiVO that you interconnect should be on the same version
- This configuration is only possible with XiVO 15.19 and above
- All ports necessary for communication should be open Network
Warning
If you are cloning a virtual machine or copy the database, the UUID of the two XiVO will be the same, you must regenerate them in the infos table of the asterisk database and restart all services. You must also remove all consul data that included the old UUID.
Warning
Telephony will be interrupted during the configuration period.
Warning
The configuration must be applied to each XiVO you want to interconnect. For example, if 6 different XiVO are to be connected, the configuration for all other XiVO should be added. This does not apply to the message bus which can use a ring policy, each XiVO talking to its two neighbours.
Warning
You should use your firewall to restrict access to the HTTP ports of consul and xivo-ctid, because they don’t have any authentication mechanism enabled.
Note
In an architecture with a lot of XiVO, we recommend that you centralize some services, such as xivo-dird, to make your life easier. Don’t forget redundancy. This applies also to RabbitMQ and Consul. In this case, the configuration will have to be done entirely manually in YAML config files.
For this procedure, the following name and IP addresses will be used:
- XiVO 1: 192.168.1.124
- XiVO 2: 192.168.1.125
The first thing is to make XiVO accept remote connections to your internal users directory. For this, you must create a Web service access by authorizing either an IP address or a login/password.
This can be done in


For each remote XiVO a new directory has to be created in
Note
We recommend doing a working configuration without certificate verification first. Once you get it working, enable certificate verification.


To add a new directory definition, go to

In each directory definition, add the fields to match the configured Display filters

At the moment of this writing xivo-dird profiles are mapped directly to the user’s profile. For each internal context where you want to be able to see user’s from other XiVO, add the new directory definitions in
.

To apply the new directory configuration, you can either restart from:
- on the command line service xivo-dird restart
At this point, you should be able to search for users on other XiVO from the People Xlet.
rabbitmqctl add_user xivo xivo
rabbitmqctl set_user_tags xivo administrator
rabbitmqctl set_permissions -p / xivo ".*" ".*" ".*"
rabbitmq-plugins enable rabbitmq_federation
service rabbitmq-server restart
rabbitmqctl set_parameter federation-upstream xivo-dev-2 '{"uri":"amqp://xivo:xivo@192.168.1.125","max-hops":1}' # remote IP address
rabbitmqctl set_policy federate-xivo 'xivo' '{"federation-upstream-set":"all"}' --priority 1 --apply-to exchanges
Create a configuration file for xivo-ctid, e.g /etc/xivo-ctid/conf.d/interconnection.yml
rest_api:
http:
listen: 0.0.0.0
service_discovery:
advertise_address: auto
advertise_address_interface: eth0 # Interface bearing the IP address of this XiVO, reachable from outside
service xivo-ctid restart
apt-get install consul-cli
consul-cli agent-services --ssl --ssl-verify=false
The output should include a service names xivo-ctid with an address that is reachable from other XiVO.
{"consul": {"ID": "consul",
"Service": "consul",
"Tags": [],
"Port": 8300,
"Address": ""},
"e546a652-e290-47e2-8519-ec3642daa6e6": {"ID": "e546a652-e290-47e2-8519-ec3642daa6e6",
"Service": "xivo-ctid",
"Tags": ["xivo-ctid",
"607796fc-24e2-4e26-8009-cbb48a205512"],
"Port": 9495,
"Address": "192.168.1.124"}}
xivo-service stop
rm -rf /var/lib/consul/raft/
rm -rf /var/lib/consul/serf/
rm -rf /var/lib/consul/services/
rm -rf /var/lib/consul/tmp/
rm -rf /var/lib/consul/checks/
Add a new configuration file /etc/consul/xivo/interconnection.json with the following content where advertise_addr is reachable from other XiVO.
{
"client_addr": "0.0.0.0",
"bind_addr": "0.0.0.0",
"advertise_addr": "192.168.1.124" // The IP address of this XiVO, reachable from outside
}
consul configtest --config-dir /etc/consul/xivo/
No output means that the configuration is valid.
service consul start
xivo-service start
Join another member of the Consul cluster. Only one join is required as members will be propagated.
consul join -wan 192.168.1.125
List other members of the cluster with the following command
consul members -wan
Check consul logs for problems
consul monitor
There is no further configuration needed, you should now be able to connect your XiVO Client and search contacts from the People Xlet. When looking up contacts of another XiVO, you should see their phone status, their user availability, and agent status dynamically.
Chances are that everything won’t work the first time, here are some interesting commands to help you debug the problem.
tail -f /var/log/xivo-dird.log
tail -f /var/log/xivo-ctid.log
tail -f /var/log/xivo-confd.log
consul monitor
consul members -wan
consul-cli agent-services --ssl --ssl-verify=false
rabbitmqctl eval 'rabbit_federation_status:status().'
One you get this part working, check out Phonebook Sharing.
Phonebook Sharing¶
Sharing phonebooks allows users of different XiVO servers to access the contacts in the phonebooks of the other XiVO servers.

This procedure follows the Contact and Presence Sharing (but it’s not mandatory), so we will use the same conventions.
On each XiVO, you must have a Web Services User that authorizes access from another host (not by login/password). The phonebook access does not support login/password authorization.

This Web Services user will allow other XiVO servers to access the phonebook of this XiVO.
For each remote XiVO a new phonebook has to be created in


Note that the URL of the directory must contain restricted
, not private
, e.g:
http://192.168.1.125/service/ipbx/json.php/restricted/pbx_services/phonebook
To add a new directory definition, go to

In each directory definition, add the fields to match the other phonebooks:

There is no further configuration needed, you should now be able to connect your XiVO Client and search phonebook contacts from the People Xlet.
Chances are that everything won’t work the first time, here are some interesting commands to help you debug the problem.
tail -f /var/log/xivo-dird.log
tail -f /var/log/nginx/xivo.access.log
API and SDK¶
Message Bus¶
The message bus is used to receive events from XiVO. It is provided by an AMQP 0-9-1 broker (namely, RabbitMQ) that is integrated in XiVO.
Warning
Interaction with the bus is presently experimental and some things might change in the next XiVO versions.
At the moment, the AMQP broker only listen on the 127.0.0.1 address. This means that if you want to connect to the AMQP broker from a distant machine, you must modify the RabbitMQ server configuration, which is not yet an officially supported operation. All events are sent to the xivo exchange.
Otherwise, the default connection information is:
- Virtual host: /
- User name: guest
- User password: guest
- Port: 5672
- Exchange name: xivo
- Exchange type: topic
Here’s an example of a simple client, in python, listening for the call_form_result CTI events:
import kombu
from kombu.mixins import ConsumerMixin
EXCHANGE = kombu.Exchange('xivo', type='topic')
ROUTING_KEY = 'call_form_result'
class C(ConsumerMixin):
def __init__(self, connection):
self.connection = connection
def get_consumers(self, Consumer, channel):
return [Consumer(kombu.Queue(exchange=EXCHANGE, routing_key=ROUTING_KEY),
callbacks=[self.on_message])]
def on_message(self, body, message):
print('Received:', body)
message.ack()
def main():
with kombu.Connection('amqp://guest:guest@localhost:5672//') as conn:
try:
C(conn).run()
except KeyboardInterrupt:
return
main()
If you are new to AMQP, you might want to look at the RabbitMQ tutorial.
Things to be aware when writing a client/consumer:
- The published messages are not persistent. When the AMQP broker stops, the messages that are still in queues will be lost.
- The call_held bus message has been added.
- The call_resumed bus message has been added.
- The user_status_update bus message now uses the user’s UUID instead of the user’s ID.
- The user_created bus message has been added.
- The user_edited bus message has been added.
- The user_deleted bus message has been added.
- The chat_message_event bus message has been added.
- The service_registered_event and service_deregistered_event bus messages have been added.
Events that are sent to the bus use a JSON serialization format with the content-type application/json. For example, the CTI call_form_result event looks like this:
{"name": "call_form_result",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {...}}
All events have the same basic structure, namely, a JSON object with 4 keys:
- name
- A string representing the name of the event. Each event type has a unique name.
- required_acl (optional)
- Either a string or null. Currently used by xivo-websocketd to determine if a client can receive the event or not. See the Events Access Control section for more information.
- origin_uuid
- The uuid to identify the message producer.
- data
- The data specific part of the event. This is documented on a per event type; if not this is assumed to be null.
All AMI events are broadcasted on the bus.
- routing key: ami.<event name>
- event specific data: a dictionary with the content of the AMI event
Example event with binding key QueueMemberStatus:
{
"name": "QueueMemberStatus",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"Status": "1",
"Penalty": "0",
"CallsTaken": "0",
"Skills": "",
"MemberName": "sip\/m3ylhs",
"Queue": "petak",
"LastCall": "0",
"Membership": "static",
"Location": "sip\/m3ylhs",
"Privilege": "agent,all",
"Paused": "0",
"StateInterface": "sip\/m4ylhs"
}
}
The call_form_result event is sent when a custom call form is submitted by a CTI client.
- routing key: call_form_result
- event specific data: a dictionary with 2 keys:
- user_id: an integer corresponding to the user ID of the client who saved the call form
- variables: a dictionary holding the content of the form
Example:
{
"name": "call_form_result",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"user_id": 40,
"variables": {
"firstname": "John",
"lastname": "Doe"
}
}
}
The agent_status_update is sent when an agent is logged in or logged out.
- routing key: status.agent
- required ACL: events.statuses.agents
- event specific data: a dictionary with 3 keys:
- agent_id: an integer corresponding to the agent ID of the agent who’s status changed
- status: a string identifying the status
- xivo_id: the uuid of the xivo
Example:
{
"name": "agent_status_update",
"required_acl": "events.statuses.agents",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"agent_id": 42,
"xivo_id": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"status": "logged_in"
}
}
The events call_created
, call_updated
, call_ended
are sent when a call handled by
xivo-ctid-ng is received, connected or hung up.
- routing key: calls.call.created, calls.call.updated, calls.call.ended
- required ACL: events.calls.<user_uuid>
- event specific data: a dictionary with the same fields as the REST API model of Call (See http://api.xivo.io, section xivo-ctid-ng)
Example:
{
"name": "call_created",
"required_acl": "events.calls.2e752722-0864-4665-887d-a78a024cf7c7",
"origin_uuid": "08c56466-8f29-45c7-9856-92bf1ba89b82",
"data": {
"bridges": [],
"call_id": "1455123422.8",
"caller_id_name": "Some One",
"caller_id_number": "1001",
"creation_time": "2016-02-10T11:57:02.592-0500",
"status": "Ring",
"talking_to": {},
"user_uuid": "2e752722-0864-4665-887d-a78a024cf7c7"
}
}
This message is sent when a call is placed on hold
- routing key: calls.hold.created
- event specific data:
- call_id: The asterisk channel unique ID
Example:
{"name": "call_held",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {"call_id": "1465572129.31"}}
This message is sent when a call is resumed from hold
- routing key: calls.hold.deleted
- event specific data:
- call_id: The asterisk channel unique ID
Example:
{"name": "call_resumed",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {"call_id": "1465572129.31"}}
This message is used to send a chat message to a user
- routing key: chat.message.<xivo-uuid>.<user_id>
- event specific data:
- alias: The nickname of the chatter
- to: The destination’s XiVO UUID and user UUID
- from: The chatter’s XiVO UUID and user UUID
- msg: The message
Example:
{
"name": "chat_message_event",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"alias": "Alice"
"to": ["ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3", "fcb36731-c50a-453e-92c7-571297d41616"],
"from": ["ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3", "4f2e2249-ae2b-4bc2-b5fc-ad42ee01ddaf"],
"msg": "Hi!"
}
}
The endpoint_status_update is sent when an end point status changes. This information is based on asterisk hints.
- routing key: status.endpoint
- required ACL: events.statuses.endpoints
- event specific data: a dictionary with 3 keys
- xivo_id: the uuid of the xivo
- endpoint_id: an integer corresponding to the endpoint ID
- status: an integer corresponding to the asterisk device state
Example:
{
"name": "endpoint_status_update",
"required_acl": "events.statuses.endpoints",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"endpoint_id": 67,
"xivo_id": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"status": 0
}
}
The user_created event is published when a new user is created.
- routing key: config.user.created
- event specific data: a dictionary with 2 keys
- id: the ID of the created user
- uuid: the UUID of the created user
Example:
{
"name": "user_created",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"id": 42,
"uuid": "8e58d2a7-cfed-4c2e-ac72-14e0b5c26dc2"
}
}
The user_deleted event is published when a user is deleted.
- routing key: config.user.deleted
- event specific data: a dictionary with 2 keys
- id: the ID of the deleted user
- uuid: the UUID of the deleted user
Example:
{
"name": "user_deleted",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"id": 42,
"uuid": "8e58d2a7-cfed-4c2e-ac72-14e0b5c26dc2"
}
}
The user_edited event is published when a user is modified.
- routing key: config.user.edited
- event specific data: a dictionary with 2 keys
- id: the ID of the modified user
- uuid: the UUID of the modified user
Example:
{
"name": "user_edited",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"id": 42,
"uuid": "8e58d2a7-cfed-4c2e-ac72-14e0b5c26dc2"
}
}
The user_status_update is sent when a user changes his CTI presence using the XiVO client.
- routing key: status.user
- required ACL: events.statuses.users
- event specific data: a dictionary with 3 keys
- xivo_id: the uuid of the xivo
- user_uuid: the user’s UUID
- status: a string identifying the status
Example:
{
"name": "user_status_update",
"required_acl": "events.statuses.users",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"user_uuid": "8e58d2a7-cfed-4c2e-ac72-14e0b5c26dc2",
"xivo_id": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"status": "busy"
}
}
The users_forwards_<forward_name>_updated is sent when a user changes his forward using REST API.
- forward_name:
- busy
- noanswer
- unconditional
- routing key: config.users.<user_uuid>.forwards.<forward_name>.updated
- required ACL: events.config.users.<user_uuid>.forwards.<forward_name>.updated
- event specific data: a dictionary with 3 keys
- user_uuid: the user uuid
- enabled: the state of the forward
- destination: the destination of the forward
Example:
{
"name": "users_forwards_busy_updated",
"required_acl": "events.config.users.a1223fe6-bff8-4fb6-a982-f9157dea5094.forwards.busy.updated",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"user_uuid": "a1223fe6-bff8-4fb6-a982-f9157dea5094",
"enabled": true
"destination": "1234"
}
}
The users_services_<service_name>_updated is sent when a user changes his service using REST API.
- service_name:
- dnd
- incallfilter
- routing key: config.users.<user_uuid>.services.<service_name>.updated
- required ACL: events.config.users.<user_uuid>.services.<service_name>.updated
- event specific data: a dictionary with 2 keys
- user_uuid: the user uuid
- enabled: the state of the service
Example:
{
"name": "users_services_dnd_updated",
"required_acl": "events.config.users.a1223fe6-bff8-4fb6-a982-f9157dea5094.services.dnd.updated",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"user_uuid": "a1223fe6-bff8-4fb6-a982-f9157dea5094",
"enabled": true
}
}
The service_registered_event is sent when a service is started.
- routing key: service.registered.<service_name>
- event specific data: a dictionary with 5 keys
- service_name: The name of the started service
- service_id: The consul ID of the started service
- address: The advertised address of the started service
- port: The advertised port of the started service
- tags: The advertised Consul tags of the started service
Example:
{
"name": "service_registered_event",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"service_name": "xivo-ctid",
"service_id": "8e58d2a7-cfed-4c2e-ac72-14e0b5c26dc2",
"address": "192.168.1.42",
"port": 9495,
"tags": ["xivo-ctid", "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3", "Québec"]
}
}
The service_deregistered_event is sent when a service is stopped.
- routing key: service.deregistered.<service_name>
- event specific data: a dictionary with 3 keys
- service_name: The name of the stopped service
- service_id: The consul ID of the stopped service
- tags: The advertised Consul tags of the stopped service
Example:
{
"name": "service_deregistered_event",
"origin_uuid": "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3",
"data": {
"service_name": "xivo-ctid",
"service_id": "8e58d2a7-cfed-4c2e-ac72-14e0b5c26dc2",
"tags": ["xivo-ctid", "ca7f87e9-c2c8-5fad-ba1b-c3140ebb9be3", "Québec"]
}
}
Queue logs¶
Queue logs are events logged by Asterisk in the queue_log table of the asterisk database. Queue logs are used to generate XiVO call center statistics.
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
---------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-03 15:27:23.896208 | 1341343640.4 | NONE | Agent/3001 | AGENTCALLBACKLOGIN | 1002@pcm-dev | | | |
Agent/3001 is logged in queues q1 and q2.
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
---------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-03 15:28:07.348244 | NONE | q2 | Agent/3001 | UNPAUSE | | | | |
2012-07-03 15:28:07.346320 | NONE | q1 | Agent/3001 | UNPAUSE | | | | |
2012-07-03 15:28:07.327425 | NONE | NONE | Agent/3001 | UNPAUSEALL | | | | |
2012-07-03 15:28:06.249357 | NONE | NONE | Agent/3001 | AGENTCALLBACKLOGOFF | 1002@pcm-dev | 43 | CommandLogoff | |
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
---------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-04 07:27:55.640421 | 1341401275.9 | q1 | NONE | JOINEMPTY | | | | |
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
---------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-04 07:33:23.085718 | 1341401601.24 | q1 | Agent/3001 | CONNECT | 2 | 1341401601.27 | 1 | |
2012-07-04 07:33:21.165823 | 1341401601.24 | q1 | NONE | ENTERQUEUE | | 1000 | 1 | |
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
---------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-04 07:37:46.601754 | 1341401851.34 | q1 | Agent/3001 | COMPLETEAGENT | 2 | 12 | 1 | |
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
---------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-04 07:40:17.339945 | 1341402016.44 | q1 | NONE | FULL | | | | |
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
---------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-04 07:48:03.455999 | 1341402482.49 | q1 | NONE | CLOSED | | | | |
time | callid | queuename | agent | event | data1 | data2 | data3 | data4 | data5
----------------------------+-----------------+-----------+------------+---------------------+---------------------+-----------------+---------------+-------+-------
2012-07-04 07:49:52.939802 | 1341402586.51 | q1 | NONE | ABANDON | 1 | 1 | 6 | |
REST API¶
The XiVO REST APIs are the privileged way to programmatically interact with XiVO.
You can view the API documentation at http://api.xivo.io.
- Token authentication is now required for all routes, i.e. it is not possible to interact with xivo-agentd without a xivo-auth authentication token.
- xivo-agentd now uses HTTPS instead of HTTP.
The resources returning agent statuses, i.e.:
- GET /agents
- GET /agents/by-id/{agent_id}
- GET /agents/by-number/{agent_number}
are now returning an additional argument named “state_interface”, which is “the interface (e.g. SIP/alice) that is used to determine if an agent is in use or not”.
Note
REST API 1.1 for confd is currently evolving. New features and small fixes are regularly being added over time. We invite the reader to periodically check the changelog for an update on new features and changes.
- New readonly parameters have been added to the trunks resource:
endpoint_sip
endpoint_sccp
- A new readonly parameter have been added to the endpoint_sip and endpoint_custom resource:
trunk
- A new API for associating an extension with an incall has been added:
- DELETE
/1.1/incalls/<incall_id>/extensions/<extension_id>
- PUT
/1.1/incalls/<incall_id>/extensions/<extension_id>
- DELETE
- Added incalls endpoints:
- GET
/1.1/incalls
- POST
/1.1/incalls
- DELETE
/1.1/incalls/<incall_id>
- GET
/1.1/incalls/<incall_id>
- PUT
/1.1/incalls/<incall_id>
- GET
- A new API for associating an endpoint with a trunk has been added:
- DELETE
/1.1/trunks/<trunk_id>/endpoints/sip/<endpoint_id>
- PUT
/1.1/trunks/<trunk_id>/endpoints/sip/<endpoint_id>
- GET
/1.1/trunks/<trunk_id>/endpoints/sip
- GET
/1.1/endpoints/sip/<endpoint_id>/trunks
- DELETE
/1.1/trunks/<trunk_id>/endpoints/custom/<endpoint_id>
- PUT
/1.1/trunks/<trunk_id>/endpoints/custom/<endpoint_id>
- GET
/1.1/trunks/<trunk_id>/endpoints/custom
- GET
/1.1/endpoints/custom/<endpoint_id>/trunks
- DELETE
- Added trunks endpoints:
- GET
/1.1/trunks
- POST
/1.1/trunks
- DELETE
/1.1/trunks/<trunk_id>
- GET
/1.1/trunks/<trunk_id>
- PUT
/1.1/trunks/<trunk_id>
- GET
- Added SIP general endpoints:
- GET
/1.1/asterisk/sip/general
- PUT
/1.1/asterisk/sip/general
- GET
- A new API for associating a user with an agent has been added:
- DELETE
/1.1/users/<user_id>/agents
- GET
/1.1/users/<user_id>/agents
- PUT
/1.1/users/<user_id>/agents/<agent_id>
- DELETE
- A new API to list lines associated to an extension
- GET
/1.1/extensions/<extension_id>/lines
- GET
- The following URLs have been deprecated. Please use the new API instead:
- GET
/1.1/extensions/<extension_id>/line
- GET
- Add possibility to associate many lines to the same user.
- Add possibility to associate many extensions to the same line (only if these lines are associated to the same user).
- A new API for associating a user with a voicemail has been added:
- DELETE
/1.1/users/<user_id>/voicemails
- GET
/1.1/users/<user_id>/voicemails
- PUT
/1.1/users/<user_id>/voicemails
- DELETE
- A new API for associating a line with an extension has been added:
- PUT
/1.1/lines/<line_id>/extensions/<extension_id>
- PUT
- A new API for associating a user with a line has been added:
- PUT
/1.1/users/<user_id>/lines/<line_id>
- PUT
- The following URLs have been deprecated. Please use the new API instead:
- DELETE
/1.1/users/<user_id>/voicemail
- GET
/1.1/users/<user_id>/voicemail
- POST
/1.1/users/<user_id>/voicemail
- POST
/1.1/users/<user_id>/lines
- POST
/1.1/lines/<line_id>/extensions
- DELETE
- Added entities endpoints:
- GET
/1.1/entities
- POST
/1.1/entities
- GET
/1.1/entities/<entity_id>
- DELETE
/1.1/entities/<entity_id>
- GET
- A new API for updating all user’s funckeys
- PUT
/1.1/users/<user_id>/funckeys
- PUT
- A new parameter have been added to the users resource:
dtmf_hangup_enabled
- A new API for initializing a XiVO (passing the wizard):
- GET
/1.1/wizard
- POST
/1.1/wizard
- GET
/1.1/wizard/discover
- GET
- A new API for associating a user with an entity has been added:
- GET
/1.1/users/<user_id>/entities
- PUT
/1.1/users/<user_id>/entities/<entity_id>
- GET
- A new API for associating a user with a call permission has been added:
- GET
/1.1/users/<user_id>/callpermissions
- PUT
/1.1/users/<user_id>/callpermissions/<call_permission_id>
- DELETE
/1.1/users/<user_id>/callpermissions/<call_permission_id>
- GET
/1.1/callpermissions/<call_permission_id>/users
- GET
- Two new parameters have been added to the users resource:
call_permission_password
enabled
- A new API for user’s forwards has been added:
- PUT
/1.1/users/<user_id>/forwards
- PUT
- SIP endpoint:
allow
anddisallow
options are not split into multiple options anymore. - SCCP endpoint:
allow
anddisallow
options are not split into multiple options anymore.
The
summary
view has been added to/users
(GET/users?view=summary
)A new API for user’s services has been added:
- GET
/1.1/users/<user_id>/services
- GET
/1.1/users/<user_id>/services/<service_name>
- PUT
/1.1/users/<user_id>/services/<service_name>
- GET
A new API for user’s forwards has been added:
- GET
/1.1/users/<user_id>/forwards
- GET
/1.1/users/<user_id>/forwards/<forward_name>
- PUT
/1.1/users/<user_id>/forwards/<forward_name>
- GET
GET
/1.1/users/export
now requires the following header for CSV output:Accept: text/csv; charset=utf-8
Added call permissions endpoints:
- GET
/1.1/callpermissions
- POST
/1.1/callpermissions
- GET
/1.1/callpermissions/<callpermission_id>
- PUT
/1.1/callpermissions/<callpermission_id>
- DELETE
/1.1/callpermissions/<callpermission_id>
- GET
- Added switchboard endpoints:
- GET
/1.1/switchboards
- GET
/1.1/switchboards/<switchboard_id>/stats
- GET
- A new API for associating a line with a device has been added:
- PUT
/1.1/lines/<line_id>/devices/<device_id>
- DELETE
/1.1/lines/<line_id>/devices/<device_id>
- PUT
- The following URLs have been deleted. Please use the new API instead:
- GET
/1.1/devices/<device_id>/associate_line/<line_id>
- GET
/1.1/devices/<device_id>/dissociate_line/<line_id>
- GET
- Added users endpoints in REST API:
- GET
/1.1/users/<user_uuid>/lines/main/associated/endpoints/sip
- GET
- The SIP API has been improved.
options
now accepts any extra parameter. However, due to certain database limitations, parameters that appear in Supported parameters on SIP endpoints may only appear once in the list. This limitation will be removed in future versions. - A new API for custom endpoints has been added:
/1.1/endpoints/custom
- A new API for associating custom endpoints has been added:
/1.1/lines/<line_id>/endpoints/custom/<endpoint_id>
- A new API for mass updating users has been added: PUT
/1.1/users/import
- A new API for exporting users has been added: GET
/1.1/users/export
- A new API for mass importing users has been added: POST
/1.1/users/import
- The following fields have been added to the
/users
API:- supervision_enabled
- call_tranfer_enabled
- ring_seconds
- simultaneous_calls
- Ports 50050 and 50051 have been removed. Please use 9486 and 9487 instead
- Added sccp endpoints in REST API:
- GET
/1.1/endpoints/sccp
- POST
/1.1/endpoints/sccp
- DELETE
/1.1/endpoints/sccp/<sccp_id>
- GET
/1.1/endpoints/sccp/<sccp_id>
- PUT
/1.1/endpoints/sccp/<sccp_id>
- GET
/1.1/endpoints/sccp/<sccp_id>/lines
- GET
/1.1/lines/<line_id>/endpoints/sccp
- DELETE
/1.1/lines/<line_id>/endpoints/sccp/<sccp_id>
- PUT
/1.1/lines/<line_id>/endpoints/sccp/<sccp_id>
- GET
- Added lines endpoints in REST API:
- GET
/1.1/lines/<line_id>/users
- GET
- A new API for SIP endpoints has been added. Consult the documentation on http://api.xivo.io for further details.
- The
/lines_sip
API has been deprecated. Please use/lines
and/endpoints/sip
instead. - Due to certain limitations in the database, only a limited number of optional parameters can be configured. This limitation will be removed in future releases. Supported parameters are listed further down.
- Certain fields in the
/lines
API have been modified. List of fields are further down
/lines
API¶Name | Replaced by | Editable ? | Required ? |
---|---|---|---|
id | no | ||
device_id | no | ||
name | no | ||
protocol | no | ||
device_slot | position | no | |
provisioning_extension | provisioning_code | no | |
context | yes | yes | |
provisioning_code | yes | ||
position | yes | ||
caller_id_name | yes | ||
caller_id_num | yes |
- md5secret
- language
- accountcode
- amaflags
- allowtransfer
- fromuser
- fromdomain
- subscribemwi
- buggymwi
- call-limit
- callerid
- fullname
- cid-number
- maxcallbitrate
- insecure
- nat
- promiscredir
- usereqphone
- videosupport
- trustrpid
- sendrpid
- allowsubscribe
- allowoverlap
- dtmfmode
- rfc2833compensate
- qualify
- g726nonstandard
- disallow
- allow
- autoframing
- mohinterpret
- useclientcode
- progressinband
- t38pt-udptl
- t38pt-usertpsource
- rtptimeout
- rtpholdtimeout
- rtpkeepalive
- deny
- permit
- defaultip
- setvar
- port
- regexten
- subscribecontext
- fullcontact
- vmexten
- callingpres
- ipaddr
- regseconds
- regserver
- lastms
- parkinglot
- protocol
- outboundproxy
- transport
- remotesecret
- directmedia
- callcounter
- busylevel
- ignoresdpversion
- session-timers
- session-expires
- session-minse
- session-refresher
- callbackextension
- timert1
- timerb
- qualifyfreq
- contactpermit
- contactdeny
- unsolicited_mailbox
- use-q850-reason
- encryption
- snom-aoc-enabled
- maxforwards
- disallowed-methods
- textsupport
- The parameter
skip
is now deprecated. Useoffset
instead for:GET /1.1/devices
GET /1.1/extensions
GET /1.1/voicemails
GET /1.1/users
- The users resource can be referred to by
uuid
GET /1.1/users/<uuid>
PUT /1.1/users/<uuid>
DELETE /1.1/users/<uuid>
- The field
enabled
has been added to the voicemail model- A line is no longer required when associating a voicemail with a user
- Voicemails can now be edited even when they are associated to a user
- All optional fields on a user are now always null (sometimes they were empty strings)
- The caller id is no longer automatically updated when the firstname or lastname is modified. You must update the caller id yourself if you modify the user’s name.
- Caller id will be generated if and only if it does not exist when creating a user.
- Association user-voicemail, when associating a voicemail whose id does not exist:
- before: error 404
- after: error 400
- Association line-extension, a same extension can not be associated to multiple lines
- Resource line, field
provisioning_extension
: type changed fromint
tostring
API documentation is available on http://api.xivo.io. This section contains extended documentation for certain aspects of the API.
Function keys can be used as shortcuts for dialing a number, or accomplishing other menial tasks, by pushing a button on the phone. A function key’s action is determined by its destination.
Function keys can be added directly on a user, or in a template. Templates are useful for creating a set of common function keys that can be used by the same group of people.
This page only describes the data models used by the REST API. Consult the API documentation for further details on URLs.
Field | Type | Required | Description |
---|---|---|---|
name | string | No | A name for the template. |
keys | Function Key | No | A collection of function keys under the form {"position": "funckey"} .
See the example for more details. |
{
"name": "Example template",
"keys": {
"1": {
"destination": {
"type": "user",
"user_id": 34
}
},
"2": {
"blf": true,
"label": "Call mom",
"destination": {
"type": "custom",
"exten": "5551234567"
}
}
}
}
Field | Type | Required | Description |
---|---|---|---|
blf | boolean | No | Turn on BLF when there is activity on the destination |
label | string | No | Label to display next to the function key |
destination | Destination | Yes | Destination to call |
{
"blf": True,
"label": "Call john",
"destination": {
"type": "user",
"user_id": 34
}
}
A destination determines the number to dial when using a function key. Destinations are composed of a parameter named
type
and any additional parameters required by its type.
Available destination types:
- agent
- An agent
- bsfilter
- Boss/Secretary filter
- conference
- Conference room
- custom
- A custom number to dial
- forward
- Forward a call towards another number
- group
- A group
- onlinerec
- Record a conversation during a call
- paging
- A paging
- park
- Park a call
- park_position
- Pick up a parked call
- queue
- Call queue
- service
- A call service
- transfer
- Transfer a call
- user
- A User
Here are the parameters required for each destination:
Field | Type | Value |
---|---|---|
agent_id | numeric | Agents’s id |
Field | Type | Value |
---|---|---|
filter_member_id | numeric | ID of the filter member |
Field | Type | Value |
---|---|---|
conference_id | numeric | Conference’s id |
Field | Type | Value |
---|---|---|
exten | string | Number to dial |
Field | Type | Value |
---|---|---|
forward | string | Type of forward. Possible values: busy, noanswer, unconditional |
exten | string | Number to dial (optional) |
Field | Type | Value |
---|---|---|
group_id | numeric | Group’s id |
No parameters are required for this destination
Field | Type | Value |
---|---|---|
paging_id | numeric | Pagings’s id |
No parameters are required for this destination
Field | Type | Value |
---|---|---|
position | numeric string | Position of the parking to pick up |
Field | Type | Value |
---|---|---|
queue_id | numeric | User’s id |
Field | Type | Value |
---|---|---|
service | string | Name of the service |
Currently supported services:
- phonestatus
- Phone Status
- recsnd
- Sound Recording
- callrecord
- Call recording
- incallfilter
- Incoming call filtering
- enablednd
- Enable “Do not disturb” mode
- pickup
- Group Interception
- calllistening
- Listen to online calls
- directoryaccess
- Directory access
- fwdundoall
- Disable all forwaring
- enablevm
- Enable Voicemail
- vmusermsg
- Consult the Voicemail
- vmuserpurge
- Delete messages from voicemail
Field | Type | Value |
---|---|---|
transfer | string | Type of transfer. Possible values: blind, attended |
Field | Type | Value |
---|---|---|
user_id | numeric | User’s id |
Users and common related resources can be imported onto a XiVO server by sending a CSV file with a predefined set of fields.
This page only documents additional notes useful for API users. Consult the API documentation for more details.
Files may be uploaded as usual through the web interface, or from a console by using HTTP utilities and the REST API. When uploading through the API, the header Content-Type: text/csv charset=utf-8 must be set and the CSV data must be sent in the body of the request. A file may be uploaded using curl as follows:
curl -k -H "Content-Type: text/csv; charset=utf-8" -u username:password --data-binary "@file.csv" https://xivo:9486/1.1/users/import
The response can be reindented in a more readable format by piping the output through python -m json.tool in the following way:
curl (...) | python -m json.tool
The API version 1.0 is no longer supported and has been removed. In most cases, code that used the old API can be migrated to version 1.1 without much hassle by updating the URL. For example, in 1.0, the URL to list users was:
/1.0/users/
In 1.1, it is::
/1.1/users
Please note that there are no trailing slashes in URLs for version 1.1.
For further details consult the documentation at http://api.xivo.io
Note
The HTTP API 0.1 for xivo-ctid is currently evolving. New features and small fixes are regularly being added over time. We invite the reader to periodically check the changelog for an update on new features and changes.
API documentation is available on http://api.xivo.io.
Note
The HTTP API 1.0 for xivo-ctid-ng is currently evolving. New features and small fixes are regularly being added over time. We invite the reader to periodically check the changelog for an update on new features and changes.
A new API for getting the status of lines:
- GET
/1.0/lines/{id}/presences
- GET
A new API for checking the status of the daemon:
- GET
/1.0/status
- GET
A new API for updating user presences:
- GET
/1.0/users/{uuid}/presences
- PUT
/1.0/users/{uuid}/presences
- GET
/1.0/users/me/presences
- PUT
/1.0/users/me/presences
- GET
New APIs for listing and hanging up calls of a user:
- GET
/1.0/users/me/calls
- DELETE
/1.0/users/me/calls/{id}
- GET
New APIs for listing, cancelling and completing transfers of a user:
- GET
/1.0/users/me/transfers
- DELETE
/1.0/users/me/transfers/{transfer_id}
- PUT
/1.0/users/me/transfers/{transfer_id}/complete
- GET
POST
/1.0/users/me/transfers
may now return 403 status code.Originates (POST
/*/calls
) now return 400 if an invalid extension is given.
A new API for making calls from the authenticated user:
- POST
/1.0/users/me/calls
- POST
A new API for sending chat messages:
- POST
/1.0/chats
- POST
/1.0/users/me/chats
- POST
A new parameter for transfer creation (POST
/1.0/transfers
):variables
A new API for making transfers from the authenticated user:
- POST
/1.0/users/me/transfers
- POST
API documentation is available on http://api.xivo.io.
This section describes the REST API provided by the xivo-provd application.
If you want to interact with the REST API of the xivo-provd daemon that is executing as part of XiVO, you should be careful on which operation you are doing as to not cause stability problem to other parts of the XiVO ecosystem. Mostly, this means being careful when editing or deleting devices and configs.
By default, the REST API of xivo-provd is accessible only from localhost on port 8666. No authentication is required.
Warning
Major changes could happen to this API.
The description of the API has been split into these sections:
The provd manager resource represents the main entry point to the xivo-provd REST API.
It links to the following resources:
- The
dev
relation links to a device manager. - The
cfg
relation links to a config manager. - The
pg
relation links to a plugin manager. - The
srv.configure
relation links to the provd manager configuration service.
GET /provd
GET /provd HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"links": [
{
"href": "/provd/dev_mgr",
"rel": "dev"
},
{
"href": "/provd/cfg_mgr",
"rel": "cfg"
},
{
"href": "/provd/pg_mgr",
"rel": "pg"
},
{
"href": "/provd/configure",
"rel": "srv.configure"
}
]
}
The device manager links to the following resources:
- The
dev.synchronize
relation links to the device synchronization service. - The
dev.reconfigure
relation links to the device reconfiguration service. - The
dev.dhcpinfo
relation links to the device DHCP information service. - The
dev.devices
relation links to the list of devices.
GET /provd/dev_mgr
GET /provd/dev_mgr HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"links": [
{
"href": "/provd/dev_mgr/synchronize",
"rel": "dev.synchronize"
},
{
"href": "/provd/dev_mgr/reconfigure",
"rel": "dev.reconfigure"
},
{
"href": "/provd/dev_mgr/dhcpinfo",
"rel": "dev.dhcpinfo"
},
{
"href": "/provd/dev_mgr/devices",
"rel": "dev.devices"
}
]
}
GET /provd/dev_mgr/devices
Field | Description |
---|---|
q | A selector, encoded in JSON, describing which device should be returned. All devices are
returned if not specified. Example: q={"ip":"10.34.1.119"} |
fields | A list of fields, separated by comma. Example: fields=mac,ip |
skip | An integer specifing the number of devices to skip. Example: skip=10 |
sort | The key on which to sort the results. Example: sort=id |
sort_ord | The order of sort; either ASC or DESC . |
GET /provd/dev_mgr/devices HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"devices": [
{
"added": "auto",
"config": "38e5e08ffe804b468f5aa53b9536bb25",
"configured": true,
"description": "",
"id": "38e5e08ffe804b468f5aa53b9536bb25",
"ip": "10.34.1.122",
"mac": "00:08:5d:33:e5:76",
"model": "6731i",
"plugin": "xivo-aastra-3.3.1-SP2",
"remote_state_sip_username": "je5qtq",
"vendor": "Aastra",
"version": "3.3.1.2235"
}
]
}
POST /provd/dev_mgr/devices
POST /provd/dev_mgr/devices HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"device": {
"ip": "192.168.1.1",
"mac": "00:11:22:33:44:55",
"plugin": "xivo-aastra-3.3.1-SP2"
}
}
HTTP/1.1 201 Created
Content-Type: application/vnd.proformatique.provd+json
Location: /provd/dev_mgr/devices/68b10c99945b4fb889f22a7559fc3271
{"id": "68b10c99945b4fb889f22a7559fc3271"}
If the id
field is not given, then an ID is automatically generated by the server.
GET /provd/dev_mgr/devices/<device_id>
GET /provd/dev_mgr/devices/68b10c99945b4fb889f22a7559fc3271 HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"device": {
"added": "auto",
"config": "38e5e08ffe804b468f5aa53b9536bb25",
"configured": true,
"description": "",
"id": "38e5e08ffe804b468f5aa53b9536bb25",
"ip": "10.34.1.122",
"mac": "00:08:5d:33:e5:76",
"model": "6731i",
"plugin": "xivo-aastra-3.3.1-SP2",
"remote_state_sip_username": "je5qtq",
"vendor": "Aastra",
"version": "3.3.1.2235"
}
}
PUT /provd/dev_mgr/devices/<device_id>
PUT /provd/dev_mgr/devices/68b10c99945b4fb889f22a7559fc3271 HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"device": {
"added": "auto",
"config": "38e5e08ffe804b468f5aa53b9536bb25",
"configured": true,
"description": "",
"id": "38e5e08ffe804b468f5aa53b9536bb25",
"ip": "10.34.1.122",
"mac": "00:08:5d:33:e5:76",
"model": "6731i",
"plugin": "xivo-aastra-3.4",
"remote_state_sip_username": "je5qtq",
"vendor": "Aastra",
"version": "3.3.1.2235"
}
}
HTTP/1.1 204 No Content
DELETE /provd/dev_mgr/devices/<device_id>
DELETE /provd/dev_mgr/devices/68b10c99945b4fb889f22a7559fc3271 HTTP/1.1
Host: xivoserver
HTTP/1.1 204 No Content
POST /provd/dev_mgr/synchronize
POST /provd/dev_mgr/synchronize HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"id": "d035bccaf0dd4a8396fc57a3329ca0a4"
}
HTTP/1.1 201 Created
Location: /provd/dev_mgr/synchronize/42
The URI returned in the Location
header points to an operation in progress resource.
POST /provd/dev_mgr/reconfigure
Error code | Error message | Description |
---|---|---|
400 | invalid device ID |
POST /provd/dev_mgr/reconfigure HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"id": "d035bccaf0dd4a8396fc57a3329ca0a4"
}
HTTP/1.1 204 No Content
POST /provd/dev_mgr/dhcpinfo
POST /provd/dev_mgr/dhcpinfo HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"dhcp_info": {
"ip": "192.168.1.100",
"mac": "00:11:22:33:44:55",
"op": "commit",
"options": [
"06066.6f.6f.62.61.72.a"
]
}
}
HTTP/1.1 204 No Content
The config manager links to the following resources:
- The
cfg.configs
relation links to the list of configs. - The
cfg.autocreate
relation links to the config autocreate service.
GET /provd/cfg_mgr
GET /provd/cfg_mgr HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"links": [
{
"href": "/provd/cfg_mgr/configs",
"rel": "cfg.configs"
},
{
"href": "/provd/cfg_mgr/autocreate",
"rel": "cfg.autocreate"
}
]
}
GET /provd/cfg_mgr/configs
These are the same parameters as for the list devices action.
GET /provd/cfg_mgr/configs HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"configs": [
{
"configdevice": "defaultconfigdevice",
"deletable": true,
"id": "38e5e08ffe804b468f5aa53b9536bb25",
"parent_ids": [
"base",
"defaultconfigdevice"
],
"raw_config": {
"X_key": "",
"exten_dnd": "*25",
"exten_fwd_busy": "*23",
"exten_fwd_disable_all": "*20",
"exten_fwd_no_answer": "*22",
"exten_fwd_unconditional": "*21",
"exten_park": null,
"exten_pickup_call": "*8",
"exten_pickup_group": null,
"exten_voicemail": "*98",
"funckeys": {
"1": {
"label": "",
"line": 1,
"type": "speeddial",
"value": "1005"
}
},
"protocol": "SIP",
"sip_dtmf_mode": "SIP-INFO",
"sip_lines": {
"1": {
"auth_username": "je5qtq",
"display_name": "El\u00e8s 01",
"number": "1001",
"password": "T2S7C0",
"proxy_ip": "10.34.1.11",
"registrar_ip": "10.34.1.11",
"username": "je5qtq"
}
}
}
}
]
}
POST /provd/cfg_mgr/configs
POST /provd/cfg_mgr/configs HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"config": {
"parent_ids": [
"base"
],
"raw_config": {
"sip": {
"lines": {
"1": {
"auth_username": "100",
"display_name": "Foo",
"password": "100",
"username": "100"
}
}
}
}
}
}
HTTP/1.1 201 Created
Content-Type: application/vnd.proformatique.provd+json
Location: /provd/cfg_mgr/configs/77839d0f05c84662864b0ae5c27b33e4
{"id": "77839d0f05c84662864b0ae5c27b33e4"}
If the id
field is not given, then an ID id automatically generated by the server.
GET /provd/cfg_mgr/configs/<config_id>
GET /provd/cfg_mgr/configs/77839d0f05c84662864b0ae5c27b33e4 HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"config": {
"id": "77839d0f05c84662864b0ae5c27b33e4",
"parent_ids": [
"base"
],
"raw_config": {
"sip": {
"lines": {
"1": {
"auth_username": "100",
"display_name": "Foo",
"password": "100",
"username": "100"
}
}
}
}
}
}
GET /provd/cfg_mgr/configs/<config_id>/raw
GET /provd/cfg_mgr/configs/77839d0f05c84662864b0ae5c27b33e4/raw HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"raw_config": {
"X_xivo_phonebook_ip": "10.34.1.11",
"http_port": 8667,
"ip": "10.34.1.11",
"ntp_enabled": true,
"ntp_ip": "10.34.1.11",
"sip": {
"lines": {
"1": {
"auth_username": "100",
"display_name": "John",
"password": "100",
"username": "100"
}
}
},
"tftp_port": 69
}
}
PUT /provd/cfg_mgr/configs/<config_id>
PUT /provd/cfg_mgr/configs/77839d0f05c84662864b0ae5c27b33e4 HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"config": {
"id": "77839d0f05c84662864b0ae5c27b33e4",
"parent_ids": [
"base"
],
"raw_config": {
"sip": {
"lines": {
"1": {
"auth_username": "100",
"display_name": "John",
"password": "100",
"username": "100"
}
}
}
}
}
}
HTTP/1.1 204 No Content
DELETE /provd/cfg_mgr/configs/<config_id>
DELETE /provd/cfg_mgr/configs/77839d0f05c84662864b0ae5c27b33e4
Host: xivoserver
HTTP/1.1 204 No Content
This service is used to create a new config from the config that has the autocreate
role.
POST /provd/cfg_mgr/autocreate
POST /provd/cfg_mgr/autocreate HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{}
HTTP/1.1 201 Created
Content-Type: application/vnd.proformatique.provd+json
Location: /provd/cfg_mgr/configs/autoprov1411400365
{"id":"autoprov1411400365"}
The plugin manager links to the following resources:
- The
srv.install
relation links to the plugin manager installation service. This installation service permits installing/uninstalling plugins. - The
pg.plugins
relation links to the list of plugins. - The
pg.reload
relation links to the plugin reload service.
GET /provd/pg_mgr
GET /provd/pg_mgr HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"links": [
{
"href": "/provd/pg_mgr/install",
"rel": "srv.install"
},
{
"href": "/provd/pg_mgr/plugins",
"rel": "pg.plugins"
},
{
"href": "/provd/pg_mgr/reload",
"rel": "pg.reload"
}
]
}
List the installed plugins.
If you want to install/uninstall plugins, you need to go trough the plugin installation service.
GET /provd/pg_mgr/plugins
GET /provd/pg_mgr/plugins HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"plugins": {
"xivo-aastra-3.3.1-SP2": {
"links": [
{
"href": "/provd/pg_mgr/plugins/xivo-aastra-3.3.1-SP2",
"rel": "pg.plugin"
}
]
},
"xivo-cisco-sccp-9.0.3": {
"links": [
{
"href": "/provd/pg_mgr/plugins/xivo-cisco-sccp-9.0.3",
"rel": "pg.plugin"
}
]
}
}
}
The plugin links to the following resources:
- The
pg.info
relation links to the plugin information. - The
srv.install
relation links to the plugin installation service. Plugins usually provided this service to install/uninstall firmware and language files.
GET /provd/pg_mgr/plugins/<plugin_id>
GET /provd/pg_mgr/plugins/xivo-aastra-3.3.1-SP2 HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"links": [
{
"href": "/provd/pg_mgr/plugins/xivo-aastra-3.3.1-SP2/info",
"rel": "pg.info"
},
{
"href": "/provd/pg_mgr/plugins/xivo-aastra-3.3.1-SP2/install",
"rel": "srv.install"
}
]
}
GET /provd/pg_mgr/plugins/<plugin_id>/info
GET /provd/pg_mgr/plugins/xivo-aastra-3.3.1-SP2/info HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"plugin_info": {
"capabilities": {
"Aastra, 6730i, 3.3.1.5089": {
"sip.lines": 6
},
"Aastra, 6731i, 3.3.1.2235": {
"sip.lines": 6,
"switchboard": true
},
"Aastra, 6735i, 3.3.1.5089": {
"sip.lines": 9
},
"Aastra, 6737i, 3.3.1.5089": {
"sip.lines": 9
},
"Aastra, 6739i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 6753i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 6755i, 3.3.1.2235": {
"sip.lines": 9,
"switchboard": true
},
"Aastra, 6757i, 3.3.1.2235": {
"sip.lines": 9,
"switchboard": true
},
"Aastra, 9143i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 9480i, 3.3.1.2235": {
"sip.lines": 9
}
},
"description": "Plugin for Aastra 6730i, 6731i, 6735i, 6737i, 6739i, 6753i, 6755i, 6757i, 6757i CT, 9143i, 9480i, 9480i CT in version 3.3.1 SP2.",
"version": "1.1"
}
}
Reload the given plugin. This is mostly useful during plugin development, after changing the code of the plugin, instead of restarting the xivo-provd application.
POST /provd/pg_mgr/reload
POST /provd/pg_mgr/reload HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"id": "xivo-aastra-3.3.1-SP2"
}
HTTP/1.1 204 No Content
This section describes the resources that are available from more than one URI or are generic enough to not fit in a more specific section.
This resource represents an operation in progress and is used to follow the progress of an underlying operation. Said differently, it is a monitor on an operation that can change over time.
GET <uri>
GET <uri> HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"status": "progress"
}
The status
field describe the current status of the operation. The format is
[label|]state[;current[/end]](\(sub_oips\))*
. Here’s some examples:
- progress
- download|progress
- download|progress;10
- download|progress;10/100
- download|progress(file_1|progress;20/100)(file_2|waiting;0/50)
- download|progress;20/150(file_1|progress)(file_2|waiting)
- op|progress(op1|progress(op11|progress)(op12|waiting))(op2|progress)
The state of an operation is either waiting
, progress
, success
or fail
.
Delete the “operation in progress” resource.
This does not cancel the underlying operation; it only deletes the monitor. Every monitor that is created should be deleted, else they won’t be freed by the process and they will accumulate, taking memory.
DELETE <uri>
DELETE <uri> HTTP/1.1
Host: xivoserver
HTTP/1.1 204 No Content
GET <uri>
Example request for the configuration service of the provd manager.
GET /provd/configure HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"params": [
{
"description": "The plugins repository URL",
"id": "plugin_server",
"links": [
{
"href": "/provd/configure/plugin_server",
"rel": "srv.configure.param"
}
],
"value": "http://provd.xivo.solutions/plugins/1/stable"
},
{
"description": "The proxy for HTTP requests. Format is \"http://[user:password@]host:port\"",
"id": "http_proxy",
"links": [
{
"href": "/provd/configure/http_proxy",
"rel": "srv.configure.param"
}
],
"value": null
},
{
"description": "The proxy for FTP requests. Format is \"http://[user:password@]host:port\"",
"id": "ftp_proxy",
"links": [
{
"href": "/provd/configure/ftp_proxy",
"rel": "srv.configure.param"
}
],
"value": null
},
{
"description": "The proxy for HTTPS requests. Format is \"host:port\"",
"id": "https_proxy",
"links": [
{
"href": "/provd/configure/https_proxy",
"rel": "srv.configure.param"
}
],
"value": null
},
{
"description": "The current locale. Example: fr_FR",
"id": "locale",
"links": [
{
"href": "/provd/configure/locale",
"rel": "srv.configure.param"
}
],
"value": null
},
{
"description": "Set to 1 if all the devices are behind a NAT.",
"id": "NAT",
"links": [
{
"href": "/provd/configure/NAT",
"rel": "srv.configure.param"
}
],
"value": 0
}
]
}
GET <uri>
Example request for the NAT option of the configuration service of the provd entry point.
GET /provd/configure/NAT HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"param": {
"value": 0
}
}
PUT <uri>
Example request for the NAT option of the configuration service of the provd manager.
PUT /provd/configure/NAT HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"param": {
"value": 1
}
}
HTTP/1.1 204 No Content
Content-Type: application/vnd.proformatique.provd+json
GET <uri>
Example request for the installation service of the plugin manager.
GET /provd/pg_mgr/install HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"links": [
{
"href": "/provd/pg_mgr/install/install",
"rel": "srv.install.install"
},
{
"href": "/provd/pg_mgr/install/uninstall",
"rel": "srv.install.uninstall"
},
{
"href": "/provd/pg_mgr/install/installed",
"rel": "srv.install.installed"
},
{
"href": "/provd/pg_mgr/install/installable",
"rel": "srv.install.installable"
},
{
"href": "/provd/pg_mgr/install/upgrade",
"rel": "srv.install.upgrade"
},
{
"href": "/provd/pg_mgr/install/update",
"rel": "srv.install.update"
}
]
}
The upgrade and update services are optional and not all installation service provide them.
POST <uri>
Example request for the installation service of the plugin manager.
POST /provd/pg_mgr/install/install HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"id": "xivo-polycom-4.0.4"
}
HTTP/1.1 201 Created
Location: /provd/pg_mgr/install/install/1
Content-Type: application/vnd.proformatique.provd+json
The URI returned in the Location
header points to an operation in progress resource.
POST <uri>
Example request for the installation service of the plugin manager.
POST /provd/pg_mgr/install/uninstall HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"id": "xivo-polycom-4.0.4"
}
HTTP/1.1 204 No Content
Content-Type: application/vnd.proformatique.provd+json
POST <uri>
Example request for the installation service of the plugin manager.
POST /provd/pg_mgr/install/upgrade HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{
"id": "xivo-polycom-4.0.4"
}
HTTP/1.1 201 Created
Location: /provd/pg_mgr/install/upgrade/1
Content-Type: application/vnd.proformatique.provd+json
The URI returned in the Location
header points to an operation in progress resource.
POST <uri>
Example request for the installation service of the plugin manager.
POST /provd/pg_mgr/install/update HTTP/1.1
Host: xivoserver
Content-Type: application/vnd.proformatique.provd+json
{}
HTTP/1.1 201 Created
Location: /provd/pg_mgr/install/update/1
Content-Type: application/vnd.proformatique.provd+json
The URI returned in the Location
header points to an operation in progress resource.
GET <uri>
Example request for the installation service of the plugin manager.
GET /provd/pg_mgr/install/installable HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"pkgs": {
"null": {
"capabilities": {
"*, *, *": {
"sip.lines": 0
}
},
"description": "Plugin that offers no configuration service and rejects TFTP/HTTP requests.",
"dsize": 1073,
"sha1sum": "90b2fb6c2b135a9d539488b6a85779dd95e0e876",
"version": "1.0"
},
"xivo-aastra-3.3.1-SP2": {
"capabilities": {
"Aastra, 6730i, 3.3.1.5089": {
"sip.lines": 6
},
"Aastra, 6731i, 3.3.1.2235": {
"sip.lines": 6,
"switchboard": true
},
"Aastra, 6735i, 3.3.1.5089": {
"sip.lines": 9
},
"Aastra, 6737i, 3.3.1.5089": {
"sip.lines": 9
},
"Aastra, 6739i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 6753i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 6755i, 3.3.1.2235": {
"sip.lines": 9,
"switchboard": true
},
"Aastra, 6757i, 3.3.1.2235": {
"sip.lines": 9,
"switchboard": true
},
"Aastra, 9143i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 9480i, 3.3.1.2235": {
"sip.lines": 9
}
},
"description": "Plugin for Aastra 6730i, 6731i, 6735i, 6737i, 6739i, 6753i, 6755i, 6757i, 6757i CT, 9143i, 9480i, 9480i CT in version 3.3.1 SP2.",
"dsize": 9397,
"sha1sum": "68dbed6afa87cf624a89166bdc6bdf7413cb84df",
"version": "1.1"
}
}
}
GET <uri>
Example request for the installation service of the plugin manager.
GET /provd/pg_mgr/install/installed HTTP/1.1
Host: xivoserver
Accept: application/vnd.proformatique.provd+json
HTTP/1.1 200 OK
Content-Type: application/vnd.proformatique.provd+json
{
"pkgs": {
"xivo-aastra-3.3.1-SP2": {
"capabilities": {
"Aastra, 6730i, 3.3.1.5089": {
"sip.lines": 6
},
"Aastra, 6731i, 3.3.1.2235": {
"sip.lines": 6,
"switchboard": true
},
"Aastra, 6735i, 3.3.1.5089": {
"sip.lines": 9
},
"Aastra, 6737i, 3.3.1.5089": {
"sip.lines": 9
},
"Aastra, 6739i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 6753i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 6755i, 3.3.1.2235": {
"sip.lines": 9,
"switchboard": true
},
"Aastra, 6757i, 3.3.1.2235": {
"sip.lines": 9,
"switchboard": true
},
"Aastra, 9143i, 3.3.1.2235": {
"sip.lines": 9
},
"Aastra, 9480i, 3.3.1.2235": {
"sip.lines": 9
}
},
"description": "Plugin for Aastra 6730i, 6731i, 6735i, 6737i, 6739i, 6753i, 6755i, 6757i, 6757i CT, 9143i, 9480i, 9480i CT in version 3.3.1 SP2.",
"version": "1.1"
}
}
}
This service provides a public API that can be used to change the configuration that are on a XiVO.
Warning
The 0.1 API is currently in development. Major changes could still happen and new resources will be added over time.
GET /delete_voicemail
- name
- the voicemail name
- context
- the voicemail context (default is ‘default’)
Error code | Error message | Description |
---|---|---|
404 | Not found | The voicemail does not exist |
GET /delete_voicemail HTTP/1.1
Host: xivoserver
Accept: application/json
HTTP/1.1 200 OK
Content-Type: application/json
{
nothing
}
GET /discover_netifaces
GET /discover_netifaces HTTP/1.1
Host: xivoserver
Accept: application/json
HTTP/1.1 200 OK
Content-Type: application/json
{
"lo":
{
"hwaddress": "00:00:00:00:00:00",
"typeid": 24,
"alias-raw-device": null,
"network": "127.0.0.0",
"family": "inet",
"physicalif": false,
"vlan-raw-device": null,
"vlanif": false,
"dummyif": false,
"mtu": 65536,
"broadcast": "127.255.255.255",
"hwtypeid": 772,
"netmask": "255.0.0.0",
"carrier": true,
"flags": 9,
"address": "127.0.0.1",
"vlan-id": null,
"type": "loopback",
"options": null,
"aliasif": false,
"name": "lo"
},
"eth0":
{
"alias-raw-device": null,
"family": "inet",
"hwaddress": "36:76:70:29:69:c2",
"vlan-id": null,
"network": "172.17.0.0",
"physicalif": false,
"vlan-raw-device": null,
"vlanif": false,
"type": "eth",
"aliasif": false,
"broadcast": "172.17.255.255",
"netmask": "255.255.0.0",
"address": "172.17.0.101",
"typeid": 6,
"name": "eth0",
"hwtypeid": 1,
"dummyif": false,
"mtu": 1500,
"carrier": true,
"flags": 3,
"options": null
}
}
GET /netiface/<interface>
GET /netiface/eth0 HTTP/1.1
Host: xivoserver
Content-Type: application/json
HTTP/1.1 200 OK
Content-Type: application/json
{
"eth0":
{
"alias-raw-device": null,
"family": "inet",
"hwaddress": "36:76:70:29:69:c2",
"vlan-id": null,
"network": "172.17.0.0",
"physicalif": false,
"vlan-raw-device": null,
"vlanif": false,
"type": "eth",
"aliasif": false,
"broadcast": "172.17.255.255",
"netmask": "255.255.0.0",
"address": "172.17.0.101",
"typeid": 6,
"name": "eth0",
"hwtypeid": 1,
"dummyif": false,
"mtu": 1500,
"carrier": true,
"flags": 3,
"options": null
}
}
Field | Values | Description |
---|---|---|
iface | string | Interface name like eth0 |
method | list | static or dhcp |
address | string | |
netmask | string | |
broadcast | string | |
gateway | string | |
mtu | int | |
auto | boolean | |
up | boolean | |
options | list | dns-search and dns-nameservers |
PUT /modify_physical_eth_ipv4
PUT /modify_physical_eth_ipv4 HTTP/1.1
Host: xivoserver
Content-Type: application/json
{
"ifname': "eth0",
"method': "dhcp",
"auto": "True"
}
PUT /replace_virtual_eth_ipv4
PUT /replace_virtual_eth_ipv4 HTTP/1.1
Host: xivoserver
Content-Type: application/json
{
"ifname": "eth0:0",
"new_ifname": "eth0:1",
"method": "dhcp",
"auto": "True"
}
PUT /modify_eth_ipv4
PUT /modify_eth_ipv4 HTTP/1.1
Host: xivoserver
Content-Type: application/json
{
'ifname' : 'eth0'
'address': '192.168.0.1',
'netmask': '255.255.255.0',
'broadcast': '192.168.0.255',
'gateway': '192.168.0.254',
'mtu': 1500,
'auto': True,
'up': True,
'options': [['dns-search', 'toto.tld tutu.tld'],
['dns-nameservers', '127.0.0.1 192.168.0.254']]
}
For other services, see http://api.xivo.io. This public instance does not allow you to directly test the requests (i.e. the “Try it out!” button will not work), but you may use the embedded version of your XiVO, where this button will work.
Every XiVO server embeds its own copy of the Swagger UI exposed on http://api.xivo.io. The instance embedded in the XiVO allows you to directly try the requests with the in-page buttons.
For the rest of this article, we will consider that your XiVO is accessible under the hostname
MY_XIVO
.
The instance is available at: http://MY_XIVO/api
Before using the Swagger UI, there are a few prerequisites:
- Accept the HTTPS certificate for each service of the XiVO
- Add the permissions to use the REST API to a Web Services Access user
- Obtain an authentication token
For each service on the left menu that you want to try, you need to accept the HTTPS certificate for this service. To that end:
- click on the service in the menu on the left
- copy the URL you see in the text box at the top of the page, something like:
https://MY_XIVO:9497/0.1/api/api.json
and paste it in your browser - accept the HTTPS certificate validation exception
- go back to http://MY_XIVO/api and select the service again (or click on the top-right “Explore” button)
You should now be able to see the different sections for the REST API of that service.
You must create a Web Services Access with the right permissions before using the REST API. See Web Services Access.
Each endpoint has its own ACL, but you may add wildcard ACLs, like:
auth.#
to gain access to allxivo-auth
REST API endpointsconfd.#
to gain access to allxivo-confd
REST API endpoints#
to gain access to every endpoint of every service.
Warning
Only use wildcards when doing tests, not with a production REST API access. You should always restrict the permissions to the bare minimum.
The quick and easy way is to use http://auth.xivo.io. You may log-in with the following parameters:
- Host = https://MY_XIVO:9497 (you must have accepted the HTTPS certificate of
xivo-auth
first) - Backend = XIVO Service
- Login = username of your Web Services Access
- Password = the associated password
Then click “Sign in!”, and you can get see the token. This token will expire after one hour, and you will need to re-authenticate to get a new token.
The other way you can get a token is via Swagger UI (what else?). Choose the xivo-auth
service
in the list of REST API. Under tokens
, choose POST /tokens
.
- In the top-right text box of the page (left to the “Explore” button), fill “token” with the
string
username:password
where those credentials come from the Web Services Access you created earlier. - Go back to the
POST /tokens
section and click on the yellow box to the right of thebody
parameter. This will pre-fill thebody
parameter. - In the
body
parameter, set:backend
toxivo-service
expiration
to the number of seconds for the token to be valid (e.g. 60 for one hour). After the expiration time, you will need to re-authenticate to get a new token.
- Click “Try it out” at the end of the section
- In the response, you should see a
token
attribute.
For more informations about the backends of xivo-auth, see xivo-auth plugins.
To use the authentication token, choose the service for which you want to try the REST API, then paste the token in the top-right text box. You do not need to click “Explore” to apply the token change, the new token will be used automatically at the next request you send.
You can now choose a REST API endpoint and “Try it out”.
Each REST API is available via HTTPS on different ports.
# Get the list of users
curl --insecure \
-H 'Accept: application/json' \
-H 'X-Auth-Token: 17496bfa-4653-9d9d-92aa-17def0fa9826' \
https://xivo:9486/1.1/users
# Create a user
# When sending data, you need the Content-Type header.
curl --insecure \
-X POST \
-d '{"firstname": "hello-world"} \
-H 'Accept: application/json' \
-H 'Content-Type: application/json' \
-H 'X-Auth-Token: 17496bfa-4653-9d9d-92aa-17def0fa9826' \
https://xivo:9486/1.1/users
For all REST APIs, the main way to authenticate is to use an access token obtained from
xivo-auth. This token should be given in the X-Auth-Token
header in your request. For example:
curl <options...> -H 'X-Auth-Token: 17496bfa-4653-9d9d-92aa-17def0fa9826' https://<xivo_address>:9486/1.1/users
Also, your token needs to have the right ACLs to give you access to the resource you want. See REST API Permissions.
The tokens delivered by xivo-auth have a list of permissions associated (ACL), that determine which REST resources are authorized for this token. Each REST resource has an associated required ACL. When you try to access to a REST resource, this resource requests xivo-auth with your token and the required ACL to validate the access.
An ACL contains 3 parts separated by dot (.
)
service: name of service, without prefix
xivo-
(e.g.xivo-confd
->confd
).resource: name of resource separated by dot (
.
) (e.g./users/17/lines
->users.17.lines
).action: action performed on resource. Generally, this is the following schema:
get
->read
put
->update
post
->create
delete
->delete
There are 3 substitution values for an ACL.
*
: replace only one word between dot.#
: replace one or multiple words.me
: replace theuser_uuid
from sent token.
The ACL confd.users.me.#.read
will have access to the following REST resources:
GET /users/{user_id}/cti
GET /users/{user_id}/funckeys
GET /users/{user_id}/funckeys/{position}
GET /users/{user_id}/funckeys/templates
GET /users/{user_id}/lines
GET /users/{user_id}/lines/{line_id}
GET /users/{user_id}/voicemail
- service:
confd
- resource:
users.me.#
- action:
read
The ACL confd.users.me.funckeys.*.*
will have access to the following REST resources:
DELETE /users/{user_id}funckeys/{position}
GET /users/{user_id}funckeys/{position}
PUT /users/{user_id}funckeys/{position}
GET /users/{user_id}funckeys/templates
- service:
confd
- resource:
users.me.funckeys.*
- action:
*
Where {user_id}
is the user uuid from the token.
The ACL corresponding to each resource is documented in http://auth.xivo.io. Some resources may not
have any associated ACL yet, so you must use {service}.#
instead.
See also Service Authentication for details about the token-based authentication process.
Warning
DEPRECATED
For compatibility reason, xivo-confd may accept requests without an access token. For this, you must create a webservices user in the web interface (
):if an IP address is specified for the user, no authentication is needed
if you choose not to specify an IP address for the user, you can connect to the REST API with a HTTP Digest authentication, using the user name and password you provided. For instance, the following command line allows to retrieve XiVO users through the REST API, using the login admin and the password passadmin:
curl <options...> --digest --cookie '' -u admin:passadmin https://<xivo_address>:9486/1.1/users
Standard HTTP status codes are used. For the full definition see IANA definition.
- 200: Success
- 201: Created
- 400: Incorrect syntax
- 404: Resource not found
- 406: Not acceptable
- 412: Precondition failed
- 415: Unsupported media type
- 500: Internal server error
See also Errors for general explanations about error codes.
Example usage of general parameters:
GET http://<xivo_address>:9486/1.1/voicemails?limit=X&offset=Y
- order
- Sort the list using a column (e.g. “number”). See specific resource documentation for columns allowed.
- direction
- ‘asc’ or ‘desc’. Sort list in ascending (asc) or descending (desc) order
- limit
- total number of resources to show in the list. Must be a positive integer
- offset
- number of resources to skip over before starting the list. Must be a positive integer
- search
- Search resources. Only resources with a field containing the search term will be listed.
JSON is used to encode returned or sent data. Therefore, the following headers are needed:
- when the request is supposed to return JSON:
Accept = application/json
- when the request’s body contains JSON:
Content-Type = application/json
Note
Optional properties can be added without changing the protocol version in the main list or in the object list itself. Properties will not be removed, type and name will not be modified.
GET /1.1/objects
- When returning lists the format is as follows:
- total - number of items in total in the system configuration (optional)
- items - returned data as an array of object properties list.
Other optional properties can be added later.
Response data format
{
"total": 2,
"items":
[
{
"id": "1",
"prop1": "test"
},
{
"id": "2",
"prop1": "ssd"
}
]
}
Format returned is a list of properties. The object should always have the same attributes set, the default value being the equivalent to NULL in the content-type format.
GET /1.1/objects/<id>
Response data format
{
"id": "1",
"prop1": "test"
}
The XiVO REST server implements POST and PUT methods for item creation and update respectively. Data is created using the POST method via a root URL and is updated using the PUT method via a root URL suffixed by /<id. The server expects to receive JSON encoded data. Only one item can be processed per request. The data format and required data fields are illustrated in the following example:
Request data format
{
"id": "1",
"prop1": "test"
}
When updating, only the id and updated properties are needed, omitted properties are not updated. Some properties can also be optional when creating an object.
A request to the web services may return an error. An error will always be associated to an HTTP error code, and eventually to one or more error messages. The following errors are common to all web services:
Error code | Error message | Description |
---|---|---|
406 | empty | Accept header missing or contains an unsupported content type |
415 | empty | Content-Type header missing or contains an unsupported content type |
500 | list of errors | An error occured on the server side; the content of the message depends of the type of errors which occured |
The 400, 404 and 412 errors depend on the web service you are requesting. They are separately described for each of them.
The error messages are contained in a JSON list, even if there is only one error message:
[ message_1, message_2, ... ]
Subroutine¶
The preprocess subroutine allows you to enhance XiVO features through the Asterisk dialplan. Features that can be enhanced are :
- User
- Group
- Queue
- Meetme
- Incoming call
- Outgoing call
There are three possible categories :
- Subroutine for one feature
- Subroutine for global forwarding
- Subroutine for global incoming call to an object
Subroutines are called at the latest possible moment in the dialplan, so that the maximum of variables have already been set: this way, the variables can be read and modified at will before they are used.
Here is an example of the dialplan execution flow when an external incoming call to a user being forwarded to another external number (like a forward to a mobile phone):
If you want to add a new subroutine, we propose to edit a new configuration file in the directory /etc/asterisk/extensions_extra.d
.
You can also add this file by the web interface.
An example:
[myexample]
exten = s,1,NoOp(This is an example)
same = n,Return()
Subroutines should always end with a Return()
. You may replace Return()
by a Goto()
if
you want to completely bypass the XiVO dialplan, but this is not recommended.
To plug your subroutine into the XiVO dialplan, you must add myexample
in the subroutine field
in the web interface, e.g. .
There is predefined subroutine for this feature, you can find the name and the activation in the /etc/xivo/asterisk/xivo_globals.conf
.
The variables are:
; Global Preprocess subroutine
XIVO_PRESUBR_GLOBAL_ENABLE = 1
XIVO_PRESUBR_GLOBAL_USER = xivo-subrgbl-user
XIVO_PRESUBR_GLOBAL_AGENT = xivo-subrgbl-agent
XIVO_PRESUBR_GLOBAL_GROUP = xivo-subrgbl-group
XIVO_PRESUBR_GLOBAL_QUEUE = xivo-subrgbl-queue
XIVO_PRESUBR_GLOBAL_MEETME = xivo-subrgbl-meetme
XIVO_PRESUBR_GLOBAL_DID = xivo-subrgbl-did
XIVO_PRESUBR_GLOBAL_OUTCALL = xivo-subrgbl-outcall
XIVO_PRESUBR_GLOBAL_PAGING = xivo-subrgbl-paging
So if you want to add a subroutine for all of your XiVO users you can do this:
[xivo-subrgbl-user]
exten = s,1,NoOp(This is an example for all my users)
same = n,Return()
You can also use a global subroutine for call forward.
; Preprocess subroutine for forwards
XIVO_PRESUBR_FWD_ENABLE = 1
XIVO_PRESUBR_FWD_USER = xivo-subrfwd-user
XIVO_PRESUBR_FWD_GROUP = xivo-subrfwd-group
XIVO_PRESUBR_FWD_QUEUE = xivo-subrfwd-queue
XIVO_PRESUBR_FWD_MEETME = xivo-subrfwd-meetme
XIVO_PRESUBR_FWD_VOICEMAIL = xivo-subrfwd-voicemail
XIVO_PRESUBR_FWD_SCHEDULE = xivo-subrfwd-schedule
XIVO_PRESUBR_FWD_VOICEMENU = xivo-subrfwd-voicemenu
XIVO_PRESUBR_FWD_SOUND = xivo-subrfwd-sound
XIVO_PRESUBR_FWD_CUSTOM = xivo-subrfwd-custom
XIVO_PRESUBR_FWD_EXTENSION = xivo-subrfwd-extension
Some of the XiVO variables can be used and modified in subroutines (non exhaustive list):
XIVO_CALLOPTIONS
: the value is a list of options to be passed to the Dial application, e.g.hHtT
. This variable is available in agent, user and outgoing call subroutines. Please note that it may not be set earlier, because it will be overwritten.XIVO_CALLORIGIN
: the value is:extern
for calls coming from a DIDintern
for all other calls
This variable is used by xivo-agid when selecting the ringtone for ringing a user. This variable is available only in user subroutines.
XIVO_DSTNUM
: the value is the extension dialed, as received by XiVO (e.g. an internal extension, a DID, or an outgoing extension including the local prefix). This variable is available in all subroutines.XIVO_GROUPNAME
: the value is the name of the group being called. This variable is only available in group subroutines.XIVO_GROUPOPTIONS
: the value is a list of options to be passed to the Queue application, e.g.hHtT
. This variable is only available in group subroutines.XIVO_INTERFACE
: the value is the Technology/Resource pairs that are used as the first argument of the Dial application. This variable is only available in the user subroutines.XIVO_MOBILEPHONENUMBER
: the value is the phone number of a user, as set in the web interface. This variable is only available in user subroutines.XIVO_QUEUENAME
: the value is the name of the queue being called. This variable is only available in queue subroutines.XIVO_QUEUEOPTIONS
: the value is a list of options to be passed to the Queue application, e.g.hHtT
. This variable is only available in queue subroutines.XIVO_SRCNUM
: the value is the callerid number of the originator of the call: the internal extension of a user (outgoing callerid is ignored), or the public extension of an external incoming call. This variable is available in all subroutines.
WebSocket Event Service¶
XiVO offers a service to receive messages published on the bus (e.g. RabbitMQ) over an encrypted WebSocket connection. This ease in building dynamic web applications that are using events from your XiVO.
The service is provided by the xivo-websocketd
component.
First, you need a XiVO in version 16.03 or later.
Then, to use the service, you need to:
- connect to it on port 9502 using an encrypted WebSocket connection.
- authenticate to it by providing a xivo-auth token that has the
websocketd
ACL. If you don’t know how to obtain a xivo-auth token from your XiVO, consult the documentation on xivo-auth.
For example, if you want to use the service located at example.org
with the token
some-token-id
, you would use the URL wss://example.org:9502/?token=some-token-id
.
The SSL/TLS certificate that is used by the WebSocket server is the same
as the one used by the XiVO web interface and the REST APIs. By default, this is a self-signed
certificate, and web browsers will prevent connections from being successfully established for
security reasons. On most web browsers, this can be circumvented by first visiting the
https://<xivo-ip>:9502/
URL and adding a security exception. Other solutions to this problem are
described in the connection section.
After a succesful connection and authentication to the service, the server will send the following message:
{"op": "init", "code": 0, "msg": ""}
This indicate that the server is ready to accept more commands from the client. Had an error happened, the server would have closed the connection, possibly with one of the service specific WebSocket close code.
The message you see is part of the small JSON-based protocol that is used for the client/server interaction.
To receive events on your WebSocket connection, you need to tell the server which type of events you are interested in, and then tell it to start sending you these events. For example, if you are interested in the “endpoint_status_update” events, you send the following command:
{"op": "subscribe", "data": {"event_name": "endpoint_status_update"}}
If all goes well, the server will respond with:
{"op": "subscribe", "code": 0, "msg": ""}
Once you have subscribed to all the events you are interested in, you ask the server to start sending you the matching events by sending the following command:
{"op": "start"}
The server will respond with:
{"op": "start", "code": 0, "msg": ""}
Once you have received this message, all the other messages you’ll receive will be messages originating from the bus, in the same format as they were on the bus.
Here’s a rudimentary example of a web page accessing the service:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 | <!DOCTYPE html>
<html>
<head>
<meta charset="utf-8">
<title>XiVO WebSocket Example</title>
<script>
var socket = null;
var started = false;
function connect() {
if (socket != null) {
console.log("socket already connected");
return;
}
var host = document.getElementById("host").value;
var token_id = document.getElementById("token").value;
socket = new WebSocket("wss://" + host + ":9502/?token=" + token_id);
socket.onclose = function(event) {
socket = null;
console.log("websocketd closed with code " + event.code + " and reason '" + event.reason + "'");
};
socket.onmessage = function(event) {
if (started) {
console.log("message received: " + event.data);
return;
}
var msg = JSON.parse(event.data);
switch (msg.op) {
case "init":
subscribe("*");
start();
break;
case "start":
started = true;
console.log("waiting for messages");
break;
}
};
started = false;
}
function subscribe(event_name) {
var msg = {
op: "subscribe",
data: {
event_name: event_name
}
};
socket.send(JSON.stringify(msg));
};
function start() {
var msg = {
op: "start"
};
socket.send(JSON.stringify(msg));
}
</script>
</head>
<body>
<p>Open the web console to see what's happening.</p>
<form>
<div>
<label for="host">Host:</label>
<input type="text" id="host" autofocus>
</div>
<div>
<label for="token">Token ID:</label>
<input type="text" id="token" size="35">
</div>
<div>
<button type="button" onclick="connect();">Connect</button>
</div>
</form>
</body>
</html>
|
The page has a form for the user to enter a host and token ID, and has a connect button. When the
button is clicked, the connect
function is called, and the WebSocket connection is created at
line 18 (using the WebSocket API):
socket = new WebSocket("wss://" + host + ":9502/?token=" + token_id);
Then, at line 23, a onmessage
callback is set on the WebSocket object:
socket.onmessage = function(event) {
if (started) {
console.log("message received: " + event.data);
return;
}
var msg = JSON.parse(event.data);
switch (msg.op) {
case "init":
subscribe("endpoint_status_update");
subscribe("user_status_update");
start();
break;
case "start":
started = true;
console.log("waiting for messages");
break;
}
};
After a successful connection to the service, an “init” message will be received by the client. When
the client receives this message, it sends two subscribe commands (e.g.
subscribe("endpoint_status_update")
) and a start command (e.g. start()
). When the client
receives the “start” message, it sets the started
flag. After that, all the other messages it
receives will be logged to the console.
The WebSocket service is provided by xivo-websocketd
, and its behaviour can be configured via
its configuration files located under the /etc/xivo-websocketd
directory. After modifying the configuration files, you need to restart xivo-websocketd
with
systemctl restart xivo-websocketd
.
The service is available on port 9502 on all network interfaces by default. This can be changed in the configuration file.
The canonical URL to reach the service is wss://<host>:9502/
.
The connection is always encrypted. The certificate and private key used by the server can be changed in the configuration file. By default, since the certificate is self-signed, you’ll have to either:
- add a security exception on the client machines that access the service
- use a certificate signed by an untrusted CA and add the CA bundle on the system that access the service
- use a trusted certificate
See the HTTPS certificate section for more information on certificate configuration.
Authentication is done by passing a xivo-auth token ID in the token
query parameter.
Authentication is mandatory.
The token must have the websocketd
ACL.
When the token expires, the server close the connection with the status code 4003. There is currently no way to change the token of an existing connection. A new connection must be made when the token expires.
Clients connected to xivo-websocketd
only receive events that they are authorized to receive.
For example, a client connected with a token obtained from the “xivo_user” xivo-auth
backend
will not receive call events of other users.
When a message is received from the bus by xivo-websocketd
, it extracts the ACL from the
required_acl
key of the event. If the field is missing, no clients will receive the event. If
the value is null, all subscribed clients will receive the event. If the value is a string, then all
subscribed clients which have a matching ACL will receive the event.
No authorization check is done at subscription time. Checks are only done when an event is received by the server. This mean a client can subscribe to an event “foo”, but will never receive any of these events if it does not have the matching ACL.
See the Events section for more information on the required ACL of events which are available by default on XiVO.
The WebSocket connection might be closed by the server using one of following status code:
- 4001: No token ID was provided.
- 4002: Authentication failed. Either the token ID is invalid, expired, or does not have the necessary ACL.
- 4003: Authentication expired. The token has expired or was deleted.
- 4004: Protocol error. The server received a frame that it could not understand. For example, the content was not valid JSON, or was requesting an unknown operation, or a mandatory argument to an operation was missing.
The server also uses the pre-defined WebSocket status codes.
A JSON-based protocol is used over the WebSocket connection to control which events are received by the client.
The format of the messages sent by the client are all of the same format:
{"op": "<operation-name>", "data": <operation-specific-value>}
The “op” key is mandatory, and the value is either “subscribe” or “start”. The “data” key is mandatory for the “subscribe” operation.
The “subscribe” message ask the server to subscribe the client to the given event. When a message with the same name is published on the “xivo” exchange of the bus, the server forwards the message to all the subscribed clients that are authorized to receive it. For this command, the “data” value is a dictionary with an “event_name” key (mandatory). Example:
{"op": "subscribe", "data": {"event_name": "endpoint_status_update"}}
You can subscribe to any event. The special event name *
can be used to match all events.
See the Events section for more information on the events which are available by default on XiVO.
The “start” message ask the server to start sending messages from the bus to the client. Example:
{"op": "start"}
The server won’t forward messages from the bus to the client until it receives the “start” message from the client.
If the client send a message that the server doesn’t understand, the server closes the connection.
The format of the messages sent by the server are all of the same format (until the server receives a “start” command):
{"op": "<operation-name>", "code": <status-code>, "msg": "<error message>"}
The 3 keys are always present. The value of the “op” key can be one of “init”, “subscribe” or “start”. The value of the “code” key is an integer representing the status of the operation, 0 meaning there was no error, other values meaning there was an error. The “msg” is an empty string unless “code” is non-zero, in which case it’s a human-readable message of the error.
The “init” message is only sent after the connection is successfully established between the client and the server. It’s code is always zero; if the connection could not be established, the connection is simply closed. Example:
{"op": "init", "code": 0, "msg": ""}
The “subscribe” message is sent as a response to a client “subscribe” message. The code is always zero. Example:
{"op": "subscribe", "code": 0, "msg": ""}
The “start” message is sent as a response to a client “start” message. The code is always zero. Example:
{"op": "start", "code": 0, "msg": ""}
After receiving the “start” message, the server switch into the “bus/started” mode, where all messages that the server will ever sent will be the body of the messages it received on the bus on behalf of the client.
Note that a client can subscribe to more events after sending its “start” message, but it won’t receive any response from the server, e.g. the server won’t send a corresponding “subscribe” message. Said differently, once the client has sent a “start” message, every message the client will ever receive are messages coming from the bus.
Contributors¶
General information:
Contributing to the Documentation¶
XiVO documentation is generated with Sphinx. The source code is available on GitHub at https://github.com/xivo-pbx/xivo-doc
Provided you already have Python installed on your system. You need first to install Sphinx : easy_install -U Sphinx
[1].
Quick Reference
- http://docutils.sourceforge.net/docs/user/rst/cheatsheet.txt
- http://docutils.sourceforge.net/docs/user/rst/quickref.html
- http://openalea.gforge.inria.fr/doc/openalea/doc/_build/html/source/sphinx/rest_syntax.html
[1] | easy_install can be found in the debian package python-setuptools : sudo apt-get install python-setuptools |
Here’s the guideline/conventions to follow for the XiVO documentation.
The documentation must be written in english, and only in english.
The top section of each file must be capitalized using the following rule: capitalization of all words, except for articles, prepositions, conjunctions, and forms of to be.
Correct:
The Vitamins are in My Fresh California Raisins
Incorrect:
The Vitamins Are In My Fresh California Raisins
Use the following punctuation characters:
*
with overline, for “file title”=
, for sections-
, for subsections^
, for subsubsections
Punctuation characters should be exactly as long as the section text.
Correct:
Section1
========
Incorrect:
Section2
==========
There should be 2 empty lines between sections, except when an empty section is followed by another section.
Correct:
Section1
========
Foo.
Section2
========
Bar.
Correct:
Section1
========
Foo.
.. _target:
Section2
========
Bar.
Correct:
Section1
========
Subsection1
-----------
Foo.
Incorrect:
Section1
========
Foo.
Section2
========
Bar.
Use ::
on the same line as the line containing text when possible.
The literal blocks must be indented with three spaces.
Correct:
Bla bla bla::
apt-get update
Incorrect:
Bla bla bla:
::
apt-get update
Use the following roles when applicable:
:file:
for file, i.e.:The :file:`/dev/null` file.
:menuselection:
for the web interface menu:The :menuselection:`Configuration --> Management --> Certificates` page.
:guilabel:
for designating a specific GUI element:The :guilabel:`Action` column.
- There must be no warning nor error messages when building the documentation with
make html
. - There should be one and only one newline character at the end of each file
- There should be no trailing whitespace at the end of lines
- Paragraphs must be wrapped and lines should be at most 100 characters long
Debugging Asterisk¶
To debug asterisk crashes or freezes, you need the following debug packages on your XiVO:
General rule | XiVO < 14.18 | XiVO >= 14.18 |
---|---|---|
Example version | 14.12 | 14.18 |
Commands | apt-get install xivo-fai-14.12
apt-get update
apt-get install gdb
apt-get install -t xivo-14.12 asterisk-dbg xivo-libsccp-dbg
|
xivo-dist xivo-14.18
apt-get update
apt-get install gdb
apt-get install -t xivo-14.18 asterisk-dbg xivo-libsccp-dbg
|
Find out the time of the incident from the people most likely to know
Determine if there was a segfault
The command
grep segfault /var/log/syslog
should return a line such as the following:Oct 16 16:12:43 xivo-1 kernel: [10295061.047120] asterisk[1255]: segfault at e ip b751aa6b sp b5ef14d4 error 4 in libc-2.11.3.so[b74ad000+140000]
Note the exact time of the incident from the segfault line.
Follow the Debugging Asterisk Crash procedure.
If you observe some of the following common symptoms, follow the Debugging Asterisk Freeze procedure.
- The output of command
service asterisk status
says Asterisk PBX is running. - No more calls are distributed and phones go to
No Service
. - Command
core show channels
returns only headers (no data) right before returning
- The output of command
Fetch Asterisk logs for the day of the crash (make sure file was not already logrotated):
cp -a /var/log/asterisk/full /var/local/`date +"%Y%m%d"`-`hostname`-asterisk-full.log
Fetch xivo-ctid logs for the day of the crash (make sure file was not already logrotated):
cp -a /var/log/xivo-ctid.log /var/local/`date +"%Y%m%d"`-`hostname`-xivo-ctid.log
Open a new issue on the bugtracker with following information
- Tracker: Bug
- Status: New
- Category: Asterisk
- In versions: The version of your XiVO installation where the crash/freeze happened
- Subject :
Asterisk Crash
orAsterisk Freeze
- Description : Add as much context as possible, if possible, a scenario that lead to the issue, the date and time of issue, where we can fetch logs and backtrace
- Attach logs and backtrace (if available) to the ticket (issue must be saved, then edited and files attached to a comment).
When asterisk crashes, it usually leaves a core file in /var/spool/asterisk/
.
You can create a backtrace from a core file named core_file
with:
gdb -batch -ex "bt full" -ex "thread apply all bt" asterisk core_file > bt-threads.txt
You can create a backtrace of a running asterisk process with:
gdb -batch -ex "thread apply all bt" asterisk $(pidof asterisk) > bt-threads.txt
If your version of asterisk has been compiled with the DEBUG_THREADS flag, you can get more information about locks with:
asterisk -rx "core show locks" > core-show-locks.txt
Note
Debugging freeze without this information is usually a lot more difficult.
Optionally, other information that can be interesting:
- the output of
asterisk -rx 'core show channels'
- the verbose log of asterisk just before the freeze
It’s relatively straightforward to recompile the asterisk version of your XiVO with the DEBUG_THREADS and DONT_OPTIMIZE flag, which make debugging an asterisk problem easier.
The steps are:
Uncomment the
deb-src
line for the XiVO sources:sed -i 's/^# *deb-src/deb-src/' /etc/apt/sources.list.d/xivo*
Fetch the asterisk source package:
mkdir -p ~/ast-rebuild cd ~/ast-rebuild apt-get update apt-get install -y build-essential apt-get source asterisk
Install the build dependencies:
apt-get build-dep -y asterisk
Enable the DEBUG_THREADS and DONT_OPTIMIZE flag:
cd <asterisk-source-folder> vim debian/rules
Update the changelog by appending
+debug1
in the package version:vim debian/changelog
Rebuild the asterisk binary packages:
dpkg-buildpackage -us -uc
This will create a couple of .deb files in the parent directory, which you can install via dpkg.
It is sometimes useful to produce a “vanilla” version of Asterisk, i.e. a version of Asterisk that has none of the XiVO patches applied, to make sure that the problem is present in the original upstream code. This is also sometimes necessary before opening a ticket on the Asterisk issue tracker.
The procedure is similar to the one described above. Before calling dpkg-buildpackage
, you just need to:
Make sure
quilt
is installed:apt-get install -y quilt
Unapply all the currently applied patches:
quilt pop -a
Remove all the lines in the
debian/patches/series
file:truncate -s0 debian/patches/series
When installing a vanilla version of Asterisk on a XiVO 16.08 or earlier, you’ll need to stop monit, otherwise it will restart asterisk every few minutes.
Install valgrind:
apt-get install valgrind
Recompile asterisk with the DONT_OPTIMIZE flag.
Edit
/etc/asterisk/modules.conf
so that asterisk doesn’t load unnecessary modules. This step is optional. It makes asterisk start (noticeably) faster and often makes the output of valgrind easier to analyze, since there’s less noise.Edit
/etc/asterisk/asterisk.conf
and comment thehighpriority
option. This step is optional.Stop monit and asterisk:
monit quit service asterisk stop
Stop all unneeded XiVO services. For example, it can be useful to stop xivo-ctid, so that it won’t interact with asterisk via the AMI.
Copy the valgrind.supp file into /tmp. The valgrind.supp file is located in the contrib directory of the asterisk source code.
Execute valgrind in the /tmp directory:
cd /tmp valgrind --leak-check=full --log-file=valgrind.txt --suppressions=valgrind.supp --vgdb=no asterisk -G asterisk -U asterisk -fnc
Note that when you terminate asterisk with Control-C, asterisk does not unload the modules before exiting. What this means is that you might have lots of “possibly lost” memory errors due to that. If you already know which modules is responsible for the memory leak/bug, you should explicitly unload it before terminating asterisk.
Running asterisk under valgrind takes a lots of extra memory, so make sure you have enough RAM.
Debugging Daemons¶
To activate debug mode, add debug: true
in the daemon configuration file). The output will be available in the daemon’s log file.
It is also possible to run the XiVO daemon, in command line. This will allow to run in foreground and debug mode. To see how to use it, type:
xivo-{name} -h
Note that it’s usually a good idea to stop monit before running a daemon in foreground:
systemctl stop monit.service
twistd -no -u xivo-confgend -g xivo-confgend --python=/usr/bin/xivo-confgend --logger xivo_confgen.bin.daemon.twistd_logs
No debug mode in confgend.
twistd -no -u xivo-provd -g xivo-provd -r epoll --logger provd.main.twistd_logs xivo-provd -s -v
- -s for logging to stderr
- -v for verbose
sudo -u consul /usr/bin/consul agent -config-dir /etc/consul/xivo -pid-file /var/run/consul/consul.pid
There is no log file, but you can consult the output of consul with:
consul monitor
2015/08/03 09:48:25 [INFO] consul: cluster leadership acquired
2015/08/03 09:48:25 [INFO] consul: New leader elected: this-xivo
2015/08/03 09:48:26 [INFO] raft: Disabling EnableSingleNode (bootstrap)
2015/08/03 11:04:08 [INFO] agent.rpc: Accepted client: 127.0.0.1:41545
Generate your own prompts¶
If you want your XiVO to speak in your language that is not supported by XiVO, and you don’t want to record the whole package of sounds in a studio, you may generate them yourself with some text-to-speech services.
The following procedure will generate prompts for pt_BR
(portuguese from Brazil) based on the
Google TTS service.
Note
There are two sets of prompts: the Asterisk prompts and the XiVO prompts. This procedure only covers the XiVO prompts, but it may be adapted for Asterisk prompts.
Create an account on Transifex and join the team of translation of XiVO.
Translate the prompts in the xivo-prompts resource.
Go to https://www.transifex.com/proformatique/xivo/xivo-prompt/pt_BR/download/for_use/ and download the file on your XiVO. You should have a file named like
for_use_xivo_xivo-prompt_pt_BR.ini
.On your XiVO, download the tool to automate the use of Google TTS:
wget https://github.com/zaf/asterisk-googletts/raw/master/cli/googletts-cli.pl chmod +x googletts-cli.pl
Then run the tool, and generate the sound files (set
LANGUAGE
andCOUNTRY
to your own language):LANGUAGE=pt COUNTRY=BR mkdir -p wav/{digits,letters} cat for_use_xivo_xivo-prompt_${LANGUAGE}_${COUNTRY}.ini | while IFS='=' read file text ; do echo $file ./googletts-cli.pl -t "$text" -l ${LANGUAGE}-${COUNTRY} -s 1.4 -r 8000 -o wav/$file.wav done
Install the prompts on your system:
mv wav /usr/share/asterisk/sounds/${LANGUAGE}_${COUNTRY}
Make your language available in the web interface:
sed -i "s/'nl_NL'/\0, '${LANGUAGE}_${COUNTRY}'/" /usr/share/xivo-web-interface/lib/i18n.inc systemctl restart spawn-fcgi
Note that this last modification may be erased after running xivo-upgrade
.
And that’s it, you can configure a user to use your new language and he will hear the prompts in your language. You may also want to use the xivo-confd HTTP API to mass-update your users.
XiVO Guidelines¶
Our current goal is to use only two means of communication between XiVO processes:
- a REST API over HTTP for synchronous commands
- a software bus (RabbitMQ) for asynchronous events
Each component should have its own REST API and its own events and can communicate with every other component from across a network only via those means.
The current xivo-dao Git repository contains the basis of the future services Python API. The API is split between different resources available in XiVO, such as users, groups, schedules... For each resource, there are different modules :
- service: the public module, providing possible actions. It contains only business logic and no technical logic. There must be no file name, no SQL queries and no URLs in this module.
- dao: the private Data Access Object. It knows where to get data and how to update it, such as SQL queries, file names, URLs, but has no business logic.
- model: the public class used to represent the resource. It must be self-contained and have almost no methods, except for computed fields based on other fields in the same object.
- notifier: private, it knows to whom and in which format events must be sent.
- validator: private, it checks input parameters from the service module.
The goal is to make XiVO as elastic as possible, i.e. the XiVO services need to be able to run on separate machines and still talk to each other.
To be in accordance with our goal, a XiVO daemon must (if applicable):
- Offer a REST API (with encryption, authentication and accepting cross-site requests)
- Be able to read and send events on a software bus
- Be able to run inside a container, such as Docker, and be separated from the XiVO server
- Offer a configuration file in YAML format.
- Access the XiVO database through the
xivo-dao
library - Have a configurable level of logging
- Have its own log file
- Be extendable through the use of plugins
- Not run with system privileges
- Be installable from source
- Service discovery with consul
Currently, none of the XiVO daemons meet these expectations; it is a work in progress.
Network¶
Network Flow table (IN) :
Daemon Name | Service | Protocol | Port | Listen | Authentication | Enabled |
---|---|---|---|---|---|---|
- | ICMP | ICMP | - | 0.0.0.0 | no | yes |
postfix | SMTP | TCP | 25 | 0.0.0.0 | yes | yes |
isc-dhcpd | DHCP | UDP | 67 | 0.0.0.0 | no | no |
isc-dhcpd | DHCP | UDP | 68 | 0.0.0.0 | no | no |
xivo-provd | TFTP | UDP | 69 | 0.0.0.0 | no | yes |
ntpd | NTP | UDP | 123 | 0.0.0.0 | yes | yes |
monit | HTTP | TCP | 2812 | 127.0.0.1 | no | yes |
asterisk | SIP | UDP | 5060 | 0.0.0.0 | yes | yes |
asterisk | IAX | UDP | 4569 | 0.0.0.0 | yes | yes |
asterisk | SCCP | TCP | 2000 | 0.0.0.0 | yes | yes |
asterisk | AMI | TCP | 5038 | 127.0.0.1 | yes | yes |
asterisk | HTTP | TCP | 5039 | 127.0.0.1 | yes | yes |
asterisk | HTTPS | TCP | 5040 | 127.0.0.1 | yes | yes |
sshd | SSH | TCP | 22 | 0.0.0.0 | yes | yes |
nginx | HTTP | TCP | 80 | 0.0.0.0 | yes | yes |
nginx | HTTPS | TCP | 443 | 0.0.0.0 | yes | yes |
munin | HTTP | TCP | 4949 | 127.0.0.1 | no | yes |
xivo-ctid | XiVO-CTI/S | TCP | 5003 | 0.0.0.0 | yes | yes |
postgresql | SQL | TCP | 5432 | 127.0.0.1 | yes | yes |
rabbitMQ | AMQP | TCP | 5672 | 0.0.0.0 | yes | yes |
consul | Consul RPC | TCP | 8300 | 127.0.0.1 | yes | yes |
consul | Consul Serf LAN | TCP/UDP | 8301 | 127.0.0.1 | yes | yes |
consul | Consul Serf WAN | TCP/UDP | 8302 | 127.0.0.1 | yes | yes |
consul | Consul HTTPS | TCP | 8500 | 127.0.0.1 | both | yes |
xivo-provd | HTTP | TCP | 8666 | 127.0.0.1 | no | yes |
xivo-provd | HTTP | TCP | 8667 | 0.0.0.0 | no | yes |
xivo-confgend | HTTP | TCP | 8669 | 127.0.0.1 | no | yes |
xivo-sysconfd | HTTP | TCP | 8668 | 127.0.0.1 | no | yes |
xivo-confd | HTTPS | TCP | 9486 | 0.0.0.0 | yes | yes |
xivo-confd | HTTP | TCP | 9487 | 127.0.0.1 | no | yes |
xivo-dird | HTTPS | TCP | 9489 | 0.0.0.0 | yes | yes |
xivo-amid | HTTPS | TCP | 9491 | 0.0.0.0 | yes | yes |
xivo-agentd | HTTPS | TCP | 9493 | 0.0.0.0 | yes | yes |
xivo-ctid | HTTP | TCP | 9495 | 127.0.0.1 | no | yes |
xivo-auth | HTTPS | TCP | 9497 | 0.0.0.0 | both | yes |
xivo-dird-phoned | HTTP | TCP | 9498 | 0.0.0.0 | IP filtering | yes |
xivo-dird-phoned | HTTPS | TCP | 9499 | 0.0.0.0 | IP filtering | yes |
xivo-ctid-ng | HTTPS | TCP | 9500 | 0.0.0.0 | yes | yes |
xivo-websocketd | WSS | TCP | 9502 | 0.0.0.0 | yes | yes |
Debian packaging for XiVO¶
Download the package:
apt-get download name-of-package/jessie-backports
Copy the .deb on to the mirror:
scp name-of-package.deb mirror.xivo.solutions:/tmp
Add package to distribution on mirror:
ssh mirror.xivo.solutions cd /data/reprepro/xivo reprepro includedeb xivo-dev /tmp/name-of-package.deb
Profiling Python Programs¶
Here’s an example on how to profile xivo-ctid for CPU/time usage:
Stop the monit daemon:
service monit stop
Stop the process you want to profile, i.e. xivo-ctid:
service xivo-ctid stop
Start the service in foreground mode running with the profiler:
python -m cProfile -o test.profile /usr/bin/xivo-ctid -f
This will create a file named
test.profile
when the process terminates.To profile xivo-confgend, you must use this command instead of the one above:
twistd -p test.profile --profiler=cprofile --savestats -no --python=/usr/bin/xivo-confgend
Note that profiling multi-threaded program (xivo-agid, xivo-confd) doesn’t work reliably.
The Debugging Daemons section documents how to launch the various XiVO services in foreground/debug mode.
Examine the result of the profiling:
$ python -m pstats test.profile Welcome to the profile statistics browser. % sort time % stats 15 ... % sort cumulative % stats 15
Here’s an example on how to measure the code coverage of xivo-ctid.
This can be useful when you suspect a piece of code to be unused and you want to have additional information about it.
Install the following packages:
apt-get install python-pip build-essential python-dev
Install coverage via pip:
pip install coverage
Run the program in foreground mode with
coverage run
:service monit stop service xivo-ctid stop coverage erase coverage run /usr/bin/xivo-ctid -f
The Debugging Daemons section documents how to launch the various XiVO service in foreground/debug mode.
After the process terminates, use
coverage html
to generate an HTML coverage report:coverage html --include='*xivo_cti*'
This will generate an
htlmcov
directory in the current directory.Browse the coverage report.
Either copy the directory onto your computer and open it with a web browser, or start a web server on the XiVO:
cd htmlcov python -m SimpleHTTPServer
Then open the page from your computer (i.e. not on the xivo):
firefox http://<xivo-hostname>:8000
Style Guide¶
Python files start with a UTF8 encoding comment and the GPLv3 license. A blank line should separate the license from the imports
Example:
# -*- coding: utf-8 -*-
# Copyright 2016 Avencall
# SPDX-License-Identifier: GPL-3.0+
import argparse
- Lines should not go further than 80 to 100 characters.
- In python, indentation blocks use 4 spaces
- In PHP, indentation blocks use tabs
- Imports should be ordered alphabetically
- Separate module imports and
from
imports with a blank line
Example:
import argparse
import datetime
import os
import re
import shutil
import tempfile
from StringIO import StringIO
from urllib import urlencode
When possible, use pep8 to validate your code. Generally, the following errors are ignored :
- E501 (max 80 chars per line)
Example:
pep8 --ignore=E501 xivo_cti
When possible, avoid using backslashes to separate lines.
Bad Example:
user = session.query(User).filter(User.firstname == firstname)\
.filter(User.lastname == lastname)\
.filter(User.number == number)\
.all()
Good Example:
user = (session.query(User).filter(User.firstname == firstname)
.filter(User.lastname == lastname)
.filter(User.number == number)
.all())
Avoid using the + operator for concatenating strings. Use string interpolation instead.
Bad Example:
phone_interface = "SIP" + "/" + username + "-" + password
Good Example:
phone_interface = "SIP/%s-%s" % (username, password)
Redundant comments should be avoided. Instead, effort should be put on making the code clearer.
Bad Example:
#Add the meeting to the calendar only if it was created on a week day
#(monday to friday)
if meeting.day > 0 and meeting.day < 7:
calendar.add(meeting)
Good Example:
def created_on_week_day(meeting):
return meeting.day > 0 and meeting.day < 7
if created_on_week_day(meeting):
calendar.add(meeting)
Avoid using parenthesis around if statements, unless the statement expands on multiple lines or you need to nest your conditions.
Bad Examples:
if(x == 3):
print "condition is true"
if(x == 3 and y == 4):
print "condition is true"
Good Examples:
if x == 3:
print "condition is true"
if x == 3 and y == 4:
print "condition is true"
if (extremely_long_variable == 3
and another_long_variable == 4
and yet_another_variable == 5):
print "condition is true"
if (2 + 3 + 4) - (1 + 1 + 1) == 6:
print "condition is true"
Consider refactoring your statement into a function if it becomes too long, or the meaning isn’t clear.
Bad Example:
if price * tax - bonus / reduction + fee < money:
product.pay(money)
Good Example:
def calculate_price(price, tax, bonus, reduction, fee):
return price * tax - bonus / reduction + fee
final_price = calculate_price(price, tax, bonus, reduction, fee)
if final_price < money:
product.pay(money)
- Class names are in
CamelCase
- File names are in
lower_underscore_case
Conventions for functions prefixed by find:
- Return None when nothing is found
- Return an object when a single entity is found
- Return the first element when multiple entities are found
Example:
def find_by_username(username):
users = [user1, user2, user3]
user_search = [user for user in users if user.username == username]
if len(user_search) == 0:
return None
return user_search[0]
Conventions for functions prefixed by get:
- Raise an Exception when nothing is found
- Return an object when a single entity is found
- Return the first element when multiple entities are found
Example:
def get_user(userid):
users = [user1, user2, user3]
user_search = [user for user in users if user.userid == userid]
if len(user_search) == 0:
raise UserNotFoundError(userid)
return user_search[0]
Conventions for functions prefixed by find_all:
- Return an empty list when nothing is found
- Return a list of objects when multiple entites are found
Example:
def find_all_users_by_username(username):
users = [user1, user2, user3]
user_search = [user for user in users if user.username == username]
return user_search
Magic numbers should be avoided. Arbitrary values should be assigned to variables with a clear name
Bad example:
class TestRanking(unittest.TestCase):
def test_ranking(self):
rank = Rank(1, 2, 3)
self.assertEquals(rank.position, 1)
self.assertEquals(rank.grade, 2)
self.assertEquals(rank.session, 3)
Good example:
class TestRanking(unittest.TestCase):
def test_ranking(self):
position = 1
grade = 2
session = 3
rank = Rank(position, grade, session)
self.assertEquals(rank.position, position)
self.assertEquals(rank.grade, grade)
self.assertEquals(rank.session, session)
Tests for a package are placed in their own folder named “tests” inside the package.
Example:
package1/
__init__.py
mod1.py
tests/
__init__.py
test_mod1.py
package2/
__init__.py
mod9.py
tests/
__init__.py
test_mod9.py
Unit tests should be short, clear and concise in order to make the test easy to understand. A unit test is separated into 3 sections :
- Preconditions / Preparations
- Thing to test
- Assertions
Sections are separated by a blank line. Sections that become too big should be split into smaller functions.
Example:
class UserTestCase(unittest.TestCase):
def test_fullname(self):
user = User(firstname='Bob', lastname='Marley')
expected = 'Bob Marley'
fullname = user.fullname()
self.assertEquals(expected, fullname)
def _prepare_expected_user(self, firstname, lastname, number):
user = User()
user.firstname = firstname
user.lastname = lastname
user.number = number
return user
def _assert_users_are_equal(expected_user, actual_user):
self.assertEquals(expected_user.firstname, actual_user.firstname)
self.assertEquals(expected_user.lastname, actual_user.lastname)
self.assertEquals(expected_user.number, actual_user.number)
def test_create_user(self):
expected = self._prepare_expected_user('Bob', 'Marley', '4185551234')
user = create_user('Bob', 'Marley', '4185551234')
self._assert_users_are_equal(expected, user)
Exceptions should not be used for flow control. Raise exceptions only for edge cases, or when something that isn’t usually expected happens.
Bad Example:
def is_user_available(user):
if user.available():
return True
else:
raise Exception("User isn't available")
try:
is_user_available(user)
except Exception:
disable_user(user)
Good Example:
def is_user_available(user):
if user.available():
return True
else:
return False
if not is_user_available(user):
disable_user(user)
Avoid throwing Exception
. Use one of Python’s built-in Exceptions, or create
your own custom Exception. A list of exceptions is available on the Python documentation website.
Bad Example:
def get_user(userid):
user = session.query(User).get(userid)
if not user:
raise Exception("User not found")
Good Example:
class UserNotFoundError(LookupError):
def __init__(self, userid):
message = "user with id %s not found" % userid
LookupError.__init__(self, message)
def get_user(userid):
user = session.query(User).get(userid)
if not user:
raise UserNotFoundError(userid)
Never use except:
without specifying any exception type. The reason is that it will also catch important exceptions, such as KeyboardInterrupt
and OutOfMemory
exceptions, making your program unstoppable or continuously failing, instead of stopping when wanted.
Bad Example:
try:
get_user(user_id)
except:
logger.exception("There was an error")
Good Example:
try:
get_user(user_id)
except UserNotFoundError as e:
logger.error(e.message)
raise
Translating XiVO¶
French and English are maintained by Avencall. Other languages are provided by the community.
Avencall is in contact with several studios for different languages and prompts. The information for those languages are :
- French : Super Sonic productions (supersonicprod@wanadoo.fr)
- English : Asterisk voice (allison@theasteriskvoice.com)
- German : ATS studio
- Italian : ATS studio
Prompts transcripts are listed in Transifex (*-prompts). You may translate them there.
The prompts used in XiVO are stored in xivo-sounds git repository. You may also want to generate your own sound files.
All translations are in Transifex (xivo-client). The source language is English. Translations are synchronised with the code before every release.
Translations are currently available in French and English. There are no plans to translate the Web interface in other languages.
XiVO Package File Structure¶
Let’s assume we want to organise the files for xivo-confd.
- Git repo name:
xivo-confd
- Binary file name:
xivo-confd
- Python package name:
xivo_confd
xivo-confd
|-- bin
| `-- xivo-confd
|-- contribs
| `-- docker
| |-- ...
| `-- prod
| `-- ...
|-- debian
| `-- ...
|-- Dockerfile
|-- docs
| `-- ...
|-- etc
| `-- ...
|-- integration-tests
| `-- ...
|-- LICENSE
|-- README.md
|-- requirements.txt
|-- setup.cfg
|-- setup.py
|-- test-requirements.txt
|-- .travis.yml
`-- xivo_confd
`-- ...
etc/
- Contains default configuration files.
docs/
- Contains technical documentation for this package: API doc, architecture doc, diagrams, ... Should be in RST format using Sphinx.
bin/
- Contains the binaries. Not applicable for pure libraries.
integration-tests/
- Contains the tests bigger than unit-tests. Tests should be runnable simply, e.g.
nosetests integration-tests
. README.md
- Read me in markdown (Github flavor).
LICENSE
- License (GPLv3)
.travis.yml
- Travis CI configuration file
Standard files:
- setup.py
- setup.cfg
- requirements.txt
- test-requirements.txt
- xivo_confd/ (the main sources)
debian/
- Contains the Debian packaging files (
control
,rules
, ...)
Dockerfile
- Used to build a docker image for a working production version
contribs/docker/prod/
- Contains the files necessary for running xivo-confd inside a production Docker image
contribs/docker/other/
- Contains the Dockerfile and other files to run xivo-confd inside Docker with specific configuration
- PID file:
/var/run/xivo-confd/xivo-confd.pid
- WSGI socket file:
/var/run/xivo-confd/xivo-confd.sock
- Config file:
/etc/xivo-confd/config.yml
- Log file:
/var/log/xivo-confd.log
- Static data files:
/usr/share/xivo-confd
- Storage data files:
/var/lib/xivo-confd
Component specific information:
CTI Server¶
This section describes the informations and tools for CTI Server.
Here’s how to run the various CTI client-server development/debugging tools. These tools can be found on GitHub, in the XiVO project.
You can get the scripts by using Git:
$ git clone https://github.com/xivo-pbx/xivo-tools.git
Both the ctispy, ctisave and ctistat tools work in a similar way. They both are proxies that need to be inserted between the CTI client and the CTI server message flow.
To do this, you first start the given tool on your development machine, giving it the CTI server hostname as the first argument. You then configure your CTI client to connect to the tool on port 50030 (notice the trailing 0). The tool should then accept the connection from the client, and once this is done, will make a connection to the server, thereby being able to process all the information sent between the client and the server.
In the following examples, we suppose that the CTI server is located on the host named xivo-new.
ctispy
can be used to see the message flow between the client and the server in
“real-time”.
The simplest invocation is:
$ cti-proxy/ctispy xivo-new
You can pretty-print the messages if you want by using the --pretty-print
option:
$ cti-proxy/ctispy xivo-new --pretty-print
By default, each message is displayed separately even though more than one
message can be in a single TCP packet. You can also use the --raw
option if you
want to see the raw traffic between the client and the server:
$ cti-proxy/ctispy xivo-new --raw
Note that when using the --raw
option, some other option doesn’t work because
the messages are not decoded/analyzed.
If you want to remove some fields from the messages, you can use the --strip
option:
$ cti-proxy/ctispy xivo-new --strip timenow --strip commandid --strip replyid
If you want to see only messages matching a certain key and value, use the
--include
option:
$ cti-proxy/ctispy xivo-new --include class=getlist
Finally, you can ignore all the messages from the client or the server by using
the --no-client
or --no-server
option respectively.
By default, ctispy will exit after the connection with the client is closed. You
can bypass this behavior with the --loop
option, that will make the CTI proxy
continue, whether the client is connected or not.
ctisave
save the messages from the client and the server in two separate
files. This is useful to do more careful post-analysis.
The simplest invocation is:
$ cti-proxy/ctisave xivo-new /tmp/cti-client /tmp/cti-server
To do comparison, it’s often useful to strip some fields:
$ cti-proxy/ctisave xivo-new /tmp/cti-client /tmp/cti-server --strip timenow
--strip commandid --strip replyid
One useful thing to do with files generated from different ctisave invocation is to compare them with a tool like vimdiff, for example:

ctistat
display various statistic about a CTI “session” when it ends.
The simplest invocation is:
$ cti-proxy/ctistat xivo-new
The versions below indicate the xivo version followed by the protocol version.
Warning
The CTI server protocol is subject to change without any prior warning. If you are using this protocol in your own tools please be sure to check that the protocol did not change before upgrading XiVO
- the user_id field has been added back to the User status update
- the Register user status update now uses the user_uuid instead of the user_id
- the User status update now uses the user_uuid instead of the user_id
- the Chitchat command to and from fields are now a list of two strings, xivo_uuid and user_uuid.
- the lastconnswins field has been removed from the Login capas command
- the loginkind field has been removed from the Login capas command
- the ipbxcommands and regcommands capakinds have been removed from Login capas command
- the Login password command has been modified. The hashedpassword has been replaced by the password field which is now sent verbatim.
- the Chitchat command to field is now a list of two elements, xivo_uuid and user_id.
- the
getlist
command has been removed for the channels listname. - many fields have been removed from the
getlist
command.- users list
- enableclient
- profileclient
- phones
- context
- protocol
- simultcalls
- channels
- voicemails
- fullname
- old
- waiting
- agents
- phonenumber
- users list
- some ipbxcommands have been removed:
- mailboxcount
- atxfer
- transfer
- hangup
- originate
- add the Attended transfer to voicemail command
- add the Blind transfer to voicemail command
- the Send fax command now include the size and data field.
- the filetransfer command has been removed.
- the Get relations command was added.
- the Relations message was added.
- the
people_purge_personal_contacts
message was added. - the
people_personal_contacts_purged
message was added. - the
people_personal_contact_raw
message was added. - the
people_personal_contact_raw_result
message was added. - the
people_edit_personal_contact
message was added. - the
people_personal_contact_raw_update
message was added. - the
people_import_personal_contacts_csv
message was added. - the
people_import_personal_contacts_csv_result
message was added. - the
people_export_personal_contacts_csv
message was added. - the
people_export_personal_contacts_csv_result
message was added. - for messages
people_personal_contact_deleted
andpeople_favorite_update
there are no longerdata
sub-key.
- for
channel status update
message:- the value of
commstatus
have been changed fromlinked-caller
andlinked-called
tolinked
. - the key
direction
have been removed. - the key
talkingto_kind
have been removed.
- the value of
- the
people_personal_contacts
message was added. - the
people_personal_contacts_result
message was added. - the
people_create_personal_contact
message was added. - the
people_personal_contact_created
message was added. - the
people_delete_personal_contact
message was added. - the
people_personal_contact_deleted
message was added.
people_search_result
has a new key inrelations
:source_entry_id
- the
people_favorites
message was added. - the
people_favorites_result
message was added. - the
people_set_favorite
message was added. - the
people_favorite_update
message was added.
- the
fax_progress
message was added.
- for messages of class
history
the client cannot request by mode anymore. The server returns all calls and the mode is now metadata for each call.
- for messages of class
ipbxcommand
, the commandrecord
andsipnotify
have been removed. - the
logfromclient
message has been removed
- for messages of class
faxsend
, the stepsfile_decoded
andfile_converted
have been removed.
- the
dial_success
message was added
- the
unhold_switchboard
command was renamedresume_switchboard
.
- the
actionfiche
message was renamedcall_form_result
.
- for messages of class
login_capas
from server to client: the keypresence
has been removed.
- for messages of class
getlist
, listagents
and functionupdatestatus
: the keyavailability
in thestatus
object/dictionary has changed values:- deleted values:
on_call_non_acd_incoming
andon_call_non_acd_outgoing
- added values:
*
on_call_non_acd_incoming_internal
*on_call_non_acd_incoming_external
*on_call_non_acd_outgoing_internal
*on_call_non_acd_outgoing_external
- deleted values:
- for messages of class
getlist
, listagents
and functionupdatestatus
: the keyavailability
in thestatus
object/dictionary has changed values:- deleted value:
on_call_non_acd
- added values:
on_call_non_acd_incoming
andon_call_non_acd_outgoing
- deleted value:
- for messages of class
getlist
and functionupdateconfig
, theconfig
object/dictionary does not have arules_order
key anymore.
Objects have the format: “<type>:<xivoid>/<typeid>”
- <type> can take any of the following values: user, agent, queue, phone, group, meetme, ...
- <xivoid> indicates on which server the object is defined
- <typeid> is the object id, type dependant
- e.g.
- user:xivo-test/5 I’m looking for the user that has the ID 5 on the xivo-test server.
Here is a non exaustive list of types:
- exten
- user
- vm_consult
- voicemail
Client -> Server
{"agentphonenumber": "1000", "class": "ipbxcommand", "command": "agentlogin", "commandid": 733366597}
agentphonenumber is the physical phone set where the agent is going to log on.
Server > Client
- Login successfull :
{"function": "updateconfig",
"listname": "queuemembers",
"tipbxid": "xivo",
"timenow": 1362664323.94,
"tid": "Agent/2002,blue",
"config": {"paused": "0",
"penalty": "0",
"membership": "static",
"status": "1",
"lastcall": "",
"interface": "Agent/2002",
"queue_name": "blue",
"callstaken": "0"},
"class": "getlist"}
{"function": "updatestatus",
"listname": "agents",
"tipbxid": "xivo",
"timenow": 1362664323.94,
"status": {"availability_since": 1362664323.94,
"queues": [],
"on_call": false,
"availability": "available",
"channel": null},
"tid": 7,
"class": "getlist"}
- The phone number is already used by an other agent :
{"class": "ipbxcommand", "error_string": "agent_login_exten_in_use", "timenow": 1362664158.14}
Client -> Server
{"class": "ipbxcommand", "command": "agentlogout", "commandid": 552759274}
On all queues
Client -> Server
{"class": "ipbxcommand", "command": "queuepause", "commandid": 859140432, "member": "agent:xivo/1", "queue": "queue:xivo/all"}
On all queues
Client -> Server
{"class": "ipbxcommand", "command": "queueunpause", "commandid": 822604987, "member": "agent:xivo/1", "queue": "queue:xivo/all"}
Client -> Server
{"class": "ipbxcommand", "command": "queueadd", "commandid": 542766213, "member": "agent:xivo/3", "queue": "queue:xivo/2"}
Client -> Server
{"class": "ipbxcommand", "command": "queueremove", "commandid": 742480296, "member": "agent:xivo/3", "queue": "queue:xivo/2"}
Client -> Server
{"class": "ipbxcommand", "command": "listen", "commandid": 1423579492, "destination": "xivo/1", "subcommand": "start"}
The following messages are used to retrieve XiVO configuration.
- class : getlist
- function : listid
- commandid
- tipbxid
- listname : Name of the list to be retreived : users, phones, agents, queues, voicemails, queuemembers
{
"class": "getlist",
"commandid": 489035169,
"function": "listid",
"tipbxid": "xivo",
"listname": "........."
}
Return a list of configured user id’s
Client -> Server
{"class": "getlist", "commandid": 489035169, "function": "listid", "listname": "users", "tipbxid": "xivo"}
Server -> Client
{
"class": "getlist",
"function": "listid", "listname": "users",
"list": ["11", "12", "14", "17", "1", "3", "2", "4", "9"],
"tipbxid": "xivo","timenow": 1362735061.17
}
Return a user configuration
- tid is the userid returned by Users configuration message
Client -> Server
{
"class": "getlist",
"function": "updateconfig",
"listname": "users",
"tid": "17",
"tpbxid": "xivo", "commandid": 5}
Server -> Client
{
"class": "getlist",
"function": "updateconfig",
"listname": "users",
"tid": "17",
"tipbxid": "xivo",
"timenow": 1362741166.4,
"config": {
"enablednd": 0, "destrna": "", "enablerna": 0, "enableunc": 0, "destunc": "", "destbusy": "", "enablebusy": 0, "enablexfer": 1,
"firstname": "Alice", "lastname": "Bouzat", "fullname": "Alice Bouzat",
"voicemailid": null, "incallfilter": 0, "enablevoicemail": 0, "agentid": 2, "linelist": ["7"], "mobilephonenumber": ""}
}
Client -> Server
{"class": "getlist", "commandid": 495252308, "function": "listid", "listname": "phones", "tipbxid": "xivo"}
Server > Client
{"class": "getlist", "function": "listid", "list": ["1", "3", "2", "5", "14", "7", "6", "9", "8"],
"listname": "phones", "timenow": 1364994093.38, "tipbxid": "xivo"}
Individual phone configuration request:
{"class": "getlist", "commandid": 704096693, "function": "updateconfig", "listname": "phones", "tid": "3", "tipbxid": "xivo"}
Server > Client
{"class": "getlist",
"config": {"allowtransfer": null, "identity": "SIP/ihvbur", "iduserfeatures": 1,
"initialized": null, "number": "1000"},
"function": "updateconfig", "listname": "phones", "tid": "3", "timenow": 1364994093.43, "tipbxid": "xivo"}
Client -> Server
{"class": "getlist", "commandid": 1431355191, "function": "listid", "listname": "agents", "tipbxid": "xivo"}
Client -> Server
{"class": "getlist", "commandid": 719950939, "function": "listid", "listname": "queues", "tipbxid": "xivo"}
Server -> Client
{"function": "listid", "listname": "queues", "tipbxid": "xivo",
"list": ["1", "10", "3", "2", "5", "4", "7", "6", "9", "8"], "timenow": 1382704649.64, "class": "getlist"}
tid is the id returned in the list field of the getlist response message
Client -> Server
{"commandid":7,"class":"getlist","tid":"3","tipbxid":"xivo","function":"updateconfig","listname":"queues"}
Server -> Client
{
"function": "updateconfig", "listname": "queues", "tipbxid": "xivo", "timenow": 1382704649.69, "tid": "3",
"config":
{"displayname": "red", "name": "red", "context": "default", "number": "3002"},
"class": "getlist"}
Client -> Server
{"class": "getlist", "commandid": 1034160761, "function": "listid", "listname": "voicemails", "tipbxid": "xivo"}
Client -> Server
{"class": "getlist", "commandid": 964899043, "function": "listid", "listname": "queuemembers", "tipbxid": "xivo"}
Server -> Client
{"function": "listid", "listname": "queuemembers", "tipbxid": "xivo",
"list": ["Agent/2501,blue", "Agent/2500,yellow", "Agent/2002,yellow", "Agent/2003,__switchboard",
"Agent/2003,blue", "Agent/108,blue", "Agent/2002,blue"],
"timenow": 1382717016.23,
"class": "getlist"}
Client -> Server
{"class": "faxsend",
"filename": "contract.pdf",
"destination", 41400,
"size": 100000,
"data": "<base64 of the fax content>"}
Server -> Client
- pages: number of pages sent (
NULL
if FAILED) - status
- FAILED: Failed to send fax.
- PRESENDFAX: Fax number exist and converting pdf->tiff has been done.
- SUCCESS: Fax sent with success.
{"class": "fax_progress", "status": "SUCCESS", "pages": 2 }
- destination can be any number
- destination can be a pseudo URL of the form “type:ibpx/id”
Client -> Server
{
"class": "ipbxcommand",
"command": "dial",
"commandid": <commandid>,
"destination": "exten:xivo/<extension>"
}
For example :
{
"class": "ipbxcommand",
"command": "dial",
"commandid": 1683305913,
"destination": "exten:xivo/1202"
}
The server will answer with either an error or a success:
{
"class": "ipbxcommand",
"error_string": "unreachable_extension:1202",
}
{
"class": "dial_success",
"exten": "1202"
}
Transfer the current call to a given voicemail and listen to the message before completing the transfer.
Client -> Server
{
"class": "attended_transfer_voicemail",
"voicemail": "<voicemail number>"
}
Transfer the current call to a given voicemail.
Client -> Server
{
"class": "blind_transfer_voicemail",
"voicemail": "<voicemail number>"
}
Once the network is connected at the socket level, the login process requires three steps. If one of these steps is omitted, the connection is reset by the cti server.
- login_id, the username is sent as a login to the cti server, cti server answers by giving a sessionid
- login_pass, the password is sent to the cti server, cti server answers by giving a capaid
- login_capas, the capaid is returned to the server with the user’s availability, cti server answers with a list of info relevant to the user
{
"commandid": <commandid>,
"class": "login_id",
}
- class: defined what class of command use.
- commandid : a unique integer number.
Client -> Server
{
"class": "login_id",
"commandid": 1092130023,
"company": "default",
"ident": "X11-LE-24079",
"lastlogout-datetime": "2013-02-19T11:13:36",
"lastlogout-stopper": "disconnect",
"userlogin": <userlogin>,
"xivoversion": "<cti protocol version>"
}
Server -> Client
{
"class": "login_id",
"sessionid": "21UaGDfst7",
"timenow": 1361268824.64,
"xivoversion": "<cti protocol version>"
}
Note
sessionid is used to calculate the hashed password in next step
Client -> Server
{
"class": "login_pass",
"password": "secret",
"commandid": <commandid>
}
Server -> Client
{
"capalist": [
2
],
"class": "login_pass",
"replyid": 1646064863,
"timenow": 1361268824.68
}
If no CTI profile is defined on XiVO for this user, the following message will be sent:
{
"error_string": "capaid_undefined",
"class": "login_pass",
"replyid": 1646064863,
"timenow": 1361268824.68
}
Note
the first element of the capalist is used in the next step login_capas
Client -> Server
{
"capaid": 3,
"commandid": <commandid>,
"state": "available",
"class": "login_capas"
}
Server -> Client
First message, describes all the capabilities of the client, configured at the server level
presence : actual presence of the user
userid : the user id, can be used as a reference
- capas
- userstatus : a list of available statuses
- status name
- color
- selectionnable status from this status
- default action to be done when this status is selected
- long name
services : list of availble services
phonestatus : list of available phonestatuses with default colors and descriptive names
capaxlets : List of xlets configured for this profile
appliname
{
"class": "login_capas"
"presence": "available",
"userid": "3",
"ipbxid": "xivo",
"timenow": 1361440830.99,
"replyid": 3,
"capas": {
"preferences": false,
"userstatus": {
"available": { "color": "#08FD20",
"allowed": ["available", "away", "outtolunch", "donotdisturb", "berightback"],
"actions": {"enablednd": "false"}, "longname": "Disponible"
},
"berightback": { "color": "#FFB545",
"allowed": ["available", "away", "outtolunch", "donotdisturb", "berightback"],
"actions": {"enablednd": "false"}, "longname": "Bient\u00f4t de retour"
},
"disconnected": { "color": "#202020",
"actions": {"agentlogoff": ""}, "longname": "D\u00e9connect\u00e9"
},
/* a list of other status depends on the cti server configuration */
},
"services": ["fwdrna", "fwdbusy", "fwdunc", "enablednd"],
"phonestatus": {
"16": {"color": "#F7FF05", "longname": "En Attente"},
"1": {"color": "#FF032D", "longname": "En ligne OU appelle"},
"0": {"color": "#0DFF25", "longname": "Disponible"},
"2": {"color": "#FF0008", "longname": "Occup\u00e9"},
"-1": {"color": "#000000", "longname": "D\u00e9sactiv\u00e9"},
"4": {"color": "#FFFFFF", "longname": "Indisponible"},
"-2": {"color": "#030303", "longname": "Inexistant"},
"9": {"color": "#FF0526", "longname": "(En Ligne OU Appelle) ET Sonne"},
"8": {"color": "#1B0AFF", "longname": "Sonne"}
}
},
"capaxlets": [["identity", "grid"], ["search", "tab"], ["customerinfo", "tab", "1"], ["fax", "tab", "2"], ["dial", "grid", "2"], ["tabber", "grid", "3"], ["history", "tab", "3"], ["remotedirectory", "tab", "4"], ["features", "tab", "5"], ["people", "tab", "6"], ["conference", "tab", "7"]],
"appliname": "Client",
}
Second message describes the current user configuration
{
"function": "updateconfig",
"listname": "users",
"tipbxid": "xivo",
"timenow": 1361440830.99,
"tid": "3",
"config": {"enablednd": false},
"class": "getlist"
}
Third message describes the current user status
{
"function": "updatestatus",
"listname": "users",
"status": {"availstate": "available"},
"tipbxid": "xivo",
"tid": "3",
"class": "getlist",
"timenow": 1361440830.99
}
This message is received when a call form is submitted from a client to the XiVO.
Client -> Server
{
"class": "call_form_result",
"commandid": <commandid>,
"infos": {"buttonname": "saveandclose",
"variables": {"XIVOFORM_varname1": "value1",
"XIVOFORM_varname2": "value2"}}
}
- size : Size of the list to be sent by the server
Client -> Server
{
"class": "history",
"commandid": <commandid>
"size": "8",
"xuserid": "<xivoid>/<userfeaturesid>",
}
Server > Client
Send back a table of calls :
- duration in seconds
- extension: caller/destination extension
- fullname: caller ID name
- mode
- 0 : sent calls
- 1 : received calls
- 2 : missed calls
{
"class": "history",
"history": [
{"calldate": "2013-03-29T08:44:35.273998",
"duration": 30.148765,
"extension": "*844201",
"fullname": "Alice Wonderland",
"mode": 0},
{"calldate": "2013-03-28T16:56:48.071213",
"duration": 58.134744,
"extension": "41400",
"fullname": "41400"}
"mode": 1},
],
"replyid": 529422441,
"timenow": 1364571477.33
}
Client > Server
{
"class": "chitchat",
"alias": "Alice",
"text": "Lorem ipsum dolor sit amet, consectetur adipiscing elit. Suspendisse venenatis velit nibh, ac condimentum felis rutrum id.",
"to": [<xivo_uuid>, <user_uuid>],
"commandid": <commandid>
}
Server > Client
The following message is received by the remote XiVO client
{
"class": "chitchat",
"from": [<xivo_uuid>, <user_uuid>],
"to": [<xivo_uuid>, <user_uuid>]
"alias": "Alice",
"text": "Lorem ipsum dolor sit amet, consectetur adipiscing elit. Suspendisse venenatis velit nibh, ac condimentum felis rutrum id.",
}
Request directory information, names matching pattern ignore case.
Client -> Server
{
"class": "directory",
"commandid": 1079140548,
"pattern": "pau"
}
Server > Client
{
"class": "directory",
"headers": ["Nom", "Num\u00e9ro", "Mobile", "Autre num\u00e9ro", "E-mail", "Fonction", "Site", "Source"],
"replyid": 1079140548,
"resultlist": ["Claire Mapaurtal;;+33644558899;31256;cmapaurtal@societe.com;;;",
"Paul Salvadier;+33445236988;+33678521430;31406;psalvadier@societe.com;;;"],
"status": "ok",
"timenow": 1378798928.26
}
parking
keepalive
availstate
getipbxlist
{
"class": "getipbxlist",
"commandid": <commandid>
}
This command will trigger a Relations message.
Client -> Server
{
"class": "get_relations"
}
Client -> Server
{
"class": "people_headers",
}
Server -> Client
{
"class": "people_headers_result",
"column_headers": ["Status", "Name", "Number"],
"column_types": [null, null, "number"],
}
Client -> Server
{
"class": "people_search",
"pattern": <pattern>,
}
Server -> Client
{
"class": "people_search_result",
"term": "Bob",
"column_headers": ["Firstname", "Lastname", "Phone number", "Mobile", "Fax", "Email", "Agent"],
"column_types": [null, "name", "number_office", "number_mobile", "fax", "email", "relation_agent"],
"results": [
{
"column_values": ["Bob", "Marley", "5555555", "5556666", "5553333", "mail@example.com", null],
"relations": {
"agent_id": null,
"user_id": null,
"endpoint_id": null,
"source_entry_id": null
},
"source": "my_ldap_directory"
}, {
"column_values": ["Charlie", "Boblin", "5555556", "5554444", "5552222", "mail2@example.com", null],
"relations": {
"agent_id": 12,
"user_id": 34,
"endpoint_id": 56,
"source_entry_id": "34"
},
"source": "internal"
}
]
}
This message can currently only be received as a response to the Get relations command.
- The xivo_uuid is the id of the server
- The user_id is the id of the current user.
- The endpoint_id is the id of the line of the current user or null.
- The agent_id is the id of the agent of the current user or null.
Server -> Client
{
"class": "relations",
"data": {"xivo_uuid": <the xivo uuid>,
"user_id": <the user id>,
"endpoint_id": <the endpoint id>,
"agent_id": <the agent id>}
}
Client -> Server
{
"class": "people_favorites",
}
Server -> Client
{
"class": "people_favorites_result",
"column_headers": ["Firstname", "Lastname", "Phone number", "Mobile", "Fax", "Email", "Agent", "Favorites"],
"column_types": [null, "name", "number_office", "number_mobile", "fax", "email", "relation_agent", "favorite"],
"results": [
{
"column_values": ["Bob", "Marley", "5555555", "5556666", "5553333", "mail@example.com", null, true],
"relations": {
"agent_id": null,
"user_id": null,
"endpoint_id": null,
"source_entry_id": "55"
},
"source": "my_ldap_directory"
}, {
"column_values": ["Charlie", "Boblin", "5555556", "5554444", "5552222", "mail2@example.com", null, true],
"relations": {
"agent_id": 12,
"user_id": 34,
"endpoint_id": 56,
"source_entry_id": "34"
},
"source": "internal"
}
]
}
Client -> Server
{
"class": "people_set_favorite",
"source": "my_ldap_directory"
"source_entry_id": "55"
"favorite": true
}
Server -> Client
{
"class": "people_favorite_update",
"source": "my_ldap_directory"
"source_entry_id": "55"
"favorite": true
}
The STARTTLS command is used to upgrade a connection to use SSL. Once connected, the server send a starttls offer to the client which can reply with a starttls message including the status field. The server will then send a starttls message back to the client with the same status and start the handshake if the status is true.
Server -> Client
{
"class": "starttls"
}
Client -> Server -> Client
{
"class": "starttls",
"status": true
}
Note
a client which does not reply to the starttls offer will keep it’s unencrypted connection.
Client -> Server
{
"class": "people_personal_contacts"
}
Server -> Client
{
"class": "people_personal_contacts_result",
"column_headers": ["Firstname", "Lastname", "Phone number", "Mobile", "Fax", "Email", "Agent", "Favorites", "Personal"],
"column_types": [null, "name", "number_office", "number_mobile", "fax", "email", "relation_agent", "favorite", "personal"],
"results": [
{
"column_values": ["Bob", "Marley", "5555555", "5556666", "5553333", "mail@example.com", null, false, true],
"relations": {
"agent_id": null,
"user_id": null,
"endpoint_id": null,
"source_entry_id": "abcd-12"
},
"source": "personal"
}, {
"column_values": ["Charlie", "Boblin", "5555556", "5554444", "5552222", "mail2@example.com", null, false, true],
"relations": {
"agent_id": null,
"user_id": null,
"endpoint_id": null,
"source_entry_id": "efgh-34"
},
"source": "personal"
}
]
}
Client -> Server
{
"class": "people_purge_personal_contacts",
}
Server -> Client
{
"class": "people_personal_contacts_purged",
}
Client -> Server
{
"class": "people_personal_contact_raw",
"source": "personal",
"source_entry_id": "abcd-1234"
}
Server -> Client
{
"class": "people_personal_contact_raw_result",
"source": "personal",
"source_entry_id": "abcd-1234",
"contact_infos": {
"firstname": "Bob",
"lastname": "Wonderland"
...
}
}
Client -> Server
{
"class": "people_create_personal_contact",
"contact_infos": {
"firstname": "Bob",
"lastname": "Wonderland",
...
}
}
Server -> Client
{
"class": "people_personal_contact_created"
}
Client -> Server
{
"class": "people_delete_personal_contact",
"source": "personal",
"source_entry_id": "abcd-1234"
}
Server -> Client
{
"class": "people_personal_contact_deleted",
"source": "personal",
"source_entry_id": "abcd-1234"
}
Client -> Server
{
"class": "people_edit_personal_contact",
"source": "personal",
"source_entry_id": "abcd-1234",
"contact_infos": {
"firstname": "Bob",
"lastname": "Wonderland",
...
}
}
Server -> Client
{
"class": "people_personal_contact_raw_update",
"source": "personal",
"source_entry_id": "abcd-1234"
}
Client -> Server
{
"class": "people_import_personal_contacts_csv",
"csv_contacts": "firstname,lastname\r\nBob,the Builder\r\n,Alice,Wonderland\r\n,BobMissingFields\r\n"
}
Server -> Client
{
"class": "people_import_personal_contacts_csv_result",
"created_count": 2,
"failed": [
{
"line": 3,
"errors": [
"missing fields"
]
}
]
}
Client -> Server
{
"class": "people_export_personal_contacts_csv",
}
Server -> Client
{
"class": "people_export_personal_contacts_csv_result",
"csv_contacts": "firstname,lastname\r\nBob,the Builder\r\n,Alice,Wonderland\r\n"
}
- class : featuresput
- function : incallfilter
- value : true, false activate deactivate filtering
Client -> Server
{"class": "featuresput", "commandid": 1326845972, "function": "incallfilter", "value": true}
Server > Client
{
"class": "getlist",
"config": {"incallfilter": true},
"function": "updateconfig",
"listname": "users",
"tid": "2",
"timenow": 1361456398.52, "tipbxid": "xivo" }
- function : enablednd
- value : true, false activate deactivate DND
Client -> Server
{"class": "featuresput", "commandid": 1088978942, "function": "enablednd", "value": true}
Server > Client
{
"class": "getlist",
"config": {"enablednd": true},
"function": "updateconfig",
"listname": "users",
"tid": "2",
"timenow": 1361456614.55, "tipbxid": "xivo"}
- function : enablerecording
- value : true, false
Activate / deactivate recording for a user, extension call recording has to be activated :
Client -> Server
{"class": "featuresput", "commandid": 1088978942, "function": "enablerecording", "value": true, "target" : "7" }
Server > Client
{
"class": "getlist",
"config": {"enablerecording": true},
"function": "updateconfig",
"listname": "users",
"tid": "7",
"timenow": 1361456614.55, "tipbxid": "xivo"}
Forward the call at any time, call does not reach the user
- function : fwd
Client -> Server
{
"class": "featuresput", "commandid": 2082138822, "function": "fwd",
"value": {"destunc": "1002", "enableunc": true}
}
Server > Client
{
"class": "getlist",
"config": {"destunc": "1002", "enableunc": true},
"function": "updateconfig",
"listname": "users",
"tid": "2",
"timenow": 1361456777.98, "tipbxid": "xivo"}
Forward the call to another destination if the user does not answer
- function : fwd
Client -> Server
{
"class": "featuresput", "commandid": 1705419982, "function": "fwd",
"value": {"destrna": "1003", "enablerna": true}
}
Server > Client
{
"class": "getlist",
"config": {"destrna": "1003", "enablerna": true},
"function": "updateconfig",
"listname": "users",
"tid": "2",
"timenow": 1361456966.89, "tipbxid": "xivo" }
Forward the call to another destination when the user is busy
- function : fwd
Client -> Server
{
"class": "featuresput", "commandid": 568274890, "function": "fwd",
"value": {"destbusy": "1009", "enablebusy": true}
}
Server > Client
{
"class": "getlist",
"config": {"destbusy": "1009", "enablebusy": true},
"function": "updateconfig",
"listname": "users",
"tid": "2",
"timenow": 1361457163.77, "tipbxid": "xivo"
}
This message can be sent from the client to enable statitics update on queues
Client -> Server
{"commandid":36,"class":"subscribetoqueuesstats"}
``Server > Client``
When statistic update is enable by sending message Subscribe to queues stats.
The first element of the message is the queue id
{"stats": {"10": {"Xivo-LoggedAgents": 0}},
"class": "getqueuesstats", "timenow": 1384509582.88}
{"stats": {"1": {"Xivo-WaitingCalls": 0}},
"class": "getqueuesstats", "timenow": 1384509582.89}
{"stats": {"1": {"Xivo-TalkingAgents": "0", "Xivo-AvailableAgents": "1", "Xivo-EWT": "6"}},
"class": "getqueuesstats", "timenow": 1384512350.25}
These messages can also be received without any request as unsolicited messages.
User status is to manage user presence
- Request user status update
Client -> Server
{"class": "getlist", "commandid": 107712156,
"function": "updatestatus",
"listname": "users",
"tid": "14", "tipbxid": "xivo"}
Server > Client
{"class": "getlist",
"function": "updatestatus",
"listname": "users",
"status": {"availstate": "outtolunch", "connection": "yes"},
"tid": "1", "timenow": 1364994093.48, "tipbxid": "xivo"}
- Change User status
Client -> Server
{"availstate": "away",
"class": "availstate",
"commandid": 1946092392,
"ipbxid": "xivo",
"userid": "1"}
Server > Client
{"class": "getlist",
"function": "updatestatus",
"listname": "users",
"status": {"availstate": "away"},
"tid": "1", "timenow": 1370523352.6, "tipbxid": "xivo"}
- tid is the line id, found in linelist from message User configuration
Client -> Server
{"class": "getlist", "commandid": 107712156,
"function": "updatestatus",
"listname": "phones", "tid": "8", "tipbxid": "xivo"}
Server > Client
{"class": "getlist",
"function": "updatestatus",
"listname": "phones",
"status": {"hintstatus": "0"},
"tid": "1",
"timenow": 1364994093.48,
"tipbxid": "xivo"}
Client -> Server
{"commandid":17,"class":"getlist","tid":"8","tipbxid":"xivo","function":"updatestatus","listname":"queues"}
Server > Client
{"function": "updatestatus", "listname": "queues", "tipbxid": "xivo", "timenow": 1382710430.54,
"status": {"agentmembers": ["1","5"], "phonemembers": ["8"]},
"tid": "8", "class": "getlist"}
- tid is the agent id.
Client -> Server
{"class": "getlist",
"commandid": <random_integer>,
"function": "updatestatus",
"listname": "agents",
"tid": "635",
"tipbxid": "xivo"}
Server > Client
{"class": "getlist",
"listname": "agents",
"function": "updatestatus",
"tipbxid": "xivo",
"tid": 635,
"status": {
"availability": "logged_out",
"availability_since": 1370868774.74,
"channel": null,
"groups": [],
"on_call_acd": false,
"on_call_nonacd": false,
"on_wrapup": false,
"phonenumber": null,
"queues": [
"113"
]
}}
availability can take the values:
- logged_out
- available
- unavailable
- on_call_nonacd_incoming_internal
- on_call_nonacd_incoming_external
- on_call_nonacd_outgoing_internal
- on_call_nonacd_outgoing_external
availability_since is the timestamp of the last availability change
queues is the list of queue ids from which the agent receives calls
This allows the switchboard operator to answer an incoming call or unhold a call on-hold.
{"class": "answer", "uniqueid": "12345667.89"}
These messages are received whenever one of the following corresponding event occurs: sheet message on incoming calls, or updatestatus when a phone status changes.
This message is received to display customer information if configured at the server side
{
"timenow": 1361444639.61,
"class": "sheet",
"compressed": true,
"serial": "xml",
"payload": "AAADnnicndPBToNAEAbgV1n3XgFN1AP...................",
"channel": "SIP/e6fhff-00000007"
}
How to decode payload :
>>> b64content = base64.b64decode(<payload content>)
>>> # 4 first cars are the encoded lenght of the xml string (in Big Endian format)
>>> xmllen = struck.unpack('>I',b64content[0:4])
>>> # the rest is a compressed xml string
>>> xmlcontent = zlib.decompress(toto[4:])
>>> print xmlcontent
<?xml version="1.0" encoding="utf-8"?>
<profile>
<user>
<internal name="ipbxid"><![CDATA[xivo]]></internal>
<internal name="where"><![CDATA[dial]]></internal>
<internal name="channel"><![CDATA[SIP/barometrix_jyldev-00000009]]></internal>
<internal name="focus"><![CDATA[no]]></internal>
<internal name="zip"><![CDATA[1]]></internal>
<sheet_qtui order="0010" name="qtui" type="None"><![CDATA[]]></sheet_qtui>
<sheet_info order="0010" name="Nom" type="title"><![CDATA[0230210083]]></sheet_info>
<sheet_info order="0030" name="Origine" type="text"><![CDATA[extern]]></sheet_info>
<sheet_info order="0020" name="Num\xc3\xa9ro" type="text"><![CDATA[0230210083]]></sheet_info>
<systray_info order="0010" name="Nom" type="title"><![CDATA[Maric\xc3\xa9 Sapr\xc3\xaftch\xc3\xa0]]></systray_info>
<systray_info order="0030" name="Origine" type="body"><![CDATA[extern]]></systray_info>
<systray_info order="0020" name="Num\xc3\xa9ro" type="body"><![CDATA[0230210083]]></systray_info>
</user>
</profile>
The xml file content is defined by the following xsd file:
xivo-javactilib/src/main/xsd/sheet.xsd
(online version)
Received when a phone status change
- class : getlist
- function : updatestatus
- listname : phones
{
"class": "getlist",
"function": "updatestatus",
"listname": "phones",
"tipbxid": "xivo",
"timenow": 1361447017.29,
.........
}
tid is the the object identification
Example of phone messages received when a phone is ringing :
{.... "status": {"hintstatus": "0"}, "tid": "3"}
{.... "status": {"hintstatus": "8"}, "tid": "3"}
The register_agent_status_update command is used to register to the status updates of a list of agent. Once registered to a agent’s status, the client will receive all Agent status update events for the registered agents.
This command should be sent when an agent is displayed in the people xlet to be able to update the agent status icon.
The Unregister agent status update command should be used to stop receiving updates.
Client -> Server
{
"class": "register_agent_status_update",
"agent_ids": [["<xivo-uuid>", "<agent-id1>"],
["<xivo-uuid>", "<agent-id2>"],
...,
["<xivo-uuid>", "<agent-idn>"]],
"commandid": <commandid>
}
The unregister_agent_status_update command is used to unregister from the status updates of a list of agent.
Once unregistered, the client will stop receiving the Agent status update events for the specified agents.
Client -> Server
{
"class": "unregister_agent_status_update",
"agent_ids": [["<xivo-uuid>", "<agent-id1>"],
["<xivo-uuid>", "<agent-id2>"],
...,
["<xivo-uuid>", "<agent-idn>"]],
"commandid": <commandid>
}
The agent_status_update event is received when the presence of an agent changes.
To receive this event, the user must first register to the event for a specified agent using the Register agent status update command.
To stop receiving this event, the user must send the Unregister agent status update command.
- data, a dictionary containing 3 fields:
- agent_id, is an integer containing the ID of the user affected by this status change
- xivo_uuid: a string containing the UUID of the XiVO that sent the status update
- status: a string containing the new status, “logged_in” or “logged_out”
Server -> Client
{
"class": "agent_status_update",
"data": {
"agent_id": 42,
"xivo_uuid": "<the-xivo-uuid>",
"status": "<status-name>"
}
}
The agent_status_update event contains the same data as the agent_status_update. The latter should be preferred to the former for uses that do not require a persistent connection to xivo-ctid.
The register_endpoint_status_update command is used to register to the status updates of a list of lines. Once registered to a endpoint’s status, the client will receive all Endpoint status update events for the registered agents.
This command should be sent when a endpoint is displayed in the people xlet to be able to update the agent status icon.
The Unregister endpoint status update command should be used to stop receiving updates.
Client -> Server
{
"class": "register_endpoint_status_update",
"endpoint_ids": [["<xivo-uuid>", "<endpoint-id1>"],
["<xivo-uuid>", "<endpoint-id2>"],
...,
["<xivo-uuid>", "<endpoint-idn>"]],
"commandid": <commandid>
}
The unregister_endpoint_status_update command is used to unregister from the status updates of a list of agent.
Once unregistered, the client will stop receiving the Endpoint status update events for the specified agents.
Client -> Server
{
"class": "unregister_endpoint_status_update",
"endpoint_ids": [["<xivo-uuid>", "<endpoint-id1>"],
["<xivo-uuid>", "<endpoint-id2>"],
...,
["<xivo-uuid>", "<endpoint-idn>"]],
"commandid": <commandid>
}
The endpoint_status_update event is received when the status of a line changes.
To receive this event, the user must first register to the event for a specified endpoint using the Register endpoint status update command.
To stop receiving this event, the user must send the Unregister endpoint status update command.
- data, a dictionary containing 3 fields:
- endpoint_id, is an integer containing the ID of the line affected by this status change
- xivo_uuid: a string containing the UUID of the XiVO that sent the status update
- status: an integer matching an entry in the cti hint configuration
Server -> Client
{
"class": "endpoint_status_update",
"data": {
"endpoint_id": 42,
"xivo_uuid": "<the-xivo-uuid>",
"status": <hint-status>
}
}
The endpoint_status_update event contains the same data as the endpoint_status_update. The latter should be preferred to the former for uses that do not require a persistent connection to xivo-ctid.
The register_user_status_update command is used to register to the status updates of a list of user. Once registered to a user’s status, the client will receive all User status update events for the registered users.
This command should be sent when a user is displayed in the people xlet to be able to update the presence status icon.
The Unregister user status update command should be used to stop receiving updates.
Client -> Server
{
"class": "register_user_status_update",
"user_ids": [["<xivo-uuid>", "<user-uuid1>"],
["<xivo-uuid>", "<user-uuid2>"],
...,
["<xivo-uuid>", "<user-uuidn>"]],
"commandid": <commandid>
}
The unregister_user_status_update command is used to unregister from the status updates of a list of user.
Once unregistered, the client will stop receiving the User status update events for the specified users.
Client -> Server
{
"class": "unregister_user_status_update",
"user_ids": [["<xivo-uuid>", "<user-uuid1>"],
["<xivo-uuid>", "<user-uuid2>"],
...,
["<xivo-uuid>", "<user-uuidn>"]],
"commandid": <commandid>
}
The user_status_update event is received when the presence of a user changes.
To receive this event, the user must first register to the event for a specified user using the Register user status update command.
To stop receiving this event, the user must send the Unregister user status update command.
- data, a dictionary containing the following fields:
- user_uuid, a string containing the UUID of the user.
- user_id, an integer containing the ID of the user.
- xivo_uuid: a string containing the UUID of the XiVO that sent the status update
- status: a string containing the new status of the user based on the cti profile configuration
Note
When multiple XiVO share user statuses, the cti profile configuration for presences and phone statuses should match on all XiVO to be displayed properly
Server -> Client
{
"class": "user_status_update",
"data": {
"user_uuid": "<the-user-uuid>",
"user_id": <the-user-id>,
"xivo_uuid": "<the-xivo-uuid>",
"status": "<status-name>"
}
}
Warning
The user_id field is DEPRECATED and should not be used. Use the user_uuid field instead.
In the git repository git://github.com/xivo-pbx/xivo-ctid.git
- cti_config handles the configuration coming from the WEBI
- interfaces/interface_ami, together with asterisk_ami_definitions, amiinterpret and xivo_ami handle the AMI connections (asterisk)
- interfaces/interface_info handles the CLI-like connections
- interfaces/interface_webi handles the requests and signals coming from the WEBI
- interfaces/interface_cti handles the clients’ connections, with the help of client_connection, and it often involves cti_command too
- innerdata is meant to be the place where all statuses are computed and stored
The main loop uses select() syscall to dispatch the tasks according to miscellaneous incoming requests.
Requirements for innerdata:
- the properties fetched from the WEBI configuration shall be stored in the relevant xod_config structure
- the properties fetched from elsewhere shall be stored in the relevant xod_status structure
- at least two kinds of objects are not “predefined” (as are the phones or the queues, for instance)
- the channels (in the asterisk SIP/345-0x12345678 meaning)
- the group and queue members shall be handled in a special way each
The purpose of the ‘relations’ field, in the various structures is to keep track of relations and cross-relations between different objects (a phone logged in as an agent, itself in a queue, itself called by some channels belonging to phones ...).
Messages sent from the CTI clients to the server are received by the CTIServer class.
The CTIServer then calls interface_cti.CTI
class manage_connection
method.
The interface_cti
uses his _cti_command_handler
member to parse and run the command.
The CTICommandHandler
get a list of classes that handle this message from the CTICommandFactory
.
Then the the interface_cti.CTI
calls run_commands
on the handler, which returns a list of all commands replies.
To implement a new message in the protocol you have to create a new class that inherits the CTICommand
class.
Your new class should have a static member caller required_fields
which is a list of required fields for this class.
Your class should also have a conditions
static member which is a list of tupples of conditions to detect that
an incoming message matches this class. The __init__
of your class is responsible for the initialization of
it’s fields and should call super(<ClassName>, self).__init__(msg)
. Your class should register itself to the CTICommandFactory
.
from xivo_cti.cti.cti_command import CTICommand
from xivo_cti.cti.cti_command_factory import CTICommandFactory
class InviteConfroom(CTICommand):
required_fields = ['class', 'invitee']
conditions = [('class', 'invite_confroom')]
def __init__(self):
super(InviteConfroom, self).__init__(msg)
self._invitee = msg['invitee']
CTICommandFactory.register_class(InviteConfroom)
Each CTI commands has a callback list that you can register to from anywhere. Each callback function will be called when this message is received with the command as parameter.
Refer to MeetmeList.__init__
for a callback registration example and to MeetmeList.invite
for the implementation of a callback.
from xivo_cti.cti.commands.invite_confroom import InviteConfroom
class MySuperClass(object):
def __init__(self):
InviteConfroom.register_callback(self.invite_confroom_handler)
def invite_confroom_handler(self, invite_confroom_command):
# Do your stuff here.
if ok:
return invite_confroom_command.get_message('Everything is fine')
else:
return invite_confroom_command.get_warning('I don't know you, go away', True)
Note
The client’s connection is injected in the command instance before calling callbacks functions.
The client’s connection is an interface_cti.CTI
instance.
Database¶
Strating with XiVO 14.08, the database migration is handled by alembic.
The XiVO migration scripts can be found in the xivo-manage-db repository.
On a XiVO, they are located in the /usr/share/xivo-manage-db
directory.
To add a new migration script from your developer machine, go into the root
directory of the xivo-manage-db repository. There should be an alembic.ini
file in this directory. You can then use the following command to create a new
migration script:
alembic revision -m "<description>"
This will create a file in the alembic/versions
directory, which you’ll have to edit.
When the migration scripts are executed, they use a connection to the database
with the role/user asterisk
. This means that new objects that are created
in the migration scripts will be owned by the asterisk
role and it is thus
not necessary (nor recommended) to explicitly grant access to objects to the
asterisk role (i.e. no “GRANT ALL” command after a “CREATE TABLE” command).
Diagrams¶
Graphs representing states and transitions between agent states. Used in Agent status dashboard and agent list.
Provisioning¶
This section describes the informations and tools for xivo-provd.
This page considers the configuration files of the DHCP server in /etc/dhcp/dhcpd_update/
.
The files are updated with the command dhcpd-update
, which is also run when updating the provisioning plugins. This commands fetches configurations files from the provd.xivo.solutions
server.
- On a XiVO, edit manually the file
/etc/dhcp/dhcpd_update/*.conf
service isc-dhcp-server restart
- If errors are shown in
/var/log/daemon.log
, check your modifications
- Edit the files in the Git repo
xivo-provd-plugins
, directorydhcp/
- Push your modifications
- Go in
dhcp/
- Run
make upload
to push your modifications toprovd.xivo.solutions
. There is notesting
version of these files. Once the files are uploaded, they are available for all XiVO installations.
Most plugin-related files are available in the xivo-provd-plugins repository. Following examples are relative to the repository directory tree. Any modifications should be preceeded by a git pull.
We will be using the xivo-cisco-spa plugins family as an example on this page
There is one directory per family. Here is the directory structure for xivo-cisco-spa
:
plugins/xivo-cisco-spa/
+-- model_name_xxx
+-- model_name_xxx
+-- common
+-- build.py
Every plugin has a folder called common
which regoups common ressources for each model.
Every model has its own folder with its version number.
After modifying a plugin, you must increment the version number.
You can modifiy the file plugin-info
to change the version number:
plugins/xivo-cisco-spa/
+-- model_name_xxx
+-- plugin-info
Important
If ever you modify the folder common
, you must increment the version number of all the models.
Let us suppose we want to update firmwares for xivo-snom from 8.7.3.25 to 8.7.3.25 5. Here are the steps to follow :
- Copy folder plugins/xivo-snom/8.7.3.25 to plugins/xivo-snom/8.7.3.25.5
- Update VERSION number in plugins/xivo-snom/8.7.3.25.5/entry.py
- Update VERSION number in plugins/xivo-snom/8.7.3.25.5/plugin-info
- Download new firmwares (.bin files from snom website)
- Update VERSION number and URIs in plugins/xivo-snom/8.7.3.25.5/pkgs/pkgs.db (with uris of downloaded files from snom website)
- Update sizes and sha1sums in plugins/xivo-snom/8.7.3.25.5/pkgs/pkgs.db (using helper script xivo-tools/dev-tools/check_fw)
- Update plugins/xivo-snom/build.py (duplicate and update section 8.7.3.25 > 8.7.3.25.5)
You have three different methods to test your changes on your development machine.
If the production version is 0.4, change the plugin version to 0.4.01, make your changes and upload to testing (see below).
Next modification will change the plugin version to 0.4.02, etc. When you are finished making changes, change the version to 0.5 and upload one last time.
Edit the files in /var/lib/xivo-provd/plugins
.
To apply your changes, go in xivo-provd-cli
and run:
plugins.reload('xivo-cisco-spa-7.5.4')
Edit /etc/xivo/provd/provd.conf
and add the line:
cache_plugin: True
Empty /var/cache/xivo-provd
and restart provd.
Make your changes in provd-plugins, update the plugin version to the new one and upload to testing (see below). Now, every time you uninstall/install the plugin, the new plugin will be fetched from testing, instead of being cached, even without changing the version.
Before updating a plugin, it must be passed through the testing phase. Once it has been approved it can be uploaded to the production server
Important
Before uploading a plugin in the testing provd repository, make sure to git pull the xivo-provd-plugins git repository.
To upload the modified plugin in the testing repo on provd.xivo.solutions, you can execute the following command:
$ make upload
Afterwards, in the web-interface, you must modify the URL in section
to:`http://provd.xivo.solutions/plugins/1/testing/`
You can then update the list of plugins and check the version number for the plugin that you modified. Don’t forget to install the plugin to test it.
Once checked, you must synchronize the plugin from testing to stable. If applicable, you should also update the archive repo.
To download the stable and archive plugins:
$ make download-stable
$ make download-archive
Go to the plugins/_build directory and delete the plugins that are going to be updated. Note that if you are not updating a plugin but you are instead removing it “once and for all”, you should instead move it to the archive directory:
$ rm -fi stable/xivo-cisco-spa*
Copy the files from the directory testing to stable:
$ cp testing/xivo-cisco-spa* stable
Go back to the plugins directory and upload the files to the stable and archive repo:
$ make upload-stable
$ make upload-archive
The file are now up to date and you can test by putting back the stable url in the web-interface’s configuration:
`http://provd.xivo.solutions/plugins/1/stable/`
Let’s suppose you have received a brand new SIP phone that is not supported by the provisioning system of XiVO. You would like to know if it’s possible to add auto-provisioning support for it. That said, you have never tested the phone before.
This guide will help you get through the different steps that are needed to add auto-provisioning support for a phone to XiVO.
Before continuing, you’ll need the following:
- a private LAN where only your phones and your test machines are connected to it, i.e. a LAN that you fully control.
Although it’s possible to do all the testing directly on a XiVO, it’s more comfortable and usually easier to do on a separate, dedicated machine.
That said, you’ll still need a XiVO near, since we’ll be doing the call testing part on it and not on a separate asterisk.
So, for the rest of this guide, we’ll suppose you are doing your tests on a Debian jessie with the following configuration:
Installed packages:
isc-dhcp-server tftpd-hpa apache2
Example content of the
/etc/dhcp/dhcpd.conf
file (restartisc-dhcp-server
after modification):ddns-update-style none; default-lease-time 7200; max-lease-time 86400; log-facility local7; subnet 10.34.1.0 netmask 255.255.255.0 { authoritative; range 10.34.1.200 10.34.1.250; option subnet-mask 255.255.255.0; option broadcast-address 10.34.1.255; option routers 10.34.1.6; option ntp-servers 10.34.1.6; option domain-name "my-domain.example.org"; option domain-name-servers 10.34.1.6; log(concat("[VCI: ", option vendor-class-identifier, "]")); }
Example content of the
/etc/default/tftpd-hpa
file (restarttftpd-hpa
after modifcation):TFTP_USERNAME="tftp" TFTP_DIRECTORY="/srv/tftp" TFTP_ADDRESS="0.0.0.0:69" TFTP_OPTIONS="--secure --verbose"
With this configuration, files served via TFTP will be in the /srv/tftp
directory and those served via HTTP in the /var/www
directory.
Adding auto-provisioning support for a phone is mostly a question of finding answers to the following questions.
Is it worth the time adding auto-provisioning support for the phone ?
Indeed. Adding quality auto-provisioning support for a phone to XiVO requires a non negligible amount of work, if you don’t meet any real problem and are comfortable with provisioning in XiVO. Not all phones are born equal. Some are cheap. Some are old and slow. Some are made to work on proprietary system and will only work in degraded mode on anything else.
That said, if you are uncertain, testing will help you clarifying your idea.
What is the vendor, model, MAC address and firmware version (if available) of your phone ?
Having the vendor and model name is essential when looking for documentation or other information. The MAC address will be needed later on for some tests, and it’s always good to know the firmware version of the phone if you are trying to upgrade to a newer firmware version and you’re having some troubles, and when reading the documentation.
Is the official administrator guide/documentation available publicly on the vendor web site ? Is it available only after registering and login to the vendor web site ?
Having access to the administrator guide/documentation of the phone is also essential. Once you’ve found it, download it and keep the link to the URL. If you can’t find it, it’s probably not worth going further.
Is the latest firmware of the phone available publicly on the vendor web site ? Is it available only after registering and login to the vendor web site ?
Good auto-provisioning support means you need to have an easy way to download the latest firmware of the phone. Ideally, this mean the firmware is downloadable from an URL, with no authentication whatsoever. In the worst case, you’ll need to login on some web portal before being able to download the firmware, which will be cumbersome to automatize and probably fragile. If this is the case, it’s probably not worth going further.
Does the phone need other files, like language files ? If so, are these files available publicly on the vendor web site ? After registering ?
Although you might not be able to answer to this question yet because you might not know if the phone needs such files to be either in English or in French (the two officially supported language in XiVO), you’ll need to have an easy access to these files if its the case.
Does the phone supports auto-provisioning via DHCP + HTTP (or TFTP) ?
The provisioning system in XiVO is based on the popular method of using a DHCP server to tell the phone where to download its configuration files, and a HTTP (or TFTP) server to serve these configuration files. Some phones support other methods of provisioning (like TR-069), but that’s of no use here. Also, if your phone is only configurable via its web interface, although it’s technically possible to configure it automatically by navigating its web interface, it’s an extremely bad idea since it’s impossible to guarantee that you’ll still be able to provision the phone on the next firmware release.
If the phone supports both HTTP and TFTP, pick HTTP, it usually works better with the provisioning server of XiVO.
What are the default usernames/passwords on the phone to access administrator menus (phone UI and web UI) ? How do you do a factory reset of the phone ?
Although this step is optional, it might be handy later to have these kind of information. Try to find them now, and note them somewhere.
What are the DHCP options and their values to send to the phones to tell it where its configuration files are located ?
Once you know that the phone supports DHCP + HTTP provisioning, the next question is what do you need to put in the DHCP response to tell the phone where its configuration files are located. Unless the admin documentation of the phone is really poor, this should not be too hard to find.
Once you have found this information, the easiest way to send it to the phone is to create a custom host declaration for the phone in the
/etc/dhcp/dhcpd.conf
file, like in this example:host my-phone { hardware ethernet 00:11:22:33:44:55; option tftp-server-name "http://169.254.0.1/foobar.cfg"; }
What are the configuration files the phone needs (filename and content) and what do we need to put in it for the phone to minimally be able to make and receive calls on XiVO ?
Now that you are able to tell your phone where to look for its configuration files, you need to write these files with the right content in it. Again, at this step, you’ll need to look through the documentation or examples to answer this question.
Note that you only want to have the most basic configuration here, i.e. only configure 1 line, with the right SIP registrar and proxy, and the associated username and password.
Do basic telephony services, like transfer, works correctly when using the phone buttons ?
On most phones, it’s possible to do transfer (both attended and direct), three-way conferences or put someone on hold directly from the phone. Do some tests to see if it works correctly.
Also at this step, it’s a good idea to check how the phone handle non-ascii characters, either in the caller ID or in its configuration files.
Does other “standard” features work correctly on the phone ?
For quality auto-provisioning support, you must find how to configure and make the following features work:
- NTP server
- MWI
- function keys (speed dial, BLF, directed pickup / call interception)
- timezone and DST support
- multi language
- DTMF
- hard keys, like the voicemail hard key on some phone
- non-ASCII labels (line name, function key label)
- non-ASCII caller ID
- backup proxy/registrar
- paging
Once you have answered all these questions, you’ll have a good idea on how the phone works and how to configure it. Next step would be to start the development of a new provd plugin for your phone for a specific firmware version.
FK = Funckey
HK = HardKey
Y = Supported
MN = Menu
N = Not supported
NT = Not tested
NYT = Not yet tested
SK = SoftKey
model | |
---|---|
Provisioning | Y |
H-A | Y |
Directory XIVO | Y |
Funckeys | 8 |
Supported programmable keys | |
User with supervision function | Y |
Group | Y |
Queue | Y |
Conference Room with supervision function | Y |
General Functions | |
Online call recording | N |
Phone status | Y |
Sound recording | Y |
Call recording | Y |
Incoming call filtering | Y |
Do not disturb | Y |
Group interception | Y |
Listen to online calls | Y |
Directory access | Y |
Filtering Boss - Secretary | Y |
Transfers Functions | |
Blind transfer | HK |
Indirect transfer | HK |
Forwards Functions | |
Disable all forwarding | Y |
Enable/Disable forwarding on no answer | Y |
Enable/Disable forwarding on busy | Y |
Enable/Disable forwarding unconditional | Y |
Voicemail Functions | |
Enable voicemail with supervision function | Y |
Reach the voicemail | Y |
Delete messages from voicemail | Y |
Agent Functions | |
Connect/Disconnect a static agent | Y |
Connect a static agent | Y |
Disconnect a static agent | Y |
Parking Functions | |
Parking | Y |
Parking position | Y |
Paging Functions | |
Paging | Y |
This is a configuration example to simulate the case of a hosted XiVO, i.e. an environment where:
- the XiVO has a public IP address
- the phones are behind a NAT
In this example, we’ll reproduce the following environment:

Phones behind a NAT
Where:
- the XiVO is installed inside a virtual machine
- the host machine is used as a router, a NAT and a DHCP server for the phones
- the phones are in a separate VLAN than the XiVO, and when they want to interact with it, they must pass through the NAT
With this setup, we could also put some phones in the same VLAN as the XiVO. We would then have a mixed environment, where some phones are behind the NAT and some phones aren’t.
Also, it’s easy to go from a non-NAT environment to a NAT environment with this setup. What you usually have to do is only to switch your phone from the “XiVO” VLAN to the “phones” VLAN, and reconfiguring the lines on your XiVO.
The instruction in this page are written for Debian jessie and VirtualBox.
On the host machine:
- 1 VLAN network interface for the XiVO. In our example, this will be
eth0.341
, with IP 10.34.1.254/24. - 1 VLAN network interface for the phones. In our example, this will be
eth0.342
, with IP 10.34.2.254/24.
On the guest machine, i.e. on the XiVO:
- 1 network adapter attached to the “XiVO” VLAN network interface. In our example, this interface inside the virtual machine will have the IP 10.34.1.1/24.
On the host, install the ISC DHCP server:
apt-get install isc-dhcp-server
If you do not want it to always be started:
systemctl disable isc-dhcp-server.service
Edit the DHCP server configuration file
/etc/dhcp/dhcpd.conf
. We need to configure the DHCP server to serve network configuration for the phones (Aastra and Snom in this case):ddns-update-style none; default-lease-time 3600; max-lease-time 86400; log-facility daemon; option space Aastra6700; option Aastra6700.cfg-server-name code 2 = text; option Aastra6700.contact-rcs code 3 = boolean; class "Aastra" { match if substring(option vendor-class-identifier, 0, 6) = "Aastra"; vendor-option-space Aastra6700; option Aastra6700.cfg-server-name = "http://10.34.1.1:8667/Aastra"; option Aastra6700.contact-rcs false; } class "Snom" { match if substring(option vendor-class-identifier, 0, 4) = "snom"; option tftp-server-name = "http://10.34.1.1:8667"; # the domain-name-servers option must be provided for the Snom 715 to work properly option domain-name-servers 10.34.1.1; } subnet 192.168.32.0 netmask 255.255.255.0 { } subnet 10.34.1.0 netmask 255.255.255.0 { } subnet 10.34.2.0 netmask 255.255.255.0 { authoritative; range 10.34.2.100 10.34.2.199; option subnet-mask 255.255.255.0; option broadcast-address 10.34.2.255; option routers 10.34.2.254; option ntp-servers 10.34.1.1; }
If you have many network interfaces on your host machine, you might also want to edit
/etc/default/isc-dhcp-server
to only include the “phones” VLAN network interface in the “INTERFACES” variable.Start the isc-dhcp-server:
systemctl start isc-dhcp-server.service
Add an iptables rules to do NAT:
iptables -t nat -A POSTROUTING -o eth0.341 -j MASQUERADE
Make sure that IP forwarding is enabled:
sysctl -w net.ipv4.ip_forward=1
Put all the phones in the “phones” VLAN on your switch
Activate the
NAT
andMonitoring
options on the page of your XiVO.
Note that the iptables rules and the IP forwarding setting are not persistent. If you don’t make them persistent (not documented here), don’t forget to reactivate them each time you want to recreate a NAT environment.
SCCP¶
xivo-libsccp is an alternative SCCP channel driver for Asterisk. It was originally based on chan_skinny.
This page is intended for developers and people interested in using xivo-libsccp on something other than XiVO.
Warning
If you just want to use your SCCP phones with XiVO, refer to SCCP Configuration instead.
The following packages are required to compile xivo-libsccp on Debian.
- build-essential
- asterisk-dev
apt-get update && apt-get install build-essential asterisk-dev
git clone https://github.com/xivo-pbx/xivo-libsccp.git
cd xivo-libsccp
make
make install
Warning
If you just want to use your SCCP phones with XiVO, refer to SCCP Configuration instead.
See sccp.conf.sample for a configuration file example.
Q. When is this *feature X* will be available?
A. The order in which we implement features is based on our client needs. Write
us an email that clearly explain your setup and what you would like to do and we
will see what we can do. We don't provide any timeline.
Q. I want to use the Page() application to call many phones at the same time.
A. Here a Page() example for a one way call (half-duplex):
exten => 1000,1,Verbose(2, Paging to external cisco phone)
same => n,Page(sccp/100/autoanswer&sccp/101/autoanswer,i,120 )
...for a two-way call (full-duplex):
exten => 1000,1,Verbose(2, Paging to external cisco phone)
same => n,Page(sccp/100/autoanswer&sccp/101/autoanswer,di,120 )
Here’s how to to configure a hostapd based AP on a Debian host so that both a 7920 and 7921 Wi-Fi phone can connect to it.
The 7920 is older than the 7921 and is pretty limited in its Wi-Fi functionnality:
- 802.11b
- WPA (no WPA2)
- TKIP (no CCMP/AES)
Which means that the most secure WLAN you can set up if you want both phones to connect to it is not that secure.
Make sure you have a wireless NIC capable of master mode.
If needed, install the firmware-<vendor> package. For example, if you have a ralink card like I do:
apt-get install firmware-ralink
Install the other dependencies:
apt-get install wireless-tools hostapd bridge-utils
Create an hostapd configuration file in
/etc/hostapd/hostapd.sccp.conf
with content:hostapd.sccp.conf
Update the following parameters (if applicable) in the configuration file:
- interface
- ssid
- channel
- wpa_passphrase
Create a new stanza in
/etc/network/interfaces
:iface wlan-sccp inet manual hostapd /etc/hostapd/hostapd.sccp.conf
Up the interface:
ifup wlan0=wlan-sccp
Configure your 7920/7921 to connect to the network.
To unlock the phone’s configuration menu on the 7921:
- Press the Navigation Button downwards to enter SETTINGS mode
- Navigate to and select Network Profiles
- Unlock the IP phone’s configuration menu by pressing **#. The padlock icon on the top-right of the screen will change from closed to open.
When asked for the authentication mode, select something like “Auto” or “AKM”.
You don’t have to enter anything for the username/password.
You’ll probably want to bridge your wlan0 interface with another interface, for example a VLAN interface:
brctl addbr br0 brctl addif br0 wlan0 brctl addif br0 eth0.341 ip link set br0 up
If you are using virtualbox and your guest interface is bridged to eth0.341, you’ll need to change its configuration and bridge it with br0 instead, else it won’t work properly.
This section describes the requirements to consider that a SCCP phone is working with XiVO libsccp.
- Register on Asterisk
- SCCP reset [restart]
- Call history
- Date time display
- HA
These test should be done with and without direct media enabled
- Emit a call
- Receive a call
- Receive and transfer a call
- Emit a call and transfer the call
- Hold and resume a call
- Features (*0 and others)
- Receive 2 calls simultaneously
- Emit 2 calls simultaneously
- DTMF on an external IVR
- Redial
- DND
- Hold
- Resume
- New call
- End call
- Call forward (Enable)
- Call forward (Disable)
- Try each button in each mode (on hook, in progress, etc)
- Phone book
- Caller ID and other display i18n
- MWI
- Speeddial/BLF
Web Interface¶
Default error level for XiVO web interface is E_ALL & ~E_DEPRECATED & ~E_USER_DEPRECATED & ~E_RECOVERABLE_ERROR & ~E_STRICT
If you want to display warning or other error in your browser, edit the /etc/xivo/web-interface/xivo.ini
and replace report_type level to 3:
[error]
level = E_ALL
report_type = 3
report_mode = 1
report_func = 1
email = john.doe@example.com
file = /var/log/xivo-web-interface/error.log
You may also edit /etc/xivo/web-interface/php.ini
and change the error level, but you will need to restart the cgi:
service spawn-fcgi restart
Instructions for Eclipse 4.5.
On your XiVO:
Install php5-xdebug:
apt-get install php5-xdebug
Edit the
/etc/php5/cgi/conf.d/20-xdebug.ini
(or/etc/php5/conf.d/20-xdebug.ini
on wheezy) and add these lines at the end:xdebug.remote_enable=1 xdebug.remote_host="<dev_host_ip>"
where
<dev_host_ip>
is the IP address of your machine where Eclipse is installed.Restart spawn-fcgi:
service spawn-fcgi restart
On your machine where Eclipse is installed:
Make sure you have Eclipse PDT installed
Create a PHP project named
xivo-web-interface
:- Choose “Create project at existing location”, using the
xivo-web-interface
directory
- Choose “Create project at existing location”, using the
In the Window / Preferences / PHP menu:
Add a new PHP server with the following information:
- Name: anything you want
- Base URL:
https://<xivo_ip>
- Path Mapping:
- Path on Server:
/usr/share/xivo-web-interface
- Path in Workspace:
/xivo-web-interface/src
- Path on Server:
Create a new
PHP Web Application
debug configuration:- Choose the PHP server you created in last step
- Pick some file, which can be anything if you don’t “break at first line”
- Uncheck “Auto Generate”, and set the path you want your browser to open when you’ll launch this debug configuration.
Then, to start a debugging session, set some breakpoints in the code and launch your debug configuration. This will open the page in your browser, and when the code will hit your breakpoints, you’ll be able to go through the code step by step, etc.
XiVO Client¶
This page explains how to build an executable of the XiVO Client from its sources for Windows.
You need the development files of the Qt 5 library, available on the Qt website. The currently supported Qt version is 5.5.0.
You will only need NSIS installed if you want to create an installer for the XiVO Client.
During the installer, choose the full installation.
The XiVO Client NSIS script file uses two plug-ins:
- the NSIS Application Association Registration Plug-in (download page)
- the NsProcess Plug-in (download page)
For each plug-in, download and extract the plug-in and place:
- the DLL from
/Plugins
in theNSIS/Plugins
directory - the
.nsh
from/Include
in theNSIS/Include
directory
In a Cygwin shell:
git clone git://github.com/xivo-pbx/xivo-client-qt.git
cd xivo-client-qt
touch xivoclient/qt-solutions/qtsingleapplication/src/{QtSingleApplication,QtLockedFile}
You must change the values in C:\Cygwin\home\user\xivo-client-qt\build-deps
to match
the paths of your installed programs. You must use an editor capable of understanding Unix end of
lines, such as Notepad++.
Replace C:\
with /cygdrive/c
and backslashes (\
) with slashes (/
). You must respect
the case of the directory names. Paths containing spaces must be enclosed in double quotes ("
).
For example, if you installed NSIS in C:\Program Files (x86)\nsis
, you should write:
WIN_NSIS_PATH="/cygdrive/c/Program files (x86)/nsis"
In a Cygwin shell:
source build-deps
export PATH=$WIN_QT_PATH/bin:$WIN_MINGW_PATH/bin:$PATH
qmake
mingw32-make SHELL=
Binaries are available in the bin
directory.
The version of the executable is taken from the git describe
command.
You can launch the built executable with:
source build_deps
PATH=$WIN_QT_PATH/bin:$PATH bin/xivoclient
To create the installer:
mingw32-make pack
This will result in a .exe
file in the current directory.
This page explains how to build an executable of the XiVO Client from its sources for GNU/Linux.
- Qt5 library development files: Qt website (Ubuntu packages
qt5-default qt5-qmake qttools5-dev-tools qttools5-dev libqt5svg5-dev
). The currently supported Qt version is 5.5.0. - openGL development library - libGL (Debian package
libgl1-mesa-dev
) - Git (Debian package
git
) - Generic software building tools :
make
,g++
... (Debian packagebuild-essential
)
You need to have the Qt5 binaries (qmake, lrelease, ...) in your $PATH.
Launch qmake to generate the Makefile:
$ cd xivo-client-qt
$ /path/to/qt5/bin/qmake
This will also generate a file versions.mak
that contains version informations about the code
being compiled. It is necessary for compilation and packaging.
You can then launch make
:
$ make
Binaries are available in the bin
directory.
The version of the executable is taken from the git describe
command.
To generate debug symbols:
$ make DEBUG=yes
To compile the unit tests of the XiVO Client:
$ qmake CONFIG+=tests
or, if you have a recent version of Google Mock:
$ qmake CONFIG+=tests CONFIG+=gmock
To compile the XiVO Client ready for functional tests:
$ make FUNCTESTS=yes
$ make distclean
To create the Debian package, usable on Debian and Ubuntu, you first need to modify
build-deps
to locate the Qt 5 installation directory:
$ /path/to/qt5/bin/qmake -spec linux-g++
$ make
$ make pack
This will result in a .deb
file in the current directory.
The version of the package is taken from the git describe
command.
This page explains how to build an executable of the XiVO Client from its sources for Mac OS.
You will need an Apple developer account to get development tools, such as GCC. To log in or sign in, go to the Developer portal of Apple. In the Downloads section, get the Command line Tools for XCode and install them. You might want to get XCode too, but it is rather big.
You need the development files of the Qt 5 library, available on the Qt website. The currently supported Qt version is 5.5.0.
Launch qmake to generate the Makefile:
$ cd xivo-client-qt
$ /path/to/qt5/bin/qmake -spec macx-g++
This will also generate a file versions.mak
that contains version informations about the code
being compiled. It is necessary for compilation and packaging.
You can then launch make
:
$ make
Binaries are available in the bin
directory.
The version of the executable is taken from the git describe
command.
$ make distclean
You can launch the built executable with:
$ DYLD_LIBRARY_PATH=bin bin/xivoclient.app/Contents/MacOS/xivoclient
You need to have the bin directory of Qt in your $PATH.
To create the app bundle:
$ make pack
This will result in a .dmg
file in the current directory.
The version of the package is taken from the git describe
command.
- Download
this patch
- git checkout xivo-client-1.1.23
- git apply xivoclient-1.1.23.patch
- Edit Makefile and set the variable QMAKE to the path of your qmake
- make all
- Edit cross/macos-pack.sh and set QT_PATH
- ./cross/macos-pack.sh
- Download
this patch
- Edit the patch and set the paths to Qt, NSIS, etc.
- (cygwin) git checkout xivo-client-1.0.15
- (cygwin) make all-win32
- (qt cmd) mingw32-make win32-baselib
- (qt cmd) mingw32-make win32-xivoclient
- (qt cmd) mingw32-make win32-plugins
- (cygwin) make win32packdyn-xivoclient
The folder baselib contains all files necessary to build the baselib. It contains the necessary code and data structures to communicate with the XiVO CTI server.
This library is designed to be reusable by other XiVO CTI clients. If you want to build it without the rest of the XiVO Client, go in its folder and type:
$ qmake && make
The library will be available in the new bin folder.
The folder xivoclient contains all other source files included in the XiVO Client.
src contains the source code files, images contains the images, i18n contains the translation files and qtaddons contains some Qt addons used by the XiVO Client.
The source files are separated in three categories :
- the XiVO Client itself, the source files are directly in src.
- the XLet library (xletlib) contains the code common to multiple XLets (plugins), like the XLet base class and mainly GUI stuff.
- the XLets themselves (xlets), each one is in a xlets/something subfolder.
Each XLet is compiled into a dynamic library, but some XLets are still compiled within the xivoclient executable instead of in a separated library. They are marked with a *-builtin subfolder name.
This folder contains all license informations necessary for the XiVO Client to be redistributed, i.e. the GNU GPLv3 and the additional requirements.
The settings of the application are stored in BaseEngine for runtime and in files when the client is closed :
- ~/.config/XiVO on GNU/Linux systems
- (what about other platforms?)
There are now 3 sets of functions from BaseEngine that you can use to read/store settings :
They are proxy methods to use the BaseConfig object inside BaseEngine. They use QVariantMap to store the settings values. They are currently used to store/retrieve options used in the ConfigWidget.
- You can find the available keys to access data in the detailed Doxygen documentation of BaseEngine,
- or in baseengine.h.
Note that the settings stored in BaseConfig won’t be written in the configuration file if BaseEngine is not aware of their existence (loaded in loadSettings and saved in saveSettings).
Through this function, you can access the lowest level of configuration storage, QSettings. It also contains the options stored in BaseConfig, but is less easy to use.
This direct access is used for purely graphical settings, only used to remember the appearance of the GUI until the next launch. These settings don’t have to be shared with other widgets, and storing them directly in QSettings avoids writing code to import/export to/from BaseConfig.
This pair of methods allow you to read/write settings directly in QSettings, but specifically for the current configuration profile.
When starting XiVO Client with an argument, this argument is interpreted as a profile name. This profile name allows you to separate different profiles, with different configuration options.
For example, configuration profile “profileA” will auto-connect with user A and password B and “profileB” will not auto-connect, but is set to connect with user C, no password remembered. To invoke these profiles, use :
$ xivoclient profileA
$ xivoclient profileB
The default configuration profile is default-user.
Of course, working on XiVO Client implies working with phone numbers. But how to interpret them easily, when we are not sure of the format they’re in?
You can use the PhoneNumber namespace (baselib/src/phonenumber.h) to do that, it contains routines for recognition/extraction of phone numbers, that way you don’t have to parse manually.
These subroutines are pretty basic for the moment, if you need/want to improve them, feel free to do it.
Informations are synchronized from the server to the BaseEngine when the client connects.
It is stored in BaseEngine in “lists”. It is stored in a format close to the one used to transmit it, so you can see the CTI protocol definition for further documentation.
Each list contains objects of different type. These types are :
- channel
- user
- phone
- trunk
- agent
- queue
- group
- meetme
- voicemail
- queuemember
- parking
Each type corresponds to a class derived from XInfo, e.g. channel infos are stored in ChannelInfo objects.
The basic attributes of all objects are 3 strings: the IPBX ID, the XiVO object ID and the extended ID of the object, which is the two previous attributes linked with a “/”.
If you want your XLet to receive IPBX/CTI events, you can do so by inheriting the IPBXListener interface.
You must specify which type of events you want to listen. This depends of the implemented functions in the CTI server. You can register to listen these events by calling the IPBXListener method :
registerListener(xxx);
For now, xxx, the event type, can take take the values : * chitchat * history * records_campaign * queuestats
On reception of the specified type of event, BaseEngine will call the IPBXListener method parseCommand(QVariantMap).
You should then reimplement this method to make it process the event data, stored in the QVariantMap parameter.
There are two concepts here : * Parked calls: These calls have been parked by a switchboard or an operator. They are waiting to be answered by a specific person, unlike a queue, where calls will be answered by one of the agents of the group associated to the queue. Each parked call is given a phone number so that the call can be answered by everyone.
- Parking lots: They are containers for parked calls. Each parking lot has a phone number, used to identify where to send the call we want to park.
ParkingWidget represents a parking lot and contains a table that stores all parked calls.
When you want to add a new XLet, you can use the basic XLetNull, that only prints “Hello World”. Here is a little script to accelerate the copy from XLetNull.
#!/usr/bin/env sh
newname="newname" # Replaces xletnull
NewName="NewName" # Replaces XLetNull & XletNull
NEWNAME="NEWNAME" # Replaces XLETNULL
if [ ! -d xletnull ] ; then
echo "Please execute this script in XIVO_CLIENT/plugins"
echo $newname
exit 1
fi
cp -r xletnull $newname
cd $newname
rm -f moc* *.o Makefile
for f in $(find . -type f -print) ; do
mv $f `echo $f | sed s/xletnull/$newname/`
done
find . -type f -exec sed -i "s/xletnull/$newname/g;s/X[Ll]etNull/$NewName/g;s/XLETNULL/$NEWNAME/g" {} \;</nowiki>
Before executing the script, just replace the first three variables with the name of the new XLet.
Then, you must add a line in xivoclient/xlets.pro to add your new directory to the SUBDIRS variable.
Then you can start implementing your new class. The <xletname>Plugin class is only an interface between the main app and your XLet.
If you want to localize your XLet, there are four steps.
In the <xletname>Plugin constructor, add the line :
b_engine->registerTranslation(”:/<xletname>_%1”);
before the return instruction.
Add these lines in the .pro file in your XLet directory :
TRANSLATIONS = <xletname>_fr.ts TRANSLATIONS += <xletname>_nl.ts
RESOURCES = res.qrc
Replace fr and nl with the languages you want.
In a file res.qrc in your XLet directory, put these lines :
<!DOCTYPE RCC><RCC version="1.0">
<qresource>
<file><xletname>_fr.qm</file>
<file><xletname>_nl.qm</file>
</qresource>
</RCC>
These files will be embedded in the Xlet library binary.
In your XLet directory, run :
lupdate <xletname>.pro
This creates as much .ts translation files as specified in the .pro file. You can now translate strings in these file.
The XLet will now be compiled and translated.
For now, it is not possible to add easily an XLet without changing the CTI server configuration files.
If you just want to test your new XLet, you can add the following line in baseengine.cpp :
m_capaxlets.push_back(QVariantList() << QVariant(“<xletname>”) << QVariant(“tab”));
right after the line
m_capaxlets = datamap.value(“capaxlets”).toList();
You can replace “tab” with “grid” or “dock”.
This is definitely not something funny and not easy to automatize.
You have to add, in every .pro file of the project (except xlets.pro and all those that don’t need translations), a line
TRANSLATIONS += <project>_<lang>.ts
Replace <project> with the project name (xivoclient, baselib, xlet) and <lang> by the identifier of your language (en, fr, nl, ...) Then you have to add, in every .qrc file, the .qm files corresponding to the ones you added in the .pro files, such as :
<file><project>_<lang>.qm</file>
in the <qresource> section of these XML .qrc files.
After that, you have to run, in the XiVO Client root directory, something like :
find . -name *.pro -exec lupdate {} ;
This will create or update all .ts translation files registered in the .pro files.
You can then start translating the strings in these files, in the xivoclient/i18n folder.
If you want to be able to select your new language from within the XiVO Client, you have to add it in the interface.
For that, you can add your new language in the m_locale_cbox QCombobox in ConfigWidget.
If you have a problem and you want to see what is going on between the CTI server and client, you can use a specific script, designed specifically for XiVO, instead of using something like Wireshark to listen network communications.
To get profiling informations on the XiVO Client:
Compile the XiVO Client with debugging symbols
Run the command:
LD_LIBRARY_PATH=bin valgrind --tool=callgrind bin/xivoclient
Quit the client
Open the generated file
callgrind.out.<pid>
with KCacheGrind
We use two tools to check the source code of the XiVO Client: CppCheck et Valgrind.
Usage:
LD_LIBRARY_PATH=bin valgrind --leak-check=full --suppressions=valgrind.supp --num-callers=30 --gen-suppressions=yes bin/xivoclient
You need to fill a file valgrind.supp
with Valgrind suppressions, to avoid displaying errors in code you have no control over.
Here is a template valgrind.supp
you can use. All memory in the XiVO Client is allocated using the new operator, so all calls to malloc
and co. must come from libraries:
{
malloc
Memcheck:Leak
fun:malloc
...
}
{
calloc
Memcheck:Leak
fun:calloc
...
}
{
realloc
Memcheck:Leak
fun:realloc
...
}
{
memalign
Memcheck:Leak
fun:memalign
...
}
Here’s a call graph for the presence features. Not complete, but gives a good global view of the internal mechanism.
Here’s a call graph describing the chaining of calls when the XiVO Client connects to the server.
This sections describes how to manage XiVO Client translations from a developer point of view. If you want to help translate the XiVO Client, see Translating XiVO
You need to install these tools:
pip install transifex-client
apt-get install qt4-dev-tools
String to be translated is marked using the tr macro in the source code.
Example:
tr("Number");
Run the following commands from the root of the xivo-client-qt project:
make pushtr
After this command, you can visit Transifex, and check that the xivo-client is 100% translated for your language. Once all the translations have been checked, run the 3 following commands:
make pulltr
git commit
git push
Warning
Under Arch Linux, you must have qt5 installed and prepend QT_PATH=/usr/bin
before
make {pull,push}tr
.
Localizing the XiVO Client goes through four steps :
- Creating the new translation in Transifex
- Generatint the translation files
- Embedding the translation in the binaries
- Displaying the new locale to be chosen
Log into Transifex and click the Create language
option.
The translation files will be automatically generated from the source code.
For the command to create files for your locale, you need to ensure it is listed in the project file.
There are a few project files you should edit, each one will translate a module of the XiVO Client :
baselib/baselib.pro
xivoclient/xivoclient.pro
xivoclient/xletlib.pro
xivoclient/src/xlets/*/*.pro
In these files, you should add a line like this one:
TRANSLATIONS += $$ROOT_DIR/i18n/xivoclient_fr.ts
This line adds a translation file for french. Please replace fr by the code of
your locale. The $$ROOT_DIR
variable references either xivoclient or
baselib.
You can use a command like the following to automate this ($LANG is the new language):
find . -name '*.pro' -exec sed -i -e 's|^TRANSLATIONS += $${\?ROOT_DIR}\?/i18n/\(.*\)_en.ts|\0\nTRANSLATIONS += $$ROOT_DIR/i18n/\1_$LANG.ts|' {} \;
To actually create the files, you will have to use the translation managing
script. But first, you must tell the script about your new locale. Edit the
utils/translations.sh
file and add your locale to the LOCALES
variable. Then, you can run the script:
$ make pulltr
For each project previously edited, you should have a corresponding .qrc
file. These resource files list all files that will be embedded in the XiVO
Client binaries. You should then add the corresponding translation files like
below:
<file>obj/xivoclient_fr.qm</file>
This embeds the French translation of the xivoclient
module, corresponding
to the translation file above. The path is changed to obj/
because the
.qm
file will be generated from the .ts
file.
You can use a command like the following to automate this ($LANG is the new language):
find . -name '*.qrc' -exec sed -i -e 's|^\( *\)<file>\(.*\)obj/\(.*\)_fr.qm</file>|\0\n\1<file>\2obj/\3_$LANG.qm</file>|' {} \;
You have to edit the source file xivoclient/src/configwidget.cpp
and add
the entry corresponding to your locale in the locale-choosing combobox.
Quality assurance¶
Testing architecture¶
Legend:
- assu is our production XiVO, used to make calls in the company. We also use it as a source of “external” calls to the test servers.
- dev-gateway is a simple gateway, to link all other servers.
- xivo-daily is reinstalled every day and runs all the automatic tests in xivo-acceptance.
- xivo-load handles a lot of calls all day long, and we monitor the system metrics while it does.
- callgen makes the calls towards xivo-load
- xivo-test and xivo-test-slave are used for manual tests we run before each release
- xivo-premium (not yet installed) will allow us to test the new xivo-premium hardware
Troubleshooting¶
The list of current bugs can be found on the official XiVO issue tracker.
Transfers using DTMF¶
When transfering a call using DTMF (*1) you get an invalid extension error when dialing the extension.
The workaround to this problem is to create a preprocess subroutine and assign it to the destinations where you have the problem.
Under
add a new file containing the following dialplan:[allow-transfer]
exten = s,1,NoOp(## Setting transfer context ##)
same = n,Set(__TRANSFER_CONTEXT=<internal-context>)
same = n,Return()
Do not forget to substitute <internal-context> with your internal context.
Some places where you might want to add this preprocess subroutine is on queues and outgoing calls to be able to transfer the called person to another extension.
Fax detection¶
XiVO does not currently support Fax detection. The following describe a workaround to use this feature. The behavior is to answer all incoming (external) call, wait for a number of seconds (4 in this example) : if a fax is detected, receive it otherwise route the call normally.
Note
This workaround works only :
- on incoming calls towards an User (and an User only),
- if the incoming trunk is a DAHDI or a SIP trunk,
- if the user has a voicemail which is activated and with the email field filled
- XiVO >= 13.08 (needs asterisk 11)
Be aware that this workaround will probably not survive any upgrade.
In the Web Interface and under
add a new file named fax-detection.conf containing the following dialplan:;; Fax Detection [pre-user-global-faxdetection] exten = s,1,NoOp(Answer call to be able to detect fax if call is external AND user has an email configured) same = n,GotoIf($["${XIVO_CALLORIGIN}" = "extern"]?:return) same = n,GotoIf(${XIVO_USEREMAIL}?:return) same = n,Set(FAXOPT(faxdetect)=yes) ; Activate dynamically fax detection same = n,Answer() same = n,Wait(4) ; You can change the number of seconds it will wait for fax (4 to 6 is good) same = n,Set(FAXOPT(faxdetect)=no) ; If no fax was detected deactivate dyamically fax detection (needed if you want directmedia to work) same = n(return),Return() exten = fax,1,NoOp(Fax detected from ${CALLERID(num)} towards ${XIVO_DSTNUM} - will be sent upon reception to ${XIVO_USEREMAIL}) same = n,GotoIf($["${CHANNEL(channeltype)}" = "DAHDI"]?changeechocan:continue) same = n(changeechocan),Set(CHANNEL(echocan_mode)=fax) ; if chan type is dahdi set echo canceller in fax mode same = n(continue),Gosub(faxtomail,s,1(${XIVO_USEREMAIL}))
In the file
/etc/xivo/asterisk/xivo_globals.conf
set the global user subroutine topre-user-global-faxdetection
: this subroutine will be executed each time a user is called:XIVO_PRESUBR_GLOBAL_USER = pre-user-global-faxdetection
Reload asterisk configuration (both for dialplan and dahdi):
asterisk -rx 'core reload'
Berofos Integration with PBX¶
You can use a Berofos failover switch to secure the ISDN provider lines when installing a XiVO in front of an existing PBX. The goal of this configuration is to mitigate the consequences of an outage of the XiVO : with this equipment the ISDN provider links could be switched to the PBX directly if the XiVO goes down.
XiVO does not offer natively the possibility to configure Berofos in this failover mode. This section describes a workaround.
Logical view:
+------+ +-----+
-- Provider ----| XiVO | -- ISDN Interconnection --| PBX | -- Phones
+------+ +-----+
Connection:
+-------------Bero*fos---------------+
| A B C D |
| o o o o o o o o o o o o o o o o |
+-+-+------+-+------+-+------+-+-----+
| | | | | | | |
/ / | | | | | |
/ / +--------+ / / +---------+
2 T2 | XiVO | / / | PBX |
+--------+ / / +---------+
| | / /
\ \__/ /
\____/
The following describes how to configure your XiVO and your Berofos.
Follow the Berofos general configuration (firmware, IP, login/password) described in the the Berofos Installation and Configuration page.
When done, apply these specific parameters to the berofos:
bnfos --set scenario=1 -h 10.105.2.26 -u admin:berofos bnfos --set mode=1 -h 10.105.2.26 -u admin:berofos bnfos --set modedef=1 -h 10.105.2.26 -u admin:berofos bnfos --set wdog=1 -h 10.105.2.26 -u admin:berofos bnfos --set wdogdef=1 -h 10.105.2.26 -u admin:berofos bnfos --set wdogitime=60 -h 10.105.2.26 -u admin:berofos
Add the following script
/usr/local/sbin/berofos-workaround
:#!/bin/bash # Script workaround for berofos integration with a XiVO in front of PABX res=$(/usr/sbin/service asterisk status) does_ast_run=$? if [ $does_ast_run -eq 0 ]; then /usr/bin/logger "$0 - Asterisk is running" # If asterisk is running, we (re)enable wdog and (re)set the mode /usr/bin/bnfos --set mode=1 -f fos1 -s /usr/bin/bnfos --set modedef=1 -f fos1 -s /usr/bin/bnfos --set wdog=1 -f fos1 -s # Now 'kick' berofos ten times each 5 seconds for ((i == 1; i <= 10; i += 1)); do /usr/bin/bnfos --kick -f fos1 -s /bin/sleep 5 done else /usr/bin/logger "$0 - Asterisk is not running" fi
Add execution rights to script:
chmod +x /usr/local/sbin/berofos-workaround
Create a cron to launch the script every minutes
/etc/cron.d/berofos-cron-workaround
:# Workaround to berofos integration MAILTO="" */1 * * * * root /usr/local/sbin/berofos-workaround
Upgrading from XiVO 1.2.3¶
There is an issue with
xivo-libsccp
andpf-xivo-base-config
during an upgrade from 1.2.3:dpkg: error processing /var/cache/apt/archives/pf-xivo-base-config_13%3a1.2.4-1_all.deb (--unpack): trying to overwrite '/etc/asterisk/sccp.conf', which is also in package xivo-libsccp 1.2.3.1-1 ... Errors were encountered while processing: /var/cache/apt/archives/pf-xivo-base-config_13%3a1.2.4-1_all.deb E: Sub-process /usr/bin/dpkg returned an error code (1)
You have to remove
/var/lib/dpkg/info/xivo-libsccp.conffiles
:rm /var/lib/dpkg/info/xivo-libsccp.conffiles
You have to edit
/var/lib/dpkg/info/xivo-libsccp.list
and remove the following line:/etc/asterisk/sccp.conf
and remove
/etc/asterisk/sccp.conf
:rm /etc/asterisk/sccp.conf
Now, you can launch
xivo-upgrade
to finish the upgrade process
CTI server is unexpectedly terminating¶
If you observes that your CTI server is sometimes unexpectedly terminating with the following
message in /var/log/xivo-ctid.log
:
(WARNING) (main): AMI: CLOSING
Then you might be in the case where asterisk generates lots of data in a short period of time on the AMI while the CTI server is busy processing other thing and is not actively reading from its AMI connection. If the CTI server takes too much time before consuming some data from the AMI connection, asterisk will close the AMI connection. The CTI server will terminate itself once it detects the connection to the AMI has been lost.
There’s a workaround to this problem called the ami-proxy, which is a process which buffers the AMI connection between the CTI server and asterisk. This should only be used as a last resort solution, since this increases the latency between the processes and does not fix the root issue.
To enable the ami-proxy, you must:
Add a file
/etc/systemd/system/xivo-ctid.service.d/ami-proxy.conf
:mkdir -p /etc/systemd/system/xivo-ctid.service.d cat >/etc/systemd/system/xivo-ctid.service.d/ami-proxy.conf <<EOF [Service] Environment=XIVO_CTID_AMI_PROXY=1 EOF systemctl daemon-reload
Restart the CTI server:
systemctl restart xivo-ctid.service
If you are on a XiVO cluster, you must do the same procedure on the slave if you want the ami-proxy to also be enabled on the slave.
To disable the ami-proxy:
rm /etc/systemd/system/xivo-ctid.service.d/ami-proxy.conf
systemctl daemon-reload
systemctl restart xivo-ctid.service
Agents receiving two ACD calls¶
An agent can sometimes receive more than 1 ACD call at the same time, even if the queues he’s in have the “ringinuse” parameter set to no (default).
This behaviour is caused by a bug in asterisk: https://issues.asterisk.org/jira/browse/ASTERISK-16115
It’s possible to workaround this bug in XiVO by adding an agent subroutine. The subroutine can be either set globally or per agent:
[pre-limit-agentcallback]
exten = s,1,NoOp()
same = n,Set(LOCKED=${LOCK(agentcallback-${XIVO_AGENT_ID})})
same = n,GotoIf(${LOCKED}?:not-locked,1)
same = n,Set(GROUP(agentcallback)=${XIVO_AGENT_ID})
same = n,Set(COUNT=${GROUP_COUNT(${XIVO_AGENT_ID}@agentcallback)})
same = n,NoOp(${UNLOCK(agentcallback-${XIVO_AGENT_ID})})
same = n,GotoIf($[ ${COUNT} <= 1 ]?:too-many-calls,1)
same = n,Return()
exten = not-locked,1,NoOp()
same = n,Log(ERROR,Could not obtain lock)
same = n,Wait(0.5)
same = n,Hangup()
exten = too-many-calls,1,NoOp()
same = n,Log(WARNING,Not calling agent ID/${XIVO_AGENT_ID} because already in use)
same = n,Wait(0.5)
same = n,Hangup()
This workaround only applies to queues with agent members; it won’t work for queues with user members.
Also, the subroutine prevent asterisk from calling an agent twice by hanguping the second call. In the agent statistics, this will be shown as a non-answered call by the agent.
PostgreSQL localization errors¶
The database and the underlying database cluster used by XiVO is sensitive to the system locale configuration. The locale used by the database and the database cluster is set when XiVO is installed. If you change your system locale without particular attention to PostgreSQL, you might make the database and database cluster temporarily unusable.
When working with locale and PostgreSQL, there’s a few useful commands and things to know:
locale -a
to see the list of currently available locales on your systemlocale
to display information about the current locale of your shellgrep ^lc_ /etc/postgresql/9.4/main/postgresql.conf
to see the locale configuration of your database clustersudo -u postgres psql -l
to see the locale of your databases- the
/etc/locale.gen
file and the associatedlocale-gen
command to configure the available system locales systemctl restart postgresql.service
to restart your database cluster- the PostgreSQL log file located at
/var/log/postgresql/postgresql-9.4-main.log
Note
You can use any locale with XiVO as long as it uses an UTF-8 encoding.
If the database cluster doesn’t start and you have the following errors in your log file:
LOG: invalid value for parameter "lc_messages": "en_US.UTF-8"
LOG: invalid value for parameter "lc_monetary": "en_US.UTF-8"
LOG: invalid value for parameter "lc_numeric": "en_US.UTF-8"
LOG: invalid value for parameter "lc_time": "en_US.UTF-8"
FATAL: configuration file "/etc/postgresql/9.4/main/postgresql.conf" contains errors
Then this usually means that the locale that is configured in postgresql.conf
(here en_US.UTF-8
)
is not currently available on your system, i.e. does not show up the output of locale -a
. You
have two choices to fix this issue:
- either make the locale available by uncommenting it in the
/etc/locale.gen
file and runninglocale-gen
- or modify the
/etc/postgresql/9.4/main/postgresql.conf
file to set the variouslc_*
options to a locale that is available on your system
Once this is done, restart your database cluster.
If the database cluster is up but you get the following error when trying to connect to the
asterisk
database:
FATAL: database locale is incompatible with operating system
DETAIL: The database was initialized with LC_COLLATE "en_US.UTF-8", which is not recognized by setlocale().
HINT: Recreate the database with another locale or install the missing locale.
Then this usually means that the database locale is not currently available on your system. You have two choices to fix this issue:
- either make the locale available by uncommenting it in the
/etc/locale.gen
file, runninglocale-gen
and restarting your database cluster - or recreate the database using a different locale
Then you are mostly in one of the cases described above. Check your log file.
If during a database restore, you get the following error:
pg_restore: [archiver (db)] Error while PROCESSING TOC:
pg_restore: [archiver (db)] Error from TOC entry 4203; 1262 24745 DATABASE asterisk asterisk
pg_restore: [archiver (db)] could not execute query: ERROR: invalid locale name: "en_US.UTF-8"
Command was: CREATE DATABASE asterisk WITH TEMPLATE = template0 ENCODING = 'UTF8' LC_COLLATE = 'en_US.UTF-8' LC_CTYPE = 'en_US.UTF-8';
Then this usually means that your database backup has a locale that is not currently available on your system. You have two choices to fix this issue:
either make the locale available by uncommenting it in the
/etc/locale.gen
file, runninglocale-gen
and restarting your database clusteror if you want to restore your backup using a different locale (for example
fr_FR.UTF-8
), then restore your backup using the following commands instead:sudo -u postgres dropdb asterisk sudo -u postgres createdb -l fr_FR.UTF-8 -O asterisk -T template0 asterisk sudo -u postgres pg_restore -d asterisk asterisk-*.dump
Then the slave database is most likely not using an UTF-8 encoding. You’ll need to recreate the database using a different locale
If you have decided to change the locale of your database, you must:
- make sure that you have enough space on your hard drive, more precisely in the file system holding
the
/var/lib/postgresql
directory. You’ll have, for a moment, two copies of theasterisk
database. - prepare for a service interruption. The procedure requires the services to be restarted twice, and the system performance will be degraded while the database with the new locale is being created, which can take a few hours if you have a really large database.
- make sure the new locale is available on your system, i.e. shows up in the output of
locale -a
Then use the following commands (replacing fr_FR.UTF-8
by your locale):
xivo-service restart all
sudo -u postgres createdb -l fr_FR.UTF-8 -O asterisk -T template0 asterisk_newlocale
sudo -u postgres pg_dump asterisk | sudo -u postgres psql -d asterisk_newlocale
xivo-service stop
sudo -u postgres psql <<'EOF'
DROP DATABASE asterisk;
ALTER DATABASE asterisk_newlocale RENAME TO asterisk;
EOF
xivo-service start
You should also modify the /etc/postgresql/9.4/main/postgresql.conf
file to set the various
lc_*
options to the new locale value.
For more information, consult the official documentation on PostgreSQL localization support.
Originate a call from the Asterisk console¶
It is sometimes useful to ring a phone from the asterisk console. For example, if you want
to call the 1234
extension in context default
:
channel originate Local/1234@default extension 42@xivo-callme
WebRTC¶
- http.conf - asterisk’s webserver must accept connection from outside, the listen address must be updated, for the sake of simplicity let’s use 0.0.0.0, you can also pick an address of one of the network interfaces:
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=5039
prefix=
tlsenable=yes
tlsbindaddr=127.0.0.1:5040
tlscertfile=/usr/share/xivo-certs/server.crt
tlsprivatekey=/usr/share/xivo-certs/server.key
servername=XiVO PBX
Do not forget to reload the configuration by the module reload http command on the Asterisk CLI.
- rtp.conf - the ICE support must be activated:
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
icesupport=yes
stunaddr=stun.l.google.com:19302
The configuration is reloaded by module reload res_rtp_asterisk.so.
- WebRTC requires DTLS keys to be generated in /etc/asterisk/keys. If you need to manually generate the DTLS
certificates following instructions on the Asterisk Wiki: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial.
You just need to generate the TLS certificates (first call of ast_tls_cert), other steps are not necessary.
Make sure asterisk can read files by executing:
chown -R asterisk.asterisk /etc/asterisk/keys
Community Documentation¶
This page provides links to resources on various topics around XiVO. They have been generously created by people from the community.
Tutorials¶
Please note that these resources are provided on an “as is basis”. They have not been reviewed by the XiVO team, therefore the information presented may be innaccurate. We also accept resources provided in other languages besides English.
Unless specified, the license is CC BY-SA.
Contribute¶
We gladly accept new contributions. There are two ways to contribute:
- The preferred way: open a pull request on Github and add a line to this page (see: Contributing to the Documentation).
- You can also open a contribution ticket on the bug tracker.
Note that we only accept documents in open formats, such as PDF or ODF.
Indices and tables¶
XiVO-CC Documentation¶
XiVO-CC is an application suite developed by Avencall Group, and provides enhancements of the XiVO PBX contact center functionalities.
Table of Contents¶
Introduction¶
Xivo-CC provides enhancements of the XiVO PBX contact center functionalities. It gives especially acces to outsourced statistics, real-time supervision screens, third-party CTI integration and recording facilities.
Installation and system configuration¶
The XiVO-CC software suite is made of several independent components. Depending on your system size, they can be installed on separate virtual or physical machines. In this section, we will explain how to install these components on a single machine.
Installation¶
This page describes how to install the XiVO CC.
It describes the installation with the debian package of the whole XiVO CC.
Note
As a reference, the manual installation page is here Manual configuration and installation.
Warning
- the wizard MUST be passed on the XiVO PBX
- XiVO PBX will be reconfigured during the installation and must be restarted. You may accept the automatic restart during the installation or you need to restart it manually later before starting the docker containers.
- If you configure HA on XiVO, you have to reconfigure postgres for CC ...
The following components will be installed :
- XuC : outsourced CTI server providing telephony events, statistics and commands through a WebSocket
- XuC Management : supervision web pages based on the XuC
- Pack Reporting : statistic summaries stored in a PostgreSQL database
- Totem Support : near-real time statistics based on ElasticSearch
- SpagoBI : BI suite with default statistic reports based on the Pack Reporting
- Recording Server : web server allowing to search recorded conversations
- Xuc Rights Management : permission provider used by XuC and Recording Server to manage the user rights
We will assume your XiVO CC server meets the following requirements:
- OS : Debian 8 (jessie), 64 bits.
- you have a XiVO PBX installed in a compatible version (basically the two components XiVO and XiVO CC have to be in the same version).
- the XiVO PBX is reachable on the network.
- the XiVO PBX is setup (wizard must be passed) with users, queues and agents, you must be able to place and answer calls.
For the rest of this page, we will make the following assumptions :
- the XiVO PBX has the IP 192.168.0.1
- some data (incoming calls, internal calls etc.) might be available on XiVO (otherwise, you will not see anything in the check-list below).
- the XiVO CC server has the IP 192.168.0.2
- the package xivo-recording is available on a custom Debian mirror. If this is not the case, you will need to skip the apt-get install commands and build the packages yourself.
XiVO PBX enables a wide range of configuration, XiVO-CC is tested and validated with a number of restriction concerning configurations of XiVO PBX:
- Do not activate Contexts Separation in xivo-ctid Configuration
- Users deactivation is not supported
- Queue ringing strategy should not be Ring All
- Do not use pause on one queue status, in queue advanced configuration, autopause should be No or All
- Do not activate Call a member already on (Asterisk ringinuse) on xivo queue advanced configuration
- All users and queues have to be in the same context
- Agent and Supervisors profiles should use the same Presence Group
- Agents and Phones should be in the same context for mobile agents
- Agents must not have a password in XiVO agent configuration page
The installation and configuration of XiVO CC (with its XiVO PBX part) is handled by the xivocc-installer package which is available in the repository.
The install process consists of three parts:
- The first part is to manually run the
install-docker.sh
script to install docker and docker compose. - The second part is the installation of XiVO CC itself.
- The third part is to install the extra package for the recording.
The installation is automatic and you will be asked few questions during the process:
- When asked to generate a pair of authentication keys, leave the password field empty.
- Before copying the authentication keys, you will be prompted for the XiVO PBX root password.
- Enter IP addresses of XiVO PBX and XiVO CC.
- XiVO PBX must restart, the question will prompt you to restart during the process or to restart later.
Note
To be run on the XiVO CC server
On a fresh debian install you will probably need to install the ca-certificates
package:
apt-get install ca-certificates
Now you can download the script which will install docker and docker compose.
First, download the install-docker.sh script (in the following URL, replace CURRENT_VERSION with the current version, e.g. 2016.04):
wget https://gitlab.com/xivoxc/packaging/raw/CURRENT_VERSION/install/install-docker.sh -O install-docker.sh
And execute the script:
chmod +x install-docker.sh ./install-docker.sh
Note
To be run on the XiVO CC server
The XiVO CC server and the XiVO PBX server must be synchronized to the same NTP source.
apt-get install ntp
Recomended configuration : you should configure the NTP server of the XiVO CC server towards the XiVO PBX. In our example it means to add the following line in the file /etc/ntp.conf:
server 192.168.0.1 iburst
Note
To be run on the XiVO CC server
This step will install the XiVO CC components via the xivocc-installer
package. It is required to restart XiVO PBX during or after the setup process.
The installer will ask whether you wish to restart XiVO PBX later.
Warning
- This package must be installed on the XivoCC server.
- Wizard MUST be passed on the XiVO PBX.
- XiVO PBX services will need to be restarted. The installer will ask whether you wish to restart XiVO PBX during or after the setup process.
Also, check that you have following information:
- XiVO PBX root password;
- OpenSSH
PermitRootLogin
set toyes
(you could revert tono
after installation of XivoCC);- XiVO PBX‘s IP address;
- XiVO CC IP address (the one visible by XiVO PBX);
- Number of weeks to keep statistics;
- Number of weeks to keep recordings (beware of space disk);
Install the xivocc-installer package via apt:
Add the sources list with the key (in the following URL, replace CURRENT_VERSION with the current version, e.g. 2016.04):
echo "deb http://mirror.xivo.solutions/archive/ xivo-solutions-CURRENT_VERSION main" > /etc/apt/sources.list.d/xivo-solutions.list wget http://mirror.xivo.solutions/xivo_current.key -O - | apt-key add -
Update your source list and install the package:
apt-get update apt-get install xivocc-installer
Note
To be run on the XiVO PBX server
To be able to install the package you must have the XiVO Solutions repository on your XiVO PBX. Then install on the XiVO PBX the debian package available in the repository.
apt-get install xivo-recording
During the installation, you will be asked for :
- the recording server IP (192.168.0.2)
- the XiVO name (it must not contain any space or “-” character).
If you have more than one XiVO, you must give a different name to each of them.
This package installs two dialplan subroutines :
- xivo-incall-recording : used to record incoming calls
- xivo-outcall-recording : used to record outgoing calls
To use the subroutines, you must edit the configuration file /etc/xivo/asterisk/xivo_globals.conf
and assign them to chosen preprocess subroutines. E.g. :
XIVO_PRESUBR_FWD_QUEUE = xivo-incall-recording
...
XIVO_PRESUBR_GLOBAL_QUEUE = xivo-incall-recording
If you want to record on a gateway used with Xivo, you must not use the xivo-recording package but gateway-recording.
If you want to use call recording filtering, please install also:
apt-get install call-recording-filtering
During the installation, you will be asked for :
- the recording server address with protocol and port (http://192.168.0.2:9400)
Using the XivoCC configuration manager : http://192.168.0.2:9100/ add user xuc as administrator to be able to get call history in web assistant.
After the successful installation, start docker containers by an alias which was added to ~/.bashrc
source ~/.bashrc
dcomp up -d
If you selected to restart XiVO PBX later, please do so when possible to apply the modifications made by the installer. The XiVO CC server will not be able to connect correctly to the database on XiVO PBX.
To restart XiVO services, on XiVO PBX server run
xivo-service restart all
To reinstall the package, it is required to apt-get purge xivocc-installer followed by apt-get install xivocc-installer. This will re-run the configuration of the package, download the docker compose template and setup XiVO PBX.
Purging the package will also remove the xuc and stats users from the XiVO PBX database.
- To avoid problems when uninstalling, you should:
- to uninstall, please use
apt-get purge xivocc-installer
- if the process is aborted, it will break the installation, please
apt-get purge
andapt-get install
again
- to uninstall, please use
Compoment version can be find in the log files, on the web pages for web components. You may also get the version from the docker container itself by typing :
docker exec -ti xivocc_xucmgt_1 cat /opt/docker/conf/appli.version
Change xivocc_xucmgt_1 by the component version you want to check
The various applications are available on the following addresses:
- Xuc-related applications: http://192.168.0.2:8070/
- SpagoBI: http://192.168.0.2:9500/
- Config Management: http://192.168.0.2:9100/
- Recording server: http://192.168.0.2:9400/
- Kibana: http://192.168.0.2/
- Using the configuration manager : http://192.168.0.2:9100/ (default user avencall/superpass) add a user to be able to use the recording interface with proper rights.
Note
Xuc server default user is xuc, add xuc as administrator to be able to get call history in web assistant.
Warning
If you change the cti login username in xivo configuration, user has to be recreated with apropriate rights in configuration manager.
- Go to http://192.168.0.2:9500/SpagoBI (by default login: biadmin, password: biadmin)
- Update default language : go to “⚙ Resources” > “Configuration management” > in the “Select Category” field, chose “LANGUAGE_SUPPORTED” and change value of the label “SPAGOBI.LANGUAGE_SUPPORTED.LANGUAGE.default” in your language : fr,FR , en,US , ...
- Download the standard reports from https://gitlab.com/xivocc/sample_reports/raw/master/spagobi/standardreports.zip
- Import zip file in SpagoBI: ” Repository Management” > Click on “Browse” and choose the previous downloaded zip file > Click on “Import” (All default options, with Jasper Report Engine as Engine associations).
Use the database status report to check if replication and reporting generation is working :
XivoCC agent can make outgoing calls through an outgoing queue. This brings the statistics and supervision visualization for outgoing ACD calls. However, some special configuration steps are required:
- You need to create an outgoing queue with a name starting with ‘out’, e.g. outgoing_queue.
- This queue must be configured with preprocess subroutine xuc_outcall_acd, without On-Hold Music (tab General), Ringing Time must be 0 and Ring instead of On-Hold Music must be activated (both tab Application).
- The subroutine must be deployed on the Xivo server (to /etc/asterisk/extension_extra.d/ or through the web interface), the file is available from https://gitlab.com/xivoxc/xucserver/raw/master/xivo/outbound/xuc_outcall_acd.conf, with owner asterisk:www-data and rights 660.
- You must also deploy the file https://gitlab.com/xivoxc/xucserver/raw/master/xivo/outbound/generate_outcall_skills.py to /usr/local/sbin/, with owner root:root and rights 755.
- Furthermore, you must replace the file /etc/asterisk/queueskills.conf by the following one https://gitlab.com/xivoxc/xucserver/raw/master/xivo/outbound/queueskills.conf (be sure to backup the original one), without changing the owner or rights
- And finally you need to add a new skill rule on the Xivo server: Services -> Call center -> Skill rules -> Add, with name ‘select_agent’ and rules ‘$agent > 0’.
Once done, calls requested by an agent through the Cti.js with more than 6 digits are routed via the outgoing queue. You can change the number of digits using the parameter xuc.outboundLength in the xuc’s configuration.
Data replication can take some time if there are a lot of data in xivo cel and queue log tables. You may check xivo-db-replication log files (/var/log/xivocc/xivo-db-replication.log).
Preconfigured panels are available on http://@IP/kibana/#/dashboard/file/queues.json et http://@IP/kibana/#/dashboard/file/agents.json to be able to save this panels in elasticsearch database you have to sign on on request user admin/Kibana
- All components are running : dcomp ps
- Xuc internal database is synchronized with xivo check status page with http://xivoccserver:8090/
- CCManager is running, log a user and check if you can see and manage queues : http://xivoccserver:8070/ccmanager
- Web agent is running, log an agent and check if you can change the status : http://xivoccserver:8070/agent
- Web assistant is running, and you get call history : http://xivoccserver:8070/
- Check database replication status using spagobi system report
- Check elasticsearch database status (totem panels) http://xivoccserver:9200/queuelogs/_status
- Check that you can listen to recordings http://xivoccserver:9400/
- Check totem panels http://192.168.85.102/kibana
###### reminder: Make sure to have few calls made in your XiVO, despite you will not see anything in totem or spagobi.
Configure LDAP authent for CCmanager, Web Assistant and Web Agent
You need to include in the compose.yml file a link to a specific configuration file by adding in xuc section a specific volume and an environment variable to specify the alternate config file location
xuc:
....
environment:
....
- CONFIG_FILE=/conf/xuc.conf
volumes:
- /etc/docker/xuc:/conf
Edit in /etc/docker/xuc/ a configuration file named xuc.conf to add ldap configuration (empty by default)
include "application.conf"
authentication {
ldap {
managerDN = "uid=company,ou=people,dc=company,dc=com" # user with read rights on the whole LDAP
managerPassword = "xxxxxxxxxx" # password for this user
url = "ldap://ldap.company.com:389" # ldap URI
searchBase = "ou=people,dc=company,dc=com" # ldap entry to use as search base
userSearchFilter = "uid=%s" # filter to use to search users by login, using a string pattern
}
}
Recreate the container : dcomp up -d xuc
This section describes the manual installation of the XiVO CC components. In most cases you SHOULD NOT follow this page, and install the XiVO CC components via the xivocc-installer package (see Installation).
Note
We leave this page here :
- as a reference,
- and when one must install only a subset of the XiVO CC components (since it is not currently possible via the xivocc-installer package).
We will assume your XiVO CC server meets the following requirements:
- OS : Debian 8 (jessie), 64 bit
- the latest stable version of Docker is installed
- the latest stable version of Docker-compose is installed
- the XiVO PBX is reachable on the network
- the XiVO PBX is setup with users, queues and agents, you must be able to place and answer calls.
Note : Install only stable version of docker and docker compose.
We will make the following assumptions :
- the XiVO PBX has the IP 192.168.0.1
- some data (incoming calls, internal calls etc.) might be available on XiVO (otherwise, you will not see anything in the Post Installation Check List).
- the XiVO CC server has the IP 192.168.0.2
- the package xivo-recording is available on a custom Debian mirror. If this is not the case, you will need to skip the apt-get install commands and build the packages yourself.
Firstly, allow access to PostgreSQL from the outside. Edit /etc/postgresql/9.4/main/postgresql.conf
:
listen_addresses = '*'
Add this line to /etc/postgresql/9.4/main/pg_hba.conf
:
host asterisk all 192.168.0.2/32 md5
Create a user stats with read permissions :
sudo -u postgres psql asterisk << EOF
CREATE USER stats WITH PASSWORD 'stats';
GRANT SELECT ON ALL TABLES IN SCHEMA PUBLIC TO stats;
EOF
And run xivo-service restart all to apply these modifications.
- (XiVO PBX >= 15.18) Add a a file xuc.conf in
/etc/asterisk/manager.d
directory with :
[xuc]
secret = xucpass
deny=0.0.0.0/0.0.0.0
permit=X.X.X.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,dtmf,originate,dialplan
write = system,call,log,verbose,command,agent,user,dtmf,originate,dialplan
Replace X.X.X.0 by your xivocc network
And reload the AMI :
asterisk -rx "manager reload"
asterisk -rx "manager show user xuc" and check your if previous configuration is displayed.
Add some events in the CEL. Edit /etc/asterisk/cel.conf
:
- For Asterisk 13:
[general]
enable = yes
apps = dial,park,queue
events = APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT,USER_DEFINED,LINKEDID_END,HOLD,UNHOLD,BLINDTRANSFER,ATTENDEDTRANSFER
[manager]
enabled = yes
and reload the cel module in Asterisk :
asterisk -rx "module reload cel"
Create a user Xuc in
with the following parameters:- CTI login : xuc
- CTI password : 0000
- profil supervisor
Create a Web Services user in
with the following parameters :- Login : xivows
- Password : xivows
- Host : 192.168.0.2
Make sure Multiqueues call stats sharing is enabled in
tab.Do not forget to follow configuration steps detailed in Required configuration for phone integration.
Still on the xivo, install the package which will handle the recording :
apt-get update
apt-get install xivo-recording
During the installation, you will be asked for :
- the recording server IP (i.e. 192.168.0.2)
- and the XiVO name (it must not contain any space or “-” character).
If you have several XiVO, you must give a different name to each of them.
This package has installed two dialplan sub-routines :
- xivo-incall-recording : used to record incoming calls
- xivo-outcall-recording : used to record outgoing calls
You have to manually place them where you want.
If you want to record on a gateway used with Xivo, you must not use the xivo-recording package but gateway-recording.
If you want to use call recording filtering, please install also:
apt-get install call-recording-filtering
During the installation, you will be asked for :
- the recording server address with protocol and port (i.e. http://192.168.0.2:9400)
Now we switch to the installation of the XiVO CC server.
apt-get install ntp
XiVO CC server and XiVO PBX server must be synchronized to the same source.
Docker container log output to /dev/stdout and /dev/stderr. The Docker container log file is saved in /var/lib/docker/containers/[CONTAINER ID]/[CONTAINER_ID]-json.log.
Create a new Logrotate config file for your Docker containers in the Logrotate folder /etc/logrotate.d/docker-container.
/var/lib/docker/containers/*/*.log {
rotate 7
daily
compress
missingok
delaycompress
copytruncate
}
You can test it with logrotate -fv /etc/logrotate.d/docker-container. You should get some output and a new log file with suffix [CONTAINER ID]-json.log.1 should be created. This file is compressed in next rotation cycle.
Retrieve the configuration script and launch it:
wget https://gitlab.com/xivoxc/packaging/raw/master/install/install-docker-xivocc.sh
bash install-docker-xivocc.sh
During the installation, you will be asked for :
- the XiVO IP address (e.g. 192.168.0.1)
- the number of weeks to keep for the statistics
- the number of weeks to keep for the recording files
- the external IP of the machine (i.e. the adress used afterwards for http URLs)
Create the following alias in your .bashrc file:
vi ~/.bashrc
alias dcomp='docker-compose -p xivocc -f /etc/docker/compose/docker-xivocc.yml'
The yml file /etc/docker/compose/docker-xivocc.yml
should have the correct tag version for each imeage.
Check also that the XIVO_CTI_VERSION is correct for the xuc container.
xivo_replic :
image: xivoxc/xivo-db-replication:2016.03.latest
xivo_stats :
image: xivoxc/xivo-full-stats:2016.03.latest
pack_reporting:
image: xivoxc/pack-reporting:2016.03.latest
config_mgt:
image: xivoxc/config-mgt:2016.03.latest
recording_server:
image: xivoxc/recording-server:2016.03.latest
xuc:
image: xivoxc/xuc:2016.03.latest
environment:
- XIVO_CTI_VERSION=2.1
xucmgt:
image: xivoxc/xucmgt:2016.03.latest
Then you can launch the XiVO CC with the following command :
dcomp up -d
List XivoCC services :
# dcomp ps
Name Command State Ports
---------------------------------------------------------------------------------------------------------------------
xivocc_config_mgt_1 bin/config-mgt-docker Up 0.0.0.0:9100->9000/tcp
xivocc_elasticsearch_1 /docker-entrypoint.sh elas ... Up 0.0.0.0:9200->9200/tcp, 0.0.0.0:9300->9300/tcp
xivocc_fingerboard_1 /bin/sh -c /usr/bin/tail - ... Up
xivocc_kibana_volumes_1 /bin/sh -c /usr/bin/tail - ... Up
xivocc_nginx_1 nginx -g daemon off; Up 443/tcp, 0.0.0.0:80->80/tcp
xivocc_pack_reporting_1 /bin/sh -c echo ... Up
xivocc_pgxivocc_1 /docker-entrypoint.sh postgres Up 0.0.0.0:5443->5432/tcp
xivocc_postgresvols_1 /bin/bash Exit 0
xivocc_recording_server_1 bin/recording-server-docker Up 0.0.0.0:9400->9000/tcp
xivocc_reporting_rsync_1 /usr/local/sbin/run-rsync.sh Up 0.0.0.0:873->873/tcp
xivocc_spagobi_1 /bin/sh -c /root/start.sh Up 0.0.0.0:9500->8080/tcp
xivocc_timezone_1 /bin/bash Exit 0
xivocc_xivo_replic_1 /usr/local/bin/start.sh /o ... Up
xivocc_xivo_stats_1 /usr/local/bin/start.sh /o ... Up
xivocc_xivocclogs_1 /bin/bash Exit 0
xivocc_xuc_1 bin/xuc_docker Up 0.0.0.0:8090->9000/tcp
xivocc_xucmgt_1 bin/xucmgt_docker Up 0.0.0.0:8070->9000/tcp
The desktop assistant is available through the xucmgt application so you need to deploy this container first.
To download the latest version available on your environment, just open the following url from your computer:
http://<xucmgt_host>:<xucmgt_port>/install/win64
and then start the downloaded program.
To install the latest version, you need to add a repository linked to the xucmgt host. Edit your /etc/apt/sources.list and add the following line:
deb http://<xucmgt_host>:<xucmgt_port>/updates/debian jessie contrib
Then run
sudo apt-get update
sudo apt-get install xivo-desktop-assistant
Note
This repository is currently not signed at all.
Upgrade¶
The following components will be upgraded :
- Docker images
- xivocc-installer package
Warning
This upgrade procedure applies only to XiVO CC installed via the xivocc-installer
package.
Before upgrading you have to check or change your sources list.
It should be located in the file /etc/apt/sources.list.d/xivo-solutions.list
.
There are two cases :
- Upgrade to latest version,
- Upgrade to a specific version (or an archive version)
To upgrade to the latest version the sources list must point towards debian URI and xivo-solutions suite:
deb http://mirror.xivo.solutions/debian/ xivo-solutions main
To upgrade to a specific version the sources list must point towards archive URI and xivo-solutions-VERSION suite.
For example if you want to upgrade to 2016.03 version you should have:
deb http://mirror.xivo.solutions/archive/ xivo-solutions-2016.03 main
Note the /archive/
and -2016.03
above.
When you have checked the sources.list you can upgrade with the following commands:
apt-get update
apt-get install xivocc-installer
The current docker-compose.yml file will be renamed to docker-compose.yml.dpkg-old and new template downloaded. A new docker-compose.yml file will be rendered from the template using the current xivocc version.
Then you run the new version by dcomp up -d eventually preceded by the dcomp pull to download the new images.
Consult the 2016.04 Roadmap
System
Parameters for
/etc/docker/compose/docker-xivocc.yml
are now stored in/etc/docker/compose/.env
file. Important parameter isXIVO_AMI_SECRET
, which holds Ami password.To be able to use the
/etc/docker/compose/.env
file, a newdcomp
alias is generated in .bashrc. You must run:source .bashrc
before running
dcomp
again.
Note
If you are using docker-compose
instead of recommended alias dcomp
, make sure your current directory
is /etc/docker/compose
, otherwise /etc/docker/compose/.env
won’t be used. i.e.:
cd /etc/docker/compose
docker-compose ...
Web/Desktop Assistant
- For displaying search result, compatibility with xivo-dird of XiVO PBX has been enhanced. After upgrade you must verify the configuration of your CTI directory Display in XiVO PBX as described in Directories and Views.
Note
Integration note: the Web and Desktop Assistant support only the display of
- 1 field for name (the one of type name in the directory display)
- 3 numbers (the one of type number and the first two of type callable)
- and 1 email
Callbacks (CCManager)
- Default csv separator has been changed from pipe ‘|’ to comma ‘,’ for the callback export.
No behavior changes.
Reporting and statistics¶
Pack reporting is a part of the XivoCC, but can also be installed separately. It aims at computing historical statistics, which are stored in the xivo_stats database. Sample reports based on them are accessible in SpagoBI.
Warning
Full installation of the pack reporting requires restarting XiVO services, so telephone communcations will be cut.
- Install docker by following installation instructions: http://docs.docker.com/installation/
- Execute the following commands:
wget https://gitlab.com/xivoxc/packaging/raw/master/install/install-docker-reporting.sh bash install-docker-reporting.sh docker-compose -f /etc/docker/compose/docker-reporting.yml up -dDuring installation you will be asked for: * the XiVO IP address * the number of weeks to keep in history
At the end of the installation some configuration must be done on the XiVO:
edit /var/lib/postgresql/9.1/main/postgresql.conf and set listen_addresses to *
edit /var/lib/postgresql/9.1/main/pg_hba.conf and add the following line: host asterisk stats PACK_REPORTING_IP/32 md5
add the following events to /etc/asterisk/cel.conf: HOLD,UNHOLD,BLINDTRANSFER,ATTENDEDTRANSFER
execute the following command:
sudo -u postgres psql asterisk << EOF CREATE USER stats WITH PASSWORD 'stats'; GRANT SELECT ON ALL TABLES IN SCHEMA PUBLIC TO stats; EOF
Finish the installation by a full restart of XiVO:
xivo-service restart all
- Docker containers compose_xivo_replic_1, compose_xivo_stats_1 and compose_pack_reporting_1 should be started
- There should be no errors in /var/log/xivocc/xivo-db-replication/xivo-db-replication.log and /var/log/xivocc/xivo-full-stats/xivo-full-stats.log
- Data replication can take a long time, so you may need to be patient before finding data in the reports
- Some panels are preconfigured :
- To save these panels in Elasticsearch and make them accessible through Kibana menu, you will have to authenticate with admin/Kibana
Queue members should only be agents. If users are members of a queue, their statistics will be incomplete.
Configuration modifications on the XiVO (such as an agent deletion) are replicated on the statistics server, and their previous value is not kept. However, statistics history is preserved.
POPC statistics are wrong.
If two agents are associated to the same call, they will have the same hold time for this call.
Transfer statistics limitation : given two queues Q1 and Q2, two agents A1 and A2, and an external caller C.
- C calls Q1 and A1 answers
- A1 transfers to Q2 and A2 answers
- A2 transfers to the outside
Then the second transfer is seen as a transfer to the outside.
The pack reporting allows to attach as mush data as wished to a given call, in order to find them in the reporting database for future use. This data must be in the form of a set of key-value pairs.
To attach data to a call, you must use the dialplan’s CELGenUserEvent application:
exten = s,n,CELGenUserEvent(ATTACHED_DATA,my_key=my_value)
This will insert the following tuple in the attached_data table:
key | value |
---|---|
my_key | my_value |
These notes include upgrade procedures for old versions of the Pack reporting, before XivoCC starts and before it was packaged with Docker. In those cases, run the following command to find the installed version of the pack reporting:
dpkg -l|grep pack-reporting
- data retention time will be lost during upgrade : save it and write it back in /etc/xivo-reporting-db.conf
- the upgrade is likely to be long if there is a lot of data in queue_log. Purge old data out of this table if possible in order to accelerate the upgrade
- at the end of the upgrade, run apt-get autoremove (deletion of xivo-stat, xivo-libdao and xivo-lib-python)
- XiVO in version < 14.08 is not supported anymore
- if it is required, the upgrade of the XiVO must be done before the upgrade of the pack reporting, and no call must be performed between the two upgrades
- Beware: this will require a migration of the original PostgreSQL database to the Dockerised one. For this you need to have free disk space : the amount of free disk space must equal the size of /var/lib/postgresql. This check must be performed after docker images have been pulled.
- Run the following commands:
apt-get update
apt-get install pack-reporting xivo-full-stats xivo-reporting-db xivo-db-replication db-utils
service xivo-db-replication stop
service xivo-full-stats stop
wget https://gitlab.com/xivoxc/packaging/raw/master/install/install-docker-reporting.sh
bash install-docker-reporting.sh
docker-compose -f /etc/docker/compose/docker-reporting.yml up -d pgxivocc
# Database migration. CHECK THE FREE DISK SPACE
sudo -u postgres pg_dump --format c xivo_stats | docker exec -i xivocc_pgxivocc_1 pg_restore -U postgres -d xivo_stats
docker-compose -f /etc/docker/compose/docker-reporting.yml up -d
- Run the following commands:
docker exec -ti compose_pgxivocc_1 psql -U postgres -c 'CREATE EXTENSION IF NOT EXISTS "uuid-ossp"' xivo_stats
docker exec -ti compose_pgxivocc_1 psql -U postgres -c 'CREATE EXTENSION IF NOT EXISTS "uuid-ossp"' xuc_rights
Calls list
Column | Type | Description |
---|---|---|
id | INTEGER | |
uniqueid | VARCHAR | Call unique reference, generated by Asterisk |
dst_num | VARCHAR | Called number |
start_time | TIMESTAMP | Call start time |
answer_time | TIMESTAMP | Call answer time |
end_time | TIMESTAMP | Call end time |
status | status_type | Call status. Beware: only answered is properly filled. |
ring_duration_on_answer | INTEGER | Ring time of the endpoint answering the call, in seconds |
transfered | BOOLEAN | True if the call has been transfered |
call_direction | call_direction_type | Call direction (‘’incoming’’ : call from the outisde, received by XiVO ; ‘’outgoing’’ : call to the outside, originated by an endpoint associated to XiVO ; ‘’internal’’ : call taking place entirely inside the XiVO) |
src_num | VARCHAR | Calling number |
transfer_direction | call_direction_type | Indicates the transfer direction, if relevant |
src_agent | VARCHAR | Agent originating the call |
dst_agent | VARCHAR | Agent receiving the call, if it is a direct call on an agent. Not filled when the call is destined to a queue |
src_interface | VARCHAR | Interface originating the call (in the Asterisk sense, ex : SCCP/01234) |
Data attached to the call (cf. Attached Data)
Column | Type | Description |
---|---|---|
id | INTEGER | |
id_call_data | INTEGER | Id of the associated tuple in call_data |
key | VARCHAR | Name of the attached data |
value | VARCHAR | Value of the attached data |
Part of a call matching the reaching of an endpoint
Column | Type | Description |
---|---|---|
id | INTEGER | |
call_data_id | INTEGER | Id of the associated tuple in call_data |
start_time | TIMESTAMP | Time at which the endpoint was called |
answer_time | TIMESTAMP | Asnwer time for the endpoint |
end_time | TIMESTAMP | End time of this call part |
interface | VARCHAR | Endpoint interface |
Calls on a queue
Column | Type | Description |
---|---|---|
id | INTEGER | |
callid | VARCHAR | Call unique reference, generated by Asterisk |
queue_time | TIMESTAMP | Time of entrance in the queue |
total_ring_seconds | INTEGER | Total ring time, in seconds (includes ringing of non-answered calls) |
answer_time | TIMESTAMP | Answer time |
hangup_time | TIMESTAMP | Hangup time |
status | call_exit_type | Call status (full: full queue; closed: closed queue; joinempty: call arrived on empty queue; leaveempty : exit when queue becomes empty; divert_ca_ratio : call redirected because the ratio waiting calls/agents was exceeded ; divert_waittime: call redirected because estimated waiting time was exceeded; answered: call answered ; abandoned: call abandoned; timeout : maximum waiting time exceeded) |
queue_ref | VARCHAR | Technical queue name |
agent_num | VARCHAR | Number of the agent taking the call, if relevant |
Hold periods
Column | Type | Description |
---|---|---|
id | INTEGER | |
linkedid | VARCHAR | Call unique reference, generated by Asterisk |
start | TIMESTAMP | Hold start time |
end | TIMESTAMP | Hold end time |
Statistics aggregated by queue and time interval (15 minutes)
Column | Type | Description |
---|---|---|
id | INTEGER | |
time | TIMESTAMP | Start time of the considered interval |
queue | VARCHAR | Queue technical name |
answered | INTEGER | Number of answered calls |
abandoned | INTEGER | Number of abandoned calls |
total | INTEGER | Total number of calls received on the queue (which excludes the calls dissuaded before entering the queue) |
full | INTEGER | Number of calls arrived on a full queue (diversion before entering the queue) |
closed | INTEGER | Number of calls arrived on a closed queue, outsided of the configured schedules (diversion before entering the queue) |
joinempty | INTEGER | Number of calls arrived on an empty queue (diversion before entering the queue) |
leaveempty | INTEGER | Number of calls redirected becouse of a queue becoming empty |
divert_ca_ratio | INTEGER | Number of calls arrived when the calls / available agents ratio is exceeded (diversion before entering the queue) |
divert_waittime | INTEGER | Number of calls arrived when the estimated waiting time is exceeded (diversion before entering the queue) |
timeout | INTEGER | Nombre of calls redirecting because maximum waiting time is exceeded |
Statistics aggregated by agent and time interval (15 minutes)
Column | Type | Description |
---|---|---|
id | INTEGER | |
time | TIMESTAMP | Start time of the considered interval |
agent | VARCHAR | Agent number |
login_time | INTERVAL | Login time |
pause_time | INTERVAL | Pause time |
wrapup_time | INTERVAL | Wrapup time |
Statistics aggregated by queue, called number and time interval (15 minutes)
Column | Type | Description |
---|---|---|
time | TIMESTAMP | Start time of the considered interval |
queue_ref | VARCHAR | Technicxal name of the queue |
dst_num | VARCHAR | Called number |
nb_offered | INTEGER | Number of presented calls |
nb_abandoned | INTEGER | Number of abandoned calls |
sum_resp_delay | INTEGER | Wait time, in seconds |
answer_less_t1 | INTEGER | Number of calls answered in less than t1 seconds |
abandoned_btw_t1_t2 | INTEGER | Number of calls abandoned between t1 and t2 seconds |
answer_btw_t1_t2 | INTEGER | Number of calls answered between t1 and t2 seconds |
abandoned_more_t2 | INTEGER | Number of calls answered in more than t2 seconds |
communication_time | INTEGER | Total communication time in seconds |
hold_time | INTEGER | Total hold time in seconds |
wrapup_time | INTEGER | Total wrapup time in seconds |
The thresholds t1 and t2 are configurable:
- in the table queue_specific_time_period for the default values in seconds. Installation values are t1=15 seconds and t2=20 seconds. Data is saved in the form of (name, seconds) pairs, for example : (‘t1’, 15).
- in the table queue_threshold_time for values specific to a queue. Data is saved in the form of a tuple (queue name, t1, t2).
Statistics aggregated by agent and time interval (15 minutes)
Column | Type | Description |
---|---|---|
time | TIMESTAMP | Start time of the considered interval |
agent_num | VARCHAR | Agent number |
nb_offered | INTEGER | Number of calls presented from a queue |
nb_answered | INTEGER | Number of calls answered from a queue |
conversation_time | INTEGER | Conversation time on incoming calls from a queue, in seconds |
ringing_time | INTEGER | Ringing time on incoming cals from a queue, in seconds |
nb_outgoing_calls | INTEGER | Number of calls emitted to the outside |
conversation_time_outgoing_calls | INTEGER | Conversation time in calls emitted to the outside, in seconds |
hold_time | INTEGER | Hold time for calls from a queue, in seconds |
nb_received_internal_calls | INTEGER | Number of received internal calls |
conversation_time_received_internal_calls | INTEGER | Conversation time on received interbal calls, in seconds |
nb_transfered_intern | INTEGER | Number of calls coming from a queue and transfered to an internal destination |
nb_transfered_extern | INTEGER | Number of calls coming from a queue and transfered to an external destination |
nb_emitted_internal_calls | INTEGER | Number of emitted interbal calls |
conversation_time_emitted_internal_calls | INTEGER | Conversation time on emitted internal calls, in seconds |
nb_incoming_calls | INTEGER | Number of received incoming calls |
conversation_time_incoming_calls | INTEGER | Conversation time on received incoming calls, in seconds |
Statistics aggregated by queue, called number, agent and time interval (15 minutes)
Column | Type | Description |
---|---|---|
time | TIMESTAMP | Start time of the considered interval |
agent_num | VARCHAR | Agent number |
queue_ref | VARCHAR | Technicxal name of the queue |
dst_num | VARCHAR | Called number |
nb_answered_calls | INTEGER | Number of answered calls |
communication_time | INTEGER | Communication time, in seconds |
hold_time | INTEGER | Hold time, in seconds |
wrapup_time | INTEGER | Wrapup time, in seconds |
Tables call_data, call_on_queue et hold_periods can be linked together by doing a join on a column holding the call reference. The columns are the following:
Table | Reference column |
---|---|
call_data | uniqueid |
call_on_queue | callid |
hold_periods | linkedid |
D’autre part, les tables attached_data et call_element contiennent une clef étrangère référençant la colonne id de call_data.
Kibana is a web tool used to compute statistics based on Elasticsearch content. The reports packaged with the Pack reporting give you an outline of your recent call center activity. Here is a Kibana sample panel:
Graphs are based on the queue_log table, enriched with agent names and agent groups, and inserted into an Elasticsearch index. It contains avents about calls placed on queues, and events about agent presences.
For each entry in the queue_log index, the following attributes are available:
- queudisplayname : Queue display name
- data1: basic queue_log data, with a different meaning according to the event
- callid : Call unique identifier, generated by Asterisk
- event : Call or agent status event - please see below
- agentnumber: Agent number
- queuename : Technical queue name
- groupname : Agent group name
- queuetime: Time of the event
- agentname : Name of the agent, if available
The event can be one of the following (for a detailed explanation, please refer to https://wiki.asterisk.org/wiki/display/AST/Queue+Logs):
- Call events:
- FULL
- CONNECT
- EXITEMPTY
- CLOSED
- EXITWITHTIMEOUT
- JOINEMPTY
- ABANDON
- ENTERQUEUE
- TRANSFER
- COMPLETEAGENT
- COMPLETECALLER
- RINGNOANSWER
- Agent or queue event:
- ADDMEMBER
- PAUSEALL
- PAUSE
- WRAPUPSTART
- UNPAUSE
- UNPAUSEALL
- PENALTY
- CONFIGRELOAD
- AGENTCALLBACKLOGIN
- AGENTCALLBACKLOGOFF
- REMOVEMEMBER
- PRESENCE
- QUEUESTART
Phone integration¶
XUC based web applications like agent interface or xivo client web integrates buttons for phone control. This section details necessary configuration, supported phones and limitations.
Note: The voip vlan network have to be accessible by the xivocc xuc server
Manufacturer | Function | |||||
---|---|---|---|---|---|---|
Answer | Hangup | Hold | Conference | Attended Transfer | Direct Transfer | |
Snom 7XX | OK | OK | OK | OK | OK | OK |
Polycom VVX | OK | OK | OK | NO | OK | OK |
Yealink T4X | OK | OK | OK | NO | OK | OK |
- NO - Not available
To enable phone control buttons on web interfaces you must update the basic template of Polycom phones:
- go to the plugin directory: /var/lib/xivo-provd/plugins/xivo-polycom-VERSION
- copy the default template from templates/base.tpl to var/templates/
- then you must update app.push parameters in the else section (do not replace switchboard settings) as follows:
apps.push.messageType="5"
apps.push.username="guest"
apps.push.password="guest"
To enable phone control buttons on web interfaces you must update the basic template of Yealink phones:
- go to the plugin directory: /var/lib/xivo-provd/plugins/xivo-yealink-VERSION
- copy the default template from templates/base.tpl to var/templates/
- enable sip notify even for non switchboard profiles (do not replace switchboard settings)
{% if XX_options['switchboard'] -%}
push_xml.sip_notify = 1
call_waiting.enable = 0
{% else -%}
push_xml.sip_notify = 1
call_waiting.enable = 1
{% endif %}
- to update device configuration you must run xivo-provd-cli -c 'devices.using_plugin("xivo-polycom-VERSION").reconfigure()'
- and finally you must resynchronize the device: xivo-provd-cli -c 'devices.using_plugin("xivo-polycom-VERSION").synchronize()'
- refer to provisioning documentation for more details
- if the phone synchronization fails check if the phone uses the version of the plugin you have updated, you can use xivo-provd-cli -c 'devices.find()'
Third Party Integration¶
Third party web application integration is possible inside the XucMgt Agent application since XucMgt version 1.49.0. Upon each call, you can display a custom tab inside the agent interface:
When a call is ringing on the agent phone, the Application will call the external web service (see Configuration below). The web service response will dictate the behaviour of the integration. For example, if the speficied action is to open the application when the call is hung up, a new tab will be created and opened inside the agent interface, showing the content specified by the web service response. (see Web Service API for available options).
When the work is complete in the integrated application, the application must post a message to terminate the third party application pane inside the agent application (see Completion).
You need to specify the third party application web service url to integrate this application inside the XucMgt Agent interface. This can be done by specifying a THIRD_PARTY_URL environment variables.
For example, inside the dockerfile, in the XucMgt section:
environment:
...
- THIRD_PARTY_URL=http://some.url.com/ws/endpoint
The speficied URL must be accessible from the client browser (i.e. the end user of the Agent application). The call wil be made from his browser.
The Web Service url specified in the :Configuration must conforms to the following behaviour.
The service will receive a POST request with a payload as application/json
, for example:
{
"user":{
"userId":4,
"agentId":1,
"firstName":"James",
"lastName":"Bond",
"fullName":"James Bond"
},
"callee":"1000",
"caller":"1001",
"queue":{
"id":2,
"name":"trucks",
"displayName":"Trucks",
"number":"3001"
},
"userData":{
"XIVO_CONTEXT":"default",
"XIVO_USERID":"2",
"XIVO_SRCNUM":"1001",
"XIVO_DSTNUM":"3001"
}
}
user
contains the connected user informationcallee
contains the number calledqueue
queue propertiesuserData
call data presented by Xivo
The Web service must answer with an application/json
content. For example:
{
"action":"open",
"event":"EventReleased",
"url":"/thirdparty/open/6bd37819-b4a6-43d3-8fa3-6eb6489bb705",
"autopause":true,
"title":"Third Party Sample"
}
or:
{
"action":"none"
}
action
is one of"open"
or"none"
event
is one of"EventRinging"
,"EventEstablished"
,"EventReleased"
. The third party application will be opened when one the specified event occursurl
should be the url to open inside the application. This url should point to a valid web application that can be specific for each call.autopause
if set to true, the agent will be put on pause when the application pane is opened and back to ready when the application is completed.title
will set the title of the tabs that will be opened.
Warning, when the XucMgt application and the integrated application are on different server, domain, url,... (which should be common case), You may get CORS errors. To workaround this issue, you should implement the OPTIONS request on your web service. This method will be called by the browser before issuing the POST request to ensure the target web server allows calls from the original application. You application must set at least the following headers in order to overcome the CORS errors:
Access-Control-Allow-Origin
: * or the domain hosting the XucMgt applicationAccess-Control-Allow-Methods
: POST, OPTIONS (at least)Access-Control-Allow-Headers
: Origin, X-Requested-With, Content-Type, Accept (at least)
Once the work is complete inside the third party application, it should post a completion message (closeThirdParty
) to the application using the Web Messaging API.
For example, here is how to define a close method in javascript to send the message to the hosting application and bind it to a simple button:
(function () {
function close() {
parent.window.postMessage("closeThirdParty", "*");
}
document.getElementById("close").addEventListener("click", close, false);
})();
Features¶
Contact center management¶
CCmanager is a web application to manage and supervise a contact center
Display queues
Display agents / agents status
Move or add agents in queue / penalty
Move of add group of agents in queue / penalty
- Action on agents
- Login / Logout
- Pause / Unpause
- Listen [1]
Start the application : http://<xuc:port>/ccmanager
This interface allows a user to change queue assignement and the associated penalty. The queue table display the following columns
- “Number”: The queue number
- “Name”: The queue name
- “Penalty”: The active penalty for the corresponding queue
- “default”: The default penalty for the corresponding queue
The queue/active penalty couples can be saved as default configuration by clicking the “Set default” button, then “Save”. The queue/default penalty couples can be saved as active configuration by clicking the “Set current” button, then “Save”.
- Emptying the penalty textbox and saving will remove the queue from the active configuration for the agent.
- Emptying the default textbox and saving will remove the queue from the default configuration for the agent.
From agent view you are able to add or remove more than one agent at the same time.
Once the agent selection is done, click on the edit button to display the configuration window
Click on the plus button to add a queue for selection, click on the minus button to remove a queue to the selection. Once queue to add or removed are choosen, click on save button to apply your configuration change.
Click on “Apply default configuration” to apply existing default configuration to all selected users and make it the active configuration. This action only affects users with an existing default configuration, agents whithout default configuration remain unchanged.
From the agent view, after selecting one or more agents, you can create a base configuration by clicking on one of the menu item in the following drop down:
- ‘Create base configuration’ will allow you to create a base configuration from scratch for all the selected agents.
- ‘Create base configuration from active configuration’ will allow you to create a base configuration using the selected agents active configuration. The queue membership and penalty populated will be built based on the merged membership of all the selected agents. In case of conflict, the lowest penalty will be used.
In both cases, you will be able to review your changes before applying them. The ‘Create base configuration’ popup is similar to the single agent edition popup:
The queue table display the following columns:
- “Number”: The queue number
- “Name”: The queue name
- “Penalty”: The active penalty for the corresponding queue
Click on the plus button to add a queue for selection. Once your configuration is complete, click on save button to apply your configuration change.
Color thresholds can be defines for the waitinig calls counter and the maximum waiting time counter
Applys to the queue view and the global view
This view allows to manage callback request : importing a new list of callbacks, monitoring them and downloading the associated tickets.
Callbacks can be imported from a CSV file into a callback list.
Line delimiter must be a new line character and column separator must be one of: ‘|’ or ‘,’ or ‘;’. Columns can be optionaly enclosed by double-quote ‘”’.
The file must look like the following:
phoneNumber|mobilePhoneNumber|firstName|lastName|company|description|dueDate|period
0230210092|0689746321|John|Doe|MyCompany|Call back quickly||
0587963214|0789654123|Alice|O'Neill|YourSociety||2016-08-01|Afternoon
The header line must contain the exact field named described below:
- phoneNumber: The number to call (at least either phoneNumber or mobilePhoneNumber is required)
- mobilePhoneNumber: Alternate number to call
- firstName: The contact first name (optional)
- lastName: The contact last name (optional)
- company: The contact company name (optional)
- description: A text that will appear on the agent callback pane
- dueDate: The date when to callback, using ISO format: YYYY-MM-DD, ie. 2016-08-01 for August, 1st, 2016. If not present the next day will be used as dueDate (optional)
- period: The name of the period as defined in callback list. If not present, the default period will be used (optional)
When an agent takes a callback, the column Taken by
is updated with the number of the aget. The callback disappears when it is processed.
The tickets of the processed callbacks can be downloaded by clicking on the Download tickets
button.
The downloaded file is a csv file with the comma ‘,’ as delimiter.
Footnotes
[1] | Only supervisors which have their own lines can listen to agents, no supported on mobile supervisors, a line has to be affected to supervisors in xivo |
Agent environment¶
Web application for contact center agents
Recording can be paused or started by an agent, this feature can be disabled by changing showRecordingControls option in application.conf, you can also set the environnment variable SHOW_RECORDING_CONTROLS to false for your xucmgt container in docker compose yml file. When disabled the recording status is not displayed any more
Callbacks panel can be removed using by changing showCallbacks option in application.conf, you can also use SHOW_CALLBACKS environment variable in docker compose yml file.
By using the showQueueControls option in application.conf, you may allow an agent to enter or leave a queue. You can also use SHOW_QUEUE_CONTROLS environment variable in docker compose yml file.
Enter your XiVO client username, password and phone set you want to logged to on the login page. If you are already logged on an other application (CCmanager, Web Assistant) you only need to enter the phone number.
The agent can see the callbacks related to the queues he is logged on.
They are available in the Callbacks
tab, beside the Agents of my group
tab.
On this page, the agent only has access to basic information about the callback: the phone number to call, the person’s name and its company name. On the left of each callback line, a colored clock indicates the temporal status of this callback:
- yellow if the callback is to be processed later
- green if we are currently inside the callback period
- red if the callback period is over
To process one of these callbacks, the agent must click on one of the callbacks line. This will remove the callback from the other agents’ list, and trigger the following screen:
To launch the call, the agent must click on one of the available phone numbers. Once the callback is launched, the status can be changed and a comment can be added.
If you set ‘Callback’ as status, the callback can be rescheduled at a later time and another period:
Clicking on the calendar icon next to the “New due date” field, will popup a calendar to select another callback date.
It is possible to display customer information in an external web application using Xivo sheet mecanism.
You must define a sheet with two fields
- folderNumber
have to be defined. Can be calculated or use a default value not equal to “-“
- popupUrl
The url to open when call arrives : i.e. http://mycrm.com/customerInfo?folder= the folder number will be automatically appended to the end of the URL
Example : Using the caller number to open a customer info web page
- Define folderNumber with any default value i.e. 123456
- Define popupUrl with a display value of http://mycrm.com/customerInfo?nb={xivo-calleridnum}&fn= when call arrives web page http://mycrm.com/customerInfo?nb=1050&fn=123456 will be displayed
Configuration Management¶
The callback system in XivoCC aims at performing outgoing calls to specific numbers, to which some information can be associated such as a description ar a personal name.
The core object of the callback system is the callback request. A callback request is made of the following fields:
- First name of person to call
- Last name
- Phone number
- Mobile phone number
- Company name
- Description
- Due date
Each callback request is associated to a predefined callback period, which represents the preferred interval of the day in which the call should be performed.
A callback request cannot exist on its own: it must be stored in a callback list, which is itself associated to a queue.
Once a callback request has been performed, it generates a callback ticket. This ticket sums up the original information of the callback request, but adding some new fields:
- Start date: date at which the callback request was actually performed
- Last update: date of the last modification of the ticket
- Comment
- Status : the result of the callback
- Agent: the Call Center agent who performed the callback
A callback list is an object which will contain callback request. It is associated to a queue, and several callback lists can be associated to the same queue.
Once created, a list can be populated whether through the Callbacks tab of the CCManager, or programmatically through the web services of the configuration server.
A callback period represents an interval of the day, bounded by a start date and an end date. It can be set as the default interval, so that a newly created callback request will be associated to this period if none is specified.
Desktop Assistant¶
On first launch the application will display the settings page and ask for the xucmgt application address.
This page allows you to specify the protocol and address of the xucmgt application.
- Check “Secure” if you use “https” protocol to connect to the xucmgt application or check “Unsecure” otherwise.
- Enter the xucmgt application host address and port.
On Windows, the application will check at startup for a new version of the application and offer to upgrade if one is available.
On Debian, the update relies on the package manager behaviour. However you can check for any update by issuing the following commands:
sudo apt-get update
apt-cache policy xivo-desktop-assistant
The Desktop Assistant can be started with following options:
- -d to enable debug menu items
- –ignore-certificate-errors to disable certificate verification, this option is meant only for test purposes. You can use it with self-signed certificates.
The Desktop Assistant can be used by users with WebRTC configuration, without physical phone.
For configuration and requirements, see WebRTC Requirements.
Web Assistant¶
The Web Assistant enables a user to:
- search contacts,
- call them,
- manage its favorites,
- manage its forward.
To login, one must have a user configured on the XiVO PBX with:
- XiVO Client enabled,
- Login, password and profile configured
You can use the search section to lookup for people in the company:
For this to work, one must configure the directories in the XiVO PBX as described in Directories and Views.
Note
Integration note: the Web Assistant support only the display of
- 1 field for name (the one of type name in the directory display)
- 3 numbers (the one of type number and the first two of type callable)
- and 1 email
One can clic on the star to put a contact in its list of favorites.
For this to work, favorites must be configured in the XiVO PBX as described in Favorites.
The Web Assistant can integrate with the phone to :
- call,
- put on hold,
- transfer
- etc.
For this feature to work one must use a Supported phones and follow the Required configuration page.
The Web Assistant can be used by users with WebRTC configuration, without physical phone.
For configuration and requirements, see WebRTC Requirements.
WebRTC¶
Note
added in version 2016.04
From version 2016.04 one can use WebRTC with XiVO PBX and XiVO CC in the following environment:
- LAN network (currently no support for WAN environment),
- with the:
- Web Assistant with Chrome browser version 55 (tested on 55.0.2883.87 m 64-bit),
- or Desktop Assistant
The requirements are:
- to have a microphone and headphones for your PC,
- to configure, in the XiVO PBX, a user with a WebRTC line (see: Configuration of user with WebRTC line),
- have a SSL/TLS certificate signed by a certification authority installed on the nginx of XiVO CC,
- and use https:
- Web Assistant: you must connect to the Web Assistant via https protocol,
- Desktop Application: you must check Protocol -> Secure in the application parameters.
Note
Currently you can not have a user configured for both WebRTC and a phone set at the same time.
Administration¶
Log¶
The log of each components can be found in the /var/log/xivocc directory. Currently (it may change) the structure looks like this :
/var/log/xivocc :
├── purge-reporting-database.log
├── specific-stats.log
├── xivo-db-replication.log
├── xivo-full-stats.log
├── recording-server
│ ├── dowloads.log
│ ├── downloads.log
│ └── recording-server.log
├── xuc
│ └── xuc.log
└── xucmgt
└── xucmgt.log
Backup¶
You may backup your statistic database by using a similar command as below
docker run --rm --link demo_pgxivocc_1:db -v $(pwd):/backup -e PGPASSWORD=xivocc postgres pg_dump -h db -U postgres --format=c -f /backup/xivo_stats_dump xivo_stats
Restore¶
You may restore a backup using a similar command (to be adapted)
docker run --rm -it --link pgxivoccdemo_pgxivocc_1:db -v $(pwd):/backup postgres pg_restore -h db -c -U postgres -d xivo_stats /backup/xivo_stats_dump
Xuc Xivo Unified Communication Framework¶
Xuc is an application suite developed by Avencall Group, based on several free existing components including XiVO, and our own developments to provide communication services api and application to businesses. Xuc is build on Play using intensively Akka and written in Scala
XiVO is free software. Most of its distinctive components, and Xuc as a whole, are distributed under the LGPLv3 license.
Xuc is providing
- Javascript API
- Web services
- Sample application
- Simple agent application
- Simple unified communication application pratix
- Contact center supervision
- Contact center statistics
The proposed applications are available in English and French. The list of preferred langs sent by the browser is analyzed and the first known lang is used, so if the browser requests it, en and fr the page will be server in en. The fallback language is French. Contributions are welcome, start with opening an issue on gitlab project page.
Xuc is composed of 3 modules
- The server module
- The core module
- The statistic module.
Developer¶
- Xivo Java Cti lib ; https://gitorious.org/xivo/xivo-javactilib
- mvn install
- theatrus/akka-quartz : https://github.com/theatrus/akka-quartz
- sbt publish-local
(sudo apt-get install devscripts)
- dch -i in project root directory, parent of debian/changelog
- edit changelog to add version
- update src/sphinx/conf.py with new version
- activator make-site
- copy target/sphinx/docs content to public/doc
- Create debian package : activator debian:genChanges
Building docker image:
activator docker:publish
or
activator docker:publishLocal
docker tag xivo/xuc:2.4.32 xivo/xuc:latest
activator clean test docker:publishLocal; docker tag -f xivo/xuc:1.9.0 xivo/xuc:latest;docker push xivo/xuc:1.9.0; docker push xivo/xuc:latest
The Xuc documentation uses reStructuredText as its markup language and is built using Sphinx.
For more details see The Sphinx Documentation
For more details see The reST Quickref
Quick Reference
- http://docutils.sourceforge.net/docs/user/rst/cheatsheet.txt
- http://docutils.sourceforge.net/docs/user/rst/quickref.html
- http://openalea.gforge.inria.fr/doc/openalea/doc/_build/html/source/sphinx/rest_syntax.html
Section headings are very flexible in reST. We use the following convention in the Xuc documentation:
#
(over and under) for module headings=
for sections-
for subsections^
for subsubsections~
for subsubsubsections
Sections that may be cross-referenced across the documentation should be marked
with a reference. To mark a section use .. _ref-name:
before the section
heading. The section can then be linked with :ref:`ref-name`
. These are
unique references across the entire documentation.
For example:
.. _xuc-module:
#############
Xuc Module
#############
This is the module documentation.
.. _xuc-section:
Xuc Section
============
Xuc Subsection
---------------
Here is a reference to "xuc section": :ref:`xuc-section` which will have the
name "Xuc Section".
For the html and pdf version of the docs:
activator make-site
open <project-dir>/target/sphinx/docs/index.html
open <project-dir>/target/sphinx/docs/Xuc-doc.pdf
To be able to generate pdf and documentation you need install Sphinx and other tools:
sudo easy_install -U Sphinx
sudo apt-get install texlive-latex-base texlive-latex-recommended texlive-latex-extra texlive-fonts-recommended
Javascript API¶
The Xuc javascript API enables you to integrate enterprise communication functions to your business application. It exposes Cti functions using javascript methods calls.
You may add your own handlers for your application to react to telephony / contact center events.
This API is using websockets, and therefore needs a modern browser supporting them (firefox, chrome ...)
- Include the Cti and Callback javascript API from the Xuc Server
<script src="http://<xucserver>:<xucport>/assets/javascripts/shotgun.js" type="text/javascript"></script>
<script src="http://<xucserver>:<xucport>/assets/javascripts/cti.js" type="text/javascript"></script>
<script src="http://<xucserver>:<xucport>/assets/javascripts/callback.js" type="text/javascript"></script>
<script src="http://<xucserver>:<xucport>/assets/javascripts/membership.js" type="text/javascript"></script>
- Include also the xc_webrtc and SIPml5 javascript APIs for the webRTC support:
<script src="http://<xucserver>:<xucport>/assets/javascripts/xc_webrtc.js" type="text/javascript"></script>
<script src="http://<xucserver>:<xucport>/assets/javascripts/SIPml-api.js" type="text/javascript"></script>
- Connect to the Xuc serveur using XiVO client username and password
var wsurl = "ws://"+server+"/ctichannel?username="+username+"&agentNumber="+phoneNumber+"&password="+password;
Cti.WebSocket.init(wsurl,username,phoneNumber);
- Setup event handlers to be notified on
- Phone state changes
- Agent state changes
- Statistics
- ...
- Eventually also webRTC handlers
- general
- register
- incoming
- outgoing
- Once web socket communication is established you are able to call XuC Cti javascript methods.
- Place a call, log an agent ....
...
$('#login_btn').click(function(event){
Cti.loginAgent($('#agentPhoneNumber').val());
});
$('#logout_btn').click(function(event){
Cti.logoutAgent();
});
$('#xuc_dial_btn').click(function(event){
Cti.dial($("#xuc_destination").val());
});
...
A sample application is provided by the XuC server. This application allows to display events and using different methods exposed by the XuC
http://<sucserver>:<xucport>/sample
You may browse and use the sample.js
javascript file as an example
- Calling Cti methods :
.$('#xuc_login_btn').click(function(event) {
Cti.loginAgent($('#xuc_agentPhoneNumber').val());
});
$('#xuc_logout_btn').click(function(event) {
Cti.logoutAgent();
});
$('#xuc_pause_btn').click(function(event) {
Cti.pauseAgent();
});
$('#xuc_unpause_btn').click(function(event) {
Cti.unpauseAgent();
});
$('#xuc_subscribe_to_queue_stats_btn').click(function(event) {
Cti.subscribeToQueueStats();
});
$('#xuc_answer_btn').click(function(event) {
Cti.answer();
});
$('#xuc_hangup_btn').click(function(event) {
Cti.hangup();
});
$('#xuc_login_btn').click(function(event) {
Cti.loginAgent($('#xuc_agentPhoneNumber').val());
});
$('#xuc_logout_btn').click(function(event) {
Cti.logoutAgent();
});
$('#xuc_togglelogin_btn').click(function(event) {
Cti.toggleAgentLogin();
});
$('#xuc_pause_btn').click(function(event) {
Cti.pauseAgent();
});
$('#xuc_unpause_btn').click(function(event) {
Cti.unpauseAgent();
});
$('#xuc_subscribe_to_queue_stats_btn').click(function(event) {
Cti.subscribeToQueueStats();
});
$('#xuc_answer_btn').click(function(event) {
Cti.answer();
});
$('#xuc_hangup_btn').click(function(event) {
Cti.hangup();
});
$('#xuc_get_agent_call_history').click(function() {
Cti.getAgentCallHistory(7);
});
$('#xuc_get_user_call_history').click(function() {
Cti.getUserCallHistory(7);
});
..............
- Declaring events handlers :
Cti.setHandler(Cti.MessageType.USERSTATUSES, usersStatusesHandler);
Cti.setHandler(Cti.MessageType.USERSTATUSUPDATE, userStatusHandler);
Cti.setHandler(Cti.MessageType.USERCONFIGUPDATE, userConfigHandler);
Cti.setHandler(Cti.MessageType.LOGGEDON, loggedOnHandler);
Cti.setHandler(Cti.MessageType.PHONESTATUSUPDATE, phoneStatusHandler);
Cti.setHandler(Cti.MessageType.VOICEMAILSTATUSUPDATE, voiceMailStatusHandler);
Cti.setHandler(Cti.MessageType.LINKSTATUSUPDATE, linkStatusHandler);
Cti.setHandler(Cti.MessageType.QUEUESTATISTICS, queueStatisticsHandler);
Cti.setHandler(Cti.MessageType.QUEUECONFIG, queueConfigHandler);
Cti.setHandler(Cti.MessageType.QUEUELIST, queueConfigHandler);
Cti.setHandler(Cti.MessageType.QUEUEMEMBER, queueMemberHandler);
Cti.setHandler(Cti.MessageType.QUEUEMEMBERLIST, queueMemberHandler);
Cti.setHandler(Cti.MessageType.DIRECTORYRESULT, directoryResultHandler);
Cti.setHandler(Cti.MessageType.AGENTCONFIG, agentConfigHandler);
Cti.setHandler(Cti.MessageType.AGENTLIST, agentConfigHandler);
Cti.setHandler(Cti.MessageType.AGENTGROUPLIST, agentGroupConfigHandler);
Cti.setHandler(Cti.MessageType.AGENTSTATEEVENT, agentStateEventHandler);
Cti.setHandler(Cti.MessageType.AGENTERROR, agentErrorHandler);
Cti.setHandler(Cti.MessageType.ERROR, errorHandler);
Cti.setHandler(Cti.MessageType.AGENTDIRECTORY, agentDirectoryHandler);
Cti.setHandler(Cti.MessageType.CONFERENCES, conferencesHandler);
Cti.setHandler(Cti.MessageType.CALLHISTORY, callHistoryHandler);
xc_webrtc.setHandler(xc_webrtc.MessageType.GENERAL, webRtcGeneralEventHandler);
xc_webrtc.setHandler(xc_webrtc.MessageType.REGISTRATION, webRtcRegistrationEventHandler);
xc_webrtc.setHandler(xc_webrtc.MessageType.INCOMING, webRtcIncomingEventHandler);
xc_webrtc.setHandler(xc_webrtc.MessageType.OUTGOING, webRtcOutgoingEventHandler);
Cti messages can be logged in the console if the Cti.debugMsg
variable is set to true
, you can do it directly in
the developer tools console:
Cti.debugMsg=true;
You’ll then get send and received messages in the console log (prefixed by S>>>
and R<<<
respectively):
2016-11-23 14:48:59.180 S>>> {"claz":"web","command":"dial","destination":"111","variables":{}}
2016-11-23 14:48:59.557 R<<< {"msgType":"PhoneStatusUpdate","ctiMessage":{"status":"CALLING"}}
The WebRTC debug can be activated separately by the following method:
xc_webrtc.setDebug(sipml5level, event, handler)
Where:
- sipml5level refers to the SIPml5 library log level string as described on SIPml5 log level documentation,
- event is a boolean value activating event logging (each event is prefixed by
RE<<<
), - handler is a boolean value activating logging of message handler subscription/unsubscription.
Once logged on the sample page, you can init the webRTC through the init button, follow events shown in the webRTC section and send and receive calls. You can terminate a call by the terminate button in the phone section. Direct and attended transfer can be performed using phone section methods. Hold and DTMF features are available via the webRTC API. Current implementation support just one simultaneous call.
Current browsers doesn’t allow media sharing without secure connections - https and wss. The xivoxc_nginx docker image contains the configuration required for loading the sample page over a secure connection using an auto-signed certificate. This certificate is automatically generated by the installation script. It is meant to be used only for test purposes, you should replace it by a signed certificate before switching to production. The sample page is available on the following address: https://MACHINE_IP:8443/sample
Users can connect using login, password and phone number:
var wsurl = "ws://"+server+"/ctichannel?username="+username+"&agentNumber="+phoneNumber+"&password="+password;
Cti.WebSocket.init(wsurl,username,phoneNumber);
An agent can be logged in using Cti.loginAgent(agentPhoneNumber, agentId). For the moment, the phone number used for agent login should be the same as the one used for user login, otherwise you will get many error messages “LoggedInOnAnotherPhone”.
Following cases are handled:
- agent is not logged and requests a login to a known line: the agent is logged in
- agent is not logged and requests a login to an unknown line: an error is raised:
{"Error":"PhoneNumberUnknown"}
- agent is already logged on the requested line: the agent stays logged
- agent is already logged on another line: an error is raised and the agent stays logged (on the number where he was logged before the new request). It’s up to the implementation to handle this case.
{"Error":"LoggedInOnAnotherPhone","phoneNb":"1002","RequestedNb":"1001"}
- agent is not logged and requests a login to a line already used by another agent: the agent takes over the line and the agent previously logged on the line is unlogged
- Cti.MessageType.ERROR
- Cti.MessageType.LOGGEDON
- Cti.MessageType.SHEET
{"msgType":"Sheet","ctiMessage":{"timenow":1425055334,"compressed":true,"serial":"xml",
"payload":{"profile":{"user":{"internal":[{"content":"xivo","name":"ipbxid"},
{"content":"link","name":"where"},{"content":"1425055330.23","name":"uid"},
{"content":"no","name":"focus"},{"content":"1","name":"zip"}],
"sheetQtui":null,"sheetInfo":[{"value":"http://www.google.fr/","name":"popupUrl","order":10,"type":"url"},
{"value":"&folder=1234","name":"folderNumber","order":30,"type":"text"},
{"value":"http://www.google.fr/","name":"popupUrl1","order":20,"type":"url"}],"systrayInfo":[]}}},"channel":"SIP/1k4yj2-00000013"}}
This command deprecates previously used Cti.searchDirectory(pattern).
- Cti.MessageType.DIRECTORYRESULT
Triggered by command Cti.directoryLookUp(pattern). This command deprecates previously used Cti.searchDirectory(pattern).
{ "msgType": "DirectoryResult",
"ctiMessage": {
"entries": [
{ "status": 0, "entry": [ "hawkeye", "pierce", "1002", "0761187406", "false"]},
{ "status": -2, "entry": [ "peter", "pan", "1004", "", "false"]}],
"headers":
[""Firstname", "Lastname", "Number", "Mobile", "Favorite"]}}
- Cti.MessageType.USERSTATUSES : “UsersStatuses”
- Cti.MessageType.USERSTATUSUPDATE : “UserStatusUpdate”,
- Cti.MessageType.USERCONFIGUPDATE : “UserConfigUpdate”,
{"msgType":"UserConfigUpdate",
"ctiMessage":{"userId":9,"dndEnabled":false,"naFwdEnabled":false,"naFwdDestination":"","uncFwdEnabled":false,"uncFwdDestination":"","busyFwdEnabled":false,"busyFwdDestination":"",
"firstName":"Alice","lastName":"Johnson","fullName":"Alice Johnson","mobileNumber":"064574512","agentId":22,"lineIds":[5],"voiceMailId":58,"voiceMailEnabled":true}}
- Cti.MessageType.PHONESTATUSUPDATE
- Cti.MessageType.PHONEEVENT
Phone events are automatically sent when application is connected
Format
{
"msgType":"PhoneEvent",
"ctiMessage":{
"eventType":"EventRinging",
"DN":"1118",
"otherDN":"1058",
"linkedId":"1447670380.34",
"uniqueId":"1447670382.37",
"queueName":"blue",
"userData":{
"XIVO_CONTEXT":"default","XIVO_USERID":"9","XIVO_SRCNUM":"1058","XIVO_DSTNUM":"3000"
}
}
}
fields | Description |
---|---|
Event types |
|
DN | The directory number of the event |
otherDN | Can be calling number of called number |
queueName | Optional, the queue name for inbound acd calls |
UserData | Contains a list of attached data, system data XIVO_ or data attached to the call key beginning by USR_ |
If you use the following preprocess subroutine
[user_data_test]
exten = s,1,Log(DEBUG,**** set user data ****)
same = n,SET(USR_DATA1=hello)
same = n,SET(USR_DATA2=24)
same = n,SET(USR_DATA3=with space)
same = n,Return()
you will get these data in the events. Data can also be attached using the Cti.dial command.
- VOICEMAILSTATUSUPDATE : “VoiceMailStatusUpdate”,
{"msgType":"VoiceMailStatusUpdate","ctiMessage":{"voiceMailId":58,"newMessages":2,"waitingMessages":1,"oldMessages":1}}
- Cti.MessageType.LINKSTATUSUPDATE
- Handler on : Cti.MessageType.QUEUESTATISTICS
The handler is executed when a notification of new statistic values is received. Each message contains one or more counters for one queue. The queue is identified by its queueId. See example below for reference. The queue’s id can be used to retrieve queue’s configuration, see Queue Configuration.
Following counters are available:
- TotalNumberCallsEntered
- TotalNumberCallsAnswered
- PercentageAnsweredBefore15
- TotalNumberCallsAbandonned
- TotalNumberCallsAbandonnedAfter15
- PercentageAbandonnedAfter15
- WaitingCalls
- LongestWaitingTime
- EWT
- AvailableAgents
- TalkingAgents
{
"msgType":"QueueStatistics",
"ctiMessage":{
"queueId":11,"counters":[{"statName":"AvailableAgents","value":0},{"statName":"LoggedAgents","value":0},{"statName":"TalkingAgents","value":0},{"statName":"EWT","value":0}]
}
}
Some messages contain a queueRef with a queue’s name instead of the queueId. This issue should be eliminated in future versions.
{"queueRef":"travels","counters":[{"statName":"TotalNumberCallsAbandonned","value":19}]}
- Handler on: Cti.MessageType.QUEUECALLS
Awaiting calls in a queue. Subscription to the events with : Cti.subscribeToQueueCalls(9) (9 being the queueId). Unsubscription with: Cti.unSubscribeToQueueCalls(9).
{"queueId":9,"calls":[{"position":1,"name":"John Doe","number":"33356782212","queueTime":"2015-07-16T10:40:16.626+02:00"}]}
- QUEUECONFIG : “QueueConfig”,
{"id":8,"context":"default","name":"blue","displayName":"blue sky","number":"3506"}
- QUEUELIST : “QueueList”,
{
"msgType":"QueueList",
"ctiMessage":[
{"id":170,"context":"default","name":"bluesky","displayName":"Bl Record","number":"3012"},
{"id":5,"context":"default","name":"noagent","displayName":"noagent","number":"3050"},
{"id":6,"context":"default","name":"__switchboard_hold","displayName":"Switchboard hold","number":"3005"},
{"id":173,"context":"default","name":"outbound","displayName":"outbound","number":"3099"},
{"id":2,"context":"default","name":"yellow","displayName":"yellow stone","number":"3001"},
{"id":7,"context":"default","name":"green","displayName":"green openerp","number":"3006"},
{"id":3,"context":"default","name":"red","displayName":"red auto polycom","number":"3002"},
{"id":11,"context":"default","name":"pool","displayName":"Ugips Pool","number":"3100"},
{"id":4,"context":"default","name":"__switchboard","displayName":"Switchboard","number":"3004"}
]
}
- Handler on : Cti.MessageType.QUEUEMEMBER
Received when an agent is associated to a queue or a penalty is updated. Penalty is -1 when agent is removed from a queue
{"agentId":19,"queueId":3,"penalty":12}
- Handler on : Cti.MessageType.QUEUEMEMBERLIST
{
"msgType":"QueueMemberList",
"ctiMessage":[
{"agentId":129,"queueId":8,"penalty":2},
{"agentId":139,"queueId":168,"penalty":2},
{"agentId":129,"queueId":10,"penalty":0},
{"agentId":129,"queueId":11,"penalty":0}
]
}
Cti.MessageType.AGENTSTATEEVENT
- AgentLogin
{"name":"AgentLogin","agentId":19,"phoneNb":"1000","since":1423839787,"queues":[8,14,170,4,1],"cause":""}
- AgentReady
{"name":"AgentReady","agentId":19,"phoneNb":"1000","since":0,"queues":[8,14,170,4,1],"cause":"available"}
- AgentOnPause
{"name":"AgentOnPause","agentId":19,"phoneNb":"1000","since":0,"queues":[8,14,170,4,1],"cause":"available"}
- AgentOnWrapup
{"name":"AgentOnWrapup","agentId":19,"phoneNb":"1000","since":2,"queues":[8,14,170,4,1],"cause":"available"}
- AgentRinging
{"name":"AgentRinging","agentId":19,"phoneNb":"1000","since":0,"queues":[8,14,170,4,1],"cause":"available"}
- AgentDialing
{"name":"AgentDialing","agentId":19,"phoneNb":"1000","since":0,"queues":[8,14,170,4,1],"cause":"available"}
- AgentOnCall
{"msgType":"AgentStateEvent","ctiMessage": {"name":"AgentOnCall","agentId":19,"phoneNb":"1000","since":0,"queues":[8,14,170,4,1], "cause":"available","acd":false,"direction":"Incoming","callType":"External","monitorState":"ACTIVE"}}
- AgentLoggedOut
{"name":"AgentLoggedOut","agentId":19,"phoneNb":"1000","since":0,"queues":[8,14,170,4,1],"cause":"available"}
- Cti.MessageType.AGENTERROR
- Cti.MessageType.AGENTDIRECTORY
Triggered by command Cti.getAgentDirectory
{"directory": [
{ "agent":
{"context": "default", "firstName": "bj", "groupId": 1, "id": 8, "lastName": "agent", "number": "2000"},
"agentState": {"agentId": 8, "cause": "", "name": "AgentReady", "phoneNb": "1001", "queues": [1, 2], "since": 2 }}]}
- Cti.MessageType.AGENTCONFIG
Triggered when agent configuration changes
{"id":23,"firstName":"Jack","lastName":"Flash","number":"2501","context":"default"}
- Cti.MessageType.AGENTLIST
Receives agent configuration list in a javascript Array : Command Cti.getList(“agent”);
[
{"id":24,"firstName":"John","lastName":"Waynes","number":"2601","context":"default","groupId":1},
{"id":20,"firstName":"Maricé","lastName":"Saprïtchà","number":"2602","context":"default","groupId":1},
{"id":147,"firstName":"Etienne","lastName":"Burgad","number":"30000","context":"default","groupId":1},
{"id":148,"firstName":"Caroline","lastName":"HERONDE","number":"29000","context":"default","groupId":2},
{"id":149,"firstName":"Eude","lastName":"GARTEL","number":"75000","context":"default","groupId":3},
{"id":22,"firstName":"Alice","lastName":"Johnson","number":"2058","context":"default","groupId":5}
]
- AGENTLISTEN: “AgentListen”,
Receives agent listen stop / start event, received automatically if user is an agent, no needs to subscribe.
{"started":false,"phoneNumber":"1058","agentId":22}
- AGENTGROUPLIST : “AgentGroupList”
Agent group list triggered by command : Cti.getList(“agentgroup”)
[
{"id":1,"name":"default"},
{"id":2,"name":"boats"},
{"id":3,"name":"broum"},
{"id":4,"name":"bingba3nguh"},
{"id":5,"name":"salesexpert"},
{"id":6,"name":"a_very_long_group_name"}
]
Received on subscribe to agent statistics with method Cti.subscribeToAgentStats(), current statistics are received automatically on subscribe.
- AGENTSTATISTICS : “AgentStatistics”
{"id":22,
"statistics":[
{"name":"AgentPausedTotalTime","value":0},
{"name":"AgentWrapupTotalTime","value":0},
{"name":"AgentReadyTotalTime","value":434},
{"name":"LoginDateTime","value":"2015-04-27T08:15:01.081+02:00"},
{"name":"LogoutDateTime","value":"2015-04-27T08:14:49.427+02:00"}
]
}
Get the call history of the logged in user, limited to the last size calls.
Get the call history of the logged in agent, limited to the last size calls.
Get a call history for a queue or a set of queues. You may pass part of a queue name (not display name).
i.e. pass bl if you want to match queue name blue, black and blow
Received when calling the above methods Cti.getAgentCallHistory(size) or Cti.getUserCallHistory(size) .
- CALLHISTORY : “CallHistory”
{
"start":"2014-01-01 08:00:00",
"duration":"00:21:35",
"srcNum":"0115878",
"dstNum":"2547892",
"status":"answered"
}
For queue calls status can be :
- full - full queue
- closed - closed queue
- joinempty - call arrived on empty queue
- leaveempty - exit when queue becomes empty
- divert_ca_ratio -call redirected because the ratio waiting calls/agents was exceeded
- divert_waittime - call redirected because estimated waiting time was exceeded;
- answered - call answered
- abandoned - call abandoned
- timeout - maximum waiting time exceeded
For other calls
- emitted
- missed
- ongoing
Received when calling Callback.getCallbackLists().
- CALLBACKLISTS : “CallbackLists”
{"uuid":"b0849ac0-4f4a-4ed0-9386-53ab2afd94b1",
"name":"Liste de test",
"queueId":1,
"callbacks":[
{"uuid":"a967da84-bc41-4bf4-a4fc-2bcc54e11606",
"listUuid":"b0849ac0-4f4a-4ed0-9386-53ab2afd94b1",
"phoneNumber":"0230210082",
"mobilePhoneNumber":"0789654123",
"firstName":"Alice",
"lastName":"O'Neill",
"company":"YourSociety",
"description":null,
"agentId":null,
"dueDate": "2016-08-01",
"preferredPeriod": {
"default": false,
"name": "Afternoon",
"periodStart": "14:00:00",
"periodEnd": "17:00:00",
"uuid": "d3270038-e20e-498a-af71-3cf69b5cc792"
}}
]}
Received after taking a callback with Callback.takeCallback(uuid).
- CALLBACKTAKEN : “CallbackTaken”
{"uuid":"a967da84-bc41-4bf4-a4fc-2bcc54e11606",
"agentId":2}
Received after starting a callback with Callback.startCallback(uuid, phoneNumber).
- CALLBACKSTARTED : “CallbackStarted”
{"requestUuid":"a967da84-bc41-4bf4-a4fc-2bcc54e11606",
"ticketUuid":"8e82de0f-847a-4606-97bf-bef5a18ea8b0"}
Received after giving to a callback a status different of Callback
.
- CALLBACKCLOTURED : “CallbackClotured”
{"uuid":"a967da84-bc41-4bf4-a4fc-2bcc54e11606"}
Received after releasing a callback with Callback.releaseCallback(uuid).
- CALLBACKRELEASED : “CallbackReleased”
{"uuid":"a967da84-bc41-4bf4-a4fc-2bcc54e11606"}
Received when calling Callback.updateCallbackTicket(uuid, status, description, dueDate, periodUuid) with a new due date or period.
- CALLBACKREQUESTUPDATED : “CallbackRequestUpdated”
{"request":{
"uuid":"a967da84-bc41-4bf4-a4fc-2bcc54e11606",
"listUuid":"b0849ac0-4f4a-4ed0-9386-53ab2afd94b1",
"phoneNumber":"0230210082",
"mobilePhoneNumber":"0789654123",
"firstName":"Alice",
"lastName":"O'Neill",
"company":"YourSociety",
"description":null,
"agentId":null,
"dueDate": "2016-08-01",
"preferredPeriod": {
"default": false,
"name": "Afternoon",
"periodStart": "14:00:00",
"periodEnd": "17:00:00",
"uuid": "d3270038-e20e-498a-af71-3cf69b5cc792"
}
}}
Received when calling Membership.getUserDefaultMembership(userId).
- USERQUEUEDEFAULTMEMBERSHIP: “UserQueueDefaultMembership”
{
"userId":186,
"membership": [
{"queueId":8,"penalty":1},
{"queueId":17,"penalty":0},
{"queueId":18,"penalty":0},
{"queueId":23,"penalty":0}
]
}
Update user status using a cti server configured status name
Log an agent
Un log an agent
Change agent state to pause
Change agent state to ready
Listen to an agent
Set or unset do not disturb, state true or false
Place a call to destination with the provided variables. Variables must take the following form:
{
var1: "value 1",
var2: "value 2"
}
USR_var1 and USR_var2 will be attached to the call and propagated to Phone Events
Place a call from logged user’s mobile number to destination with the provided variables. Variables must take the following form:
{
var1: "value 1",
var2: "value 2"
}
USR_var1 and USR_var2 will be attached to the call and propagated to Phone Events
Originate a call
Hangup a call
Answer a call
Put current call on hold
Tranfert to destination
Start a transfer to a destination
Complete previously started transfer
Cancel a transfer
Start a conference using phone set capabilities
Pause call recording
Note
You can only pause the recording of a call answered by an agent (i.e. a call sent via a Queue towards an Agent).
Unpause call recording
Note
You can only pause the recording of a call answered by an agent (i.e. a call sent via a Queue towards an Agent).
Request a list of configuration objects, objectType can be :
- queue
- agent
- queuemember
Triggers handlers QUEUELIST, AGENTLIST, QUEUEMEMBERLIST. Subscribes to configuration modification changes, handlers QUEUECONFIG, AGENTCONFIG, QUEUEMEMBER can also be called
- agentId (Integer) : id of agent, returned in message Agent Configuration
- queueId (Integer) : id of queue, returned in message Queue Configuration
- penaly (Integer) : positive integer
If agent is not associated to the queue, associates it, otherwise changes the penalty
On success triggers a Queue Member event, does not send anything in case of failure :
{"agentId":<agentId>,"queueId":<queueId>,"penalty":<penalty>}
- agentId (Integer) : id of agent, returned in message Agent Configuration
- queueId (Integer) : id of queue, returned in message Queue Configuration
On success triggers a queue member event with penalty equals to -1, does not send anything in case of failure :
{"agentId":<agentId>,"queueId":<queueId>,"penalty":-1}
Subscribe to agent statistics notification. When called all current statistics are receive, and a notification is received for each updates. Both initial values and updates are transmitted by the Agent Statistics messages.
This command subscribes to the queue statistics notifications. First, all actual statistics values are sent for initialisation and then a notification is sent on each update. Both initial values and updates are transmitted by the QUEUESTATISTICS messages.
Forward on non answer
Unconditionnal forward
Forward on busy
Retrieve the lists of callbacks with teir associated callback requests, and subscribe to callback events.
Take the callback with the given uuid with the logged-in agent.
Release the callback which was previously taken
Launch the previously taken callback with the provided phone number.
Update a callback ticket wih the provided description and status. Allowaed values for status are:
- NoAnswer
- Answered
- Fax
- Callback
dueDate is an optional parameter specifying the new due date using ISO format (“YYYY-MM-DD”).
periodUuid is an optional parameter specifying the new preferred period for the callback.
Initialize the Membership library using the given Cti object.
Request the default membership for the given user id. Warning, the userId is not the same as the agentId.
Set the default membership for the given user id. Warning, the userId is not the same as the agentId. ‘membership’ should be an array of Queue membership like:
[
{"queueId":8,"penalty":1},
{"queueId":17,"penalty":0},
{"queueId":18,"penalty":0},
{"queueId":23,"penalty":0}
]
Set the default membership for the given array of user id. Warning, the userId is not the same as the agentId. ‘userIds’ should be an array of user id like :
[1, 2, 3]
‘membership’ should be an array of Queue membership like:
[
{"queueId":8,"penalty":1},
{"queueId":17,"penalty":0},
{"queueId":18,"penalty":0},
{"queueId":23,"penalty":0}
]
Apply the existing default configuration to a set of users. Warning, the userId is not the same as the agentId. ‘usersIds’ should be an array of userId like:
- ::
- [1, 2, 7, 9]
Once the cti login done, you can init the webRTC component by calling the xc_webrtc.init method.
Init the webRTC connection and register the user’s line.
- name - user’s login to get the line details,
- ssl - if set to true the wss is used,
- websocketPort, ip - port and address for the webRTC websocket connection, when ip is not passed the xivo ip is used,
- remoteAudio - id of the HTML5 audio element for remote audio player, if not passed ‘audio_remote’ is used. The element should look like:
<audio id="audio_remote" autoplay="autoplay"></audio>
Start a webRTC call.
Answer an incoming webRTC call.
Toggle hold on a webrtc call.
Send a DTMF.
Set a handler for eventName from xc_webrtc.MessageType.
Disable ICE server use, only LAN addresses will be used in the SDP.
Set a list of STUN/TURN servers, for example:
[{ url: 'stun:stun.l.google.com:19302'}, { url:'turn:turn.server.org’, username: ‘user’, credential:'myPassword'}]
There are for groups of events:
- general,
- register,
- incoming,
- outgoing.
List of associated events is defined in the xc_webrtc.General, xc_webrtc.Registration, xc_webrtc.Incoming, xc_webrtc.Outgoing. See the xc_webrtc.js on https://gitlab.com/xivoxc/xucserver/blob/master/app/assets/javascripts/xc_webrtc.js. The error state events contains a description in the reason field. Call establishment event contains caller or callee detail. Use the sample page to see some examples.
Rest API¶
http://localhost:$xucport/xuc/api/1.0/$method/$domain/$username/
withHeaders((“Content-Type”, “application/json”))
- $xucport : Xuc port number (default 8090)
- $method : See available methods below
- $domain : Represents a connection site, can be anything
- $username : XiVO client user username
Xuc post JSON formated events on URLeventUrl = "http://localhost:8090/xivo/1.0/event/avencall.com/dropbox/"
configured in /usr/share/xuc/application.conf
Related to a username, phone event is in message payload same structure as javascript Phone Events
{
"username":"alicej",
"message":{
"msgType":"PhoneEvent",
"ctiMessage":{"eventType":"EventDialing","DN":"1058","otherDN":"3000","linkedId":"1447670380.34","uniqueId":"1447670380.34","userData":{"XIVO_USERID":"9"}}}}
{"password" : "password"}
curl -XPOST -d '{"password":"<password>"}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/connect/avencall.com/<username>/
{"state" : [false|true]}
curl -XPOST -d '{"state":false}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/dnd/avencall.com/<username>/
{"number" : "1101"}
curl -XPOST -d '{"number":"<number>"}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/dial/avencall.com/<username>/
Dial command automatically applies filters to the phone number provided to make it valid for Xivo. Especially, it removes invalid characters and handles properly different notations of international country code.
Some countries don’t follow the international standard and actually keep the leading zero after the country code (e.g. Italy). Because of this, if the zero isn’t surrounded by parenthesis, the filter keeps it [1].
[1] | See Redmine ticket #150 |
All forward commands use the above payload
{"state" : [true|false],
"destination" : "1102")
curl -XPOST -d '{"state":true,"destination":"<destnb>"}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/uncForward/avencall.com/<username>/
curl -XPOST -d '{"state":true,"destination":"<destnb>"}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/naForward/avencall.com/<username>/
curl -XPOST -d '{"state":true,"destination":"<destnb>"}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/busyForward/avencall.com/<username>/
Logout un agent
curl -XPOST -d '{"phoneNumber":"<phoneNumber>"}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/agentLogout/
Change state of an agent, pause if ready, ready if on pause
curl -XPOST -d '{"phoneNumber":"<phoneNumber>"}' -H "Content-Type: application/json" http://localhost:8090/xuc/api/1.0/togglePause/
Statistics¶
These real time statistics are calculated nearly in real time from the queue_log table Statistic are reset to 0 at midnight (24h00) can be changed by configuration
name | Description |
---|---|
TotalNumberCallsEntered | Total number of calls entered in a queue |
TotalNumberCallsAbandonned | Total number of calls abandoned in a queue (not answered) |
TotalNumberCallsAbandonnedAfter15 | Total number of calls abandoned after 15 seconds |
TotalNumberCallsAnswered | Total number of calls answered |
TotalNumberCallsAnsweredBefore15 | Total number of calls answered before 15 seconds |
PercentageAnsweredBefore15 | Percentage of calls answered before 15 seconds over total number of calls entered |
PercentageAbandonnedAfter15 | Percentage of calls abandoned after 15 seconds over total number of calls entered |
TotalNumberCallsClosed | Total number or calls received when queue is closed |
TotalNumberCallsTimeout | Total number or calls diverted on queue timeout |
All queue statistics counters are also available for the sliding last hour by adding LastHour to the name .i.e. TotalNumberCallsAbandonnedLastHour
For percentage, it is the mean of the sliding last hour value
Other queue statistics are calculated by xivo cti server
- AvailableAgents
- TalkingAgents
- LongestWaitTime
- WaitingCalls
- EWT
Definition in xivo documentation xivo documentation
name | Description |
---|---|
PausedTime | Total time agent in pause |
WrapupTime | Total time agent in wraup |
ReadyTime | Total time agent ready |
InbCalls | Total number of inbound calls received internal and external |
InbCallTime | Total time for inbound calls received internal and external |
InbAnsCalls | Answered inbound calls received internal and external |
InbUnansCalls | Unanswered inbound calls received internal and external |
InbPercUnansCalls | Percentage of unanswered inbound calls received internal and external |
InbAverCallTime | Average time for inbound calls received internal and external |
OutCalls | Total number of outbound calls received internal and external |
LoginDateTime | Last login date time |
LogoutDateTime | Last logout date time |
Inbound calls, are all calls received by an agent, internal, external or acd calls. Oubound calls are all calls dialed by an agent, internal or external calls.
Agent statistics are calculated internaly on a daily basis and reset to 0 at midnight (default configuration). see javascript api
If some status are configured in xivo cti server with activate pause to all queue = true, additionnal statistics computing the total time in not ready with this status are calculated. This statistics name is equal to the presence name configuration in XiVO.
Technical structure of XiVO-CC¶
The reporting is composed of four packages: pack-reporting, xivo-full-stats, xivo-reporting-db and xivo-db replication.
These packages will feed the tables of the xivo_stats database:
- xivo-db-replication feeds the tables cel and queue_log in real time, and the configuration tables (dialaction, linefeatures, etc...) every 5 minutes
- xivo-full-stats feeds in real time tha tables call_on_queue, call_data, stat_queue_periodic, stat_agent_periodic and agent_position
- xivo-reporting-db and pack-reporting work together to feed the tables stat_queue_specific, stat_agent_queue_specific and stat_agent_specific every 15 minutes
Troubleshooting¶
In this section, we give some troubleshooting hints. Continue by choosing the component.
Check installation¶
In order for the XivoCC components to be fully functional, some customizations need to be done on the XiVO CC and the XiVO PBX.
This page can heklp to check that all the correct customization have been done by the installation package.
For the rest of this page we well make the following assumptions: - XiVO PBX has the IP 192.168.0.1 - XiVO CC has the IP 192.168.0.2
- the OS must be Debian 8 (jessie), 64 bit,
- Docker must be installed,
- Docker-compose must be installed,
- the XiVO PBX must be reachable on the network.
The XiVO CC server and the XiVO server must be synchronized to the same source NTP source.
A file /etc/logrotate.d/docker-container
must be present which should log rotate files
/var/lib/docker/containers/*/*.log
You can test it with logrotate -fv /etc/logrotate.d/docker-container. You should get some output and a new log file with suffix [CONTAINER ID]-json.log.1 should be created. This file is compressed in next rotation cycle.
An alias for docker-compose must be present like:
alias dcomp='docker-compose -p xivocc -f /etc/docker/compose/docker-xivocc.yml'
The version of the docker images in the file
/etc/docker/compose/docker-xivocc.yml
must be2016.03.latest
:... xivo_stats: image: xivoxc/xivo-full-stats:2016.03.latest ... xuc: image: xivoxc/xuc:2016.03.latest ...
The list of the services launched should look like :
# dcomp ps
Name Command State Ports
---------------------------------------------------------------------------------------------------------------------
xivocc_config_mgt_1 bin/config-mgt-docker Up 0.0.0.0:9100->9000/tcp
xivocc_elasticsearch_1 /docker-entrypoint.sh elas ... Up 0.0.0.0:9200->9200/tcp, 0.0.0.0:9300->9300/tcp
xivocc_fingerboard_1 /bin/sh -c /usr/bin/tail - ... Up
xivocc_kibana_volumes_1 /bin/sh -c /usr/bin/tail - ... Up
xivocc_nginx_1 nginx -g daemon off; Up 443/tcp, 0.0.0.0:80->80/tcp
xivocc_pack_reporting_1 /bin/sh -c echo ... Up
xivocc_pgxivocc_1 /docker-entrypoint.sh postgres Up 0.0.0.0:5443->5432/tcp
xivocc_postgresvols_1 /bin/bash Exit 0
xivocc_recording_server_1 bin/recording-server-docker Up 0.0.0.0:9400->9000/tcp
xivocc_reporting_rsync_1 /usr/local/sbin/run-rsync.sh Up 0.0.0.0:873->873/tcp
xivocc_spagobi_1 /bin/sh -c /root/start.sh Up 0.0.0.0:9500->8080/tcp
xivocc_timezone_1 /bin/bash Exit 0
xivocc_xivo_replic_1 /usr/local/bin/start.sh /o ... Up
xivocc_xivo_stats_1 /usr/local/bin/start.sh /o ... Up
xivocc_xivocclogs_1 /bin/bash Exit 0
xivocc_xuc_1 bin/xuc_docker Up 0.0.0.0:8090->9000/tcp
xivocc_xucmgt_1 bin/xucmgt_docker Up 0.0.0.0:8070->9000/tcp
Postgresql has to be configured to listen on all interfaces. See listen_addresses in file
/etc/postgresql/9.4/main/postgresql.conf
.Connection from the XiVO CC for user asterisk must be authorized. See file
/etc/postgresql/9.1/main/pg_hba.conf
which must contain a line:host asterisk all 192.168.0.2/32 md5
A user stats must exists. Use command
\dg
in psql.
A xuc user must be configured in the file
/etc/asterisk/manager.d/02-xivocc.conf
The command:
asterisk -rx "manager show user xuc"
must show the user.
The correct events must be activated in the file /etc/asterisk/cel.conf
:
[general]
enable = yes
apps = dial,park,queue
events = APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT,USER_DEFINED,LINKEDID_END,HOLD,UNHOLD,BLINDTRANSFER,ATTENDEDTRANSFER
[manager]
enabled = yes
In
a user the must be created with the following parameters:- CTI login : xuc
- CTI password : 0000
- Profile supervisor
In
a user must be created with the following parameters :- Login : xivows
- Password : xivows
- Host : 192.168.0.2
In Multiqueues call stats sharing
is checked.
Verify that the phone configuration where customized as detailed in Required configuration for phone integration.
The package xivo-recording
must be installed.
If you want to use call recording filtering, the package call-recording-filtering
must be installed too.
Xuc et Xuc_mgt - applications web ccmanager, agent et assistant¶
XUC overview page available at @XUC_IP:PORT, usually @SERVER_IP:8090. You have to check if the “Internal configuration cache database” contains agents, queues etc.
XUC sample page available at @XUC_IP:PORT/sample, usually @SERVER_IP:8090/sample. You can use this page to check user login and other API functions. CCManager, agent and assistant web use functions available on the sample page.
Desktop Assistant¶
If needed, Desktop Assistant can be started with -d
option to enable debug menu.
Application Configuration (xuc_rigths)¶
Recording¶
SpagoBI¶
Kibana¶
XiVO Centralized Interface¶
The XiVO Centralized Interface (XCI) allows to manage several XiVO servers through a unique web interface. Thanks to this interface, it becomes possible to quickly add users that are automatically routed across servers. This documentation will describe the installation process of the interface, how to use the web interface and the REST API it exposes.
Installation¶
The XiVO Centralized Interface (XCI) requires :
- A Linux server with PostgreSQL, Docker and Docker-Compose installed
- Some XiVOs to manage !
An installation script is provided to execute all the installations tasks. To run it, execute the following command :
curl https://gitlab.com/xivo-utils/icdu-packaging/raw/master/install-icdu.sh | sudo bash
It will ask you a passphrase for generating an SSH key.
The configuration files are located in /etc/docker
.
Optionally, you can set a bash alias for conveniently run XCI :
alias dcomp='docker-compose -p icdu -f /etc/docker/compose/icdu.yml'
Then simply :
dcomp up -d
XCI should now be accessible through http://my-server-ip:9001
The configuration files and the Docker-Compose files are available in a specific Git repository.
XCI stores some data in a PostgreSQL database. By default, application.conf
is configured to connect to a local database named icx
with the username icx
and password icx
. You can change these parameters if you wish. We will use the default parameters in this documentation.
First, we need to install PostgresSQL extensions to use UUID functions :
sudo apt-get install postgresql-contrib
We can now create the user and the database associated :
sudo -u postgres psql -c "CREATE USER icx WITH PASSWORD 'icx'"
sudo -u postgres psql -c "CREATE DATABASE icx WITH OWNER icx"
We then have to enable UUID extension on the icx
database. Connect as root
on the icx
database :
sudo -u postgres psql icx -c 'CREATE EXTENSION IF NOT EXISTS "uuid-ossp";'
It is possible that PostgreSQL complains when you’re trying to connect. The solution is to modify the pg_hba.conf
(in Debian, located in /etc/postgresql/X.X/main
) and add the following line at the end :
local all all trust
In order to let XCI communicate with the various XiVOs, an SSH key is used. Generate one using the following command :
ssh-keygen -t rsa -f /etc/docker/interface-centralisee/ssh_key
Web interface¶
The XiVO Centralized Interface (XCI) is managed through a web interface. In the following sections, we will highlight the main features of the system.
XCI uses a few concepts that are important to understand in order to use the interface correctly.
- XiVO
- The XiVOs servers that are managed by XCI. XCI will automatically retrieve the entities and the users from them and apply the configuration to them.
- Entity
- Entities, also called Contexts, are the parts of the dialplan. Users are attached to them.
- Line template
- Line templates are used to quickly create users : they define a few default options (ringing time, voice mail, etc.) that will be applied to the new user. A line template is required to create a user.
- User
- Actual users that are associated with a phone number
- Administrators
- Users that are able to connect to the XCI and manage the XiVOs.

The dashboard provides you some insights about your XiVO systems.
The left sidebar, displayed in every page of the application, gives you access to the various actions you can perform. The list of the configured XiVOs and their entities is shown to give a quick access to the one you want to manage.

This page allows you to add a new XiVO that will be managed by XCI. The first step is to add the displayed SSH key to the authorized keys of your XiVO server. This will allow XCI to connect and configure the XiVO server. You could do this kind of command :
echo 'ssh-rsa TheVeryLongSSHKeyYouCopied toto@someserver' | ssh root@xivoIp 'cat >> .ssh/authorized_keys'
Then, you have to provide the following informations :
- Name : name of the XiVO server that will be displayed in XCI
- Hostname : hostname or IP address of the XiVO server
You then have two options :
- Create the XiVO and configure it now : XCI will save the informations, try to connect to the XiVO server and perform the configuration. XiVO services will be unavailable during the operation.
Warning
The configuration takes a while. Relax, go drink a coffee, XCI is doing the legwork for you :)
- Create the XiVO without configuring it : XCI will only save the informations.

On the sidebar, each XiVO has its own View XiVO link. This page allows you to :
Add a new entity to this XiVO by clicking on the green button
- See the entities associated to this XiVO and perform some operations to them :
- Edit one by clicking on the yellow button with the wrench icon
- Delete one by clicking on the red button with the trash icon

This page allows you to add a new entity to a XiVO. You have to provide the following informations :
Name : name that will be used by the XiVO server
Display name : name that will be displayed on XCI
Caller ID : phone number that will be displayed on outgoing call from this entity
Intervals : ranges of phone numbers that will be available to this entity. For each one, provide :
- Start
- End
The system will return an error if the intervals overlap with other entities

On the sidebar, each entity has its own link. This page allows you to :
Add a new user to this entity by clicking on the green button
Edit the entity by clicking on the yellow button with the wrench icon
- See the users associated to this entity and perform some operations to them :
- Edit one by clicking on the yellow button with the wrench icon
- Delete one by clicking on the red button with the trash icon. At first click, the icon turns into a question mark. You have 5 seconds to click again to launch user deletion. This process prevents you from accidentally delete users.

This page allow you to modify an entity. Please refer to the Create entity section for fields details.

On the sidebar, Line template has its own link. This page allows you to :
Add a new line template by clicking on the green button
- See all the line templates and perform some operations to them :
- Edit one by clicking on the yellow button with the wrench icon
- Delete one by clicking on the red button with the trash icon

This page allows you to add a new line template. You have to provide the following informations :
Name : name that will be be displayed on XCI
XiVO : select the XiVOs for which this template will be available
Entity : select the entities for which this template will be available. Only entities of the selected XiVOs are displayed
SIP peer name : Auto or Model
Ringing time : number of seconds before incoming call is rejected
Routed :
- The text field allows you to provide the SDA prefix to call the phone
- Uncheck the checkbox if you don’t want the phone to be called from the outside
Outgoing caller id : specify what number is displayed on outgoing call. Possible values are :
- External number prefix
- Anonymous
- Customized : a text field appears to provide the custom number
Voicemail :
Activate voicemail : enable or not the voicemail
Voicemail number : specify what number is used to call the voice mail. Possible values are :
- Short line number : use the default short number
- Customized : a text field appear to provide the custom number
Voice to mail : whether or not to send an email when a new message is left

This page allows you to modify a template. Please refer to the Create template section for fields details.

This page allows you to add a new user to an entity. You have to provide the following informations :
- Template : line template to use as a template to create the user. The main options of the template are displayed below
- First name
- Last name
- Internal number : number that will be used to internally call the user. Only the available numbers are displayed
- CTI credentials : provide a login and a password to allow the user to connect through CTI interfaces

This page allows you to modify a user. Please refer to the Create user section for fields details.

On the sidebar, Administrators has its own link. This page allows you to :
Add a new administrator by clicking on the green button
- See all the administrators and perform some operations to them :
- Edit one by clicking on the yellow button with the wrench icon
- Delete one by clicking on the red button with the trash icon

This page allows you to add a new administrator. You have to provide the following informations :
- Login : login used by the administrator to connect to XCI
- Name : name that will be displayed on XCI
- LDAP : if checked, the LDAP authentication configured in
application.conf
will be used - Password : password used by the administrator to connect to XCI. Shown only if LDAP disabled
- Superadmin : whether or not this administrator is a super-administrator. Super-administrators can manage everything in XCI
- Entities : select the entities this administrator will be able to manage Shown only if Superadmin disabled

This page allows you to modify an administrator. Please refer to the Create administrator section for fields details.
REST API¶
The XiVO Centralized Interface (XCI) exposes some REST API that you can use to integrate with your tools.
http://$my-server-ip:$xciport/api/1.0/$method
withHeaders((“Content-Type”, “application/json”))
- $xciport : XCI port number (default 9001)
- $method : See available methods below
A login request is required before subsequent API calls in order to get a session cookie.
POST /api/1.0/login
Payload parameters :
login
(String)- Login to connect with
password
(String)- Password corresponding to the login
The server will return a cookie and you will be able to do other API calls. Example with CURL :
curl 'http://localhost:9000/api/1.0/login' -H 'Content-Type: application/json' -c 'xci-cookie' --data-binary '{"login":"admin","password":"superpass"}'
curl 'http://localhost:9000/api/1.0/xivo' -H 'Content-Type: application/json' -b 'xci-cookie'
The following methods allow you to operate on the XiVOs managed by XCI.
List all the XiVOs configured on XCI.
GET /api/1.0/xivo
{
"items": [
{
"id": 1,
"uuid": "8f159082-4b25-48b3-afec-1873491a60be",
"name": "xivo-220",
"host": "192.168.29.220",
"remainingSlots": 664
},
{
"id": 2,
"uuid": "15585b75-1d75-45b1-8678-520d1210ec59",
"name": "xivo-221",
"host": "192.168.29.221",
"remainingSlots": 280
}
]
}
Get a XiVO by its id
.
GET /api/1.0/xivo/$id
{
"id": 1,
"uuid": "8f159082-4b25-48b3-afec-1873491a60be",
"name": "xivo-220",
"host": "192.168.29.220",
"remainingSlots": 664
}
Create a new XiVO.
POST /api/1.0/xivo
Payload parameters :
name
(String)- Display name of the XiVO
host
(String)- Hostname or IP address of the XiVO
configure
(Boolean)- If set to
true
, XCI will immediately make the necessary configurations on the XiVO. If set tofalse
, it will only be added to XCI but not configured.
GET /api/1.0/xivo/synchronize_config_files
The following methods allow you to operate on the entities made available by the XiVOS.
List all the entities available.
GET /api/1.0/entities
{
"items": [
{
"id": 17,
"combinedId": "default@15585b75-1d75-45b1-8678-520d1210ec59",
"name": "default",
"displayName": "default",
"xivo": {
"id": 2,
"uuid": "15585b75-1d75-45b1-8678-520d1210ec59",
"name": "xivo-221",
"host": "192.168.29.221",
"remainingSlots": 280
},
"intervals": [
{
"start": "1700",
"end": "1799"
},
{
"start": "1961",
"end": ""
},
{
"start": "2600",
"end": "2799"
}
],
"presentedNumber": "inbNo"
},
{
"id": 22,
"combinedId": "default_analogique@15585b75-1d75-45b1-8678-520d1210ec59",
"name": "default_analogique",
"displayName": "default_analogique",
"xivo": {
"id": 2,
"uuid": "15585b75-1d75-45b1-8678-520d1210ec59",
"name": "xivo-221",
"host": "192.168.29.221",
"remainingSlots": 280
},
"intervals": [
{
"start": "3990000",
"end": "3999999"
},
{
"start": "39990000",
"end": "39999999"
}
],
"presentedNumber": "inbNo"
}
]
}
Get an entity by its combinedId
.
GET /api/1.0/entities/$combinedId
{
"id": 22,
"combinedId": "default_analogique@15585b75-1d75-45b1-8678-520d1210ec59",
"name": "default_analogique",
"displayName": "default_analogique",
"xivo": {
"id": 2,
"uuid": "15585b75-1d75-45b1-8678-520d1210ec59",
"name": "xivo-221",
"host": "192.168.29.221",
"remainingSlots": 280
},
"intervals": [
{
"start": "3990000",
"end": "3999999"
},
{
"start": "39990000",
"end": "39999999"
}
],
"presentedNumber": "inbNo"
}
Create a new entity.
POST /api/1.0/entities
Payload parameters :
name
(String)- Name of the entity
displayName
(String)- Displayed name of the entity
xivoId
(Integer)- Id of the XiVO the entity will be attached to
intervals
(Array)Intervals of numbers this entity will support
start
(String)- Starting number of the interval
end
(String)- Ending number of the interval
presentedNumber
(String)- Number to show on outgoing calls
List users attached to an entity.
GET /api/1.0/entities/$combinedId/users
{
"items": [
{
"id": 559,
"entity": {
"id": 22,
"combinedId": "default_analogique@15585b75-1d75-45b1-8678-520d1210ec59",
"name": "default_analogique",
"displayName": "default_analogique",
"xivo": {
"id": 2,
"uuid": "15585b75-1d75-45b1-8678-520d1210ec59",
"name": "xivo-221",
"host": "192.168.29.221",
"remainingSlots": 280
},
"intervals": [
{
"start": "3990000",
"end": "3999999"
},
{
"start": "39990000",
"end": "39999999"
}
],
"presentedNumber": "inbNo"
},
"firstName": "Sous sol Logistique",
"lastName": "CLF 88:40 P3",
"internalNumber": "6260",
"externalNumber": "\"Sous sol Logistique CLF 88:40 P3\"",
"mail": null,
"ctiLogin": null,
"ctiPassword": null,
"provisioningNumber": "114133"
}
]
}
List available numbers for an entity
GET /api/1.0/entities/$combinedId/available_numbers
{
"items": [
"3990000",
"3990001",
"3990002",
"3990003",
"3990004"
]
}
The following methods allow you to operate on the users made available by the XiVOS.
Get a user by its id
.
GET /api/1.0/users/$id
{
"id": 559,
"entity": {
"id": 22,
"combinedId": "default_analogique@15585b75-1d75-45b1-8678-520d1210ec59",
"name": "default_analogique",
"displayName": "default_analogique",
"xivo": {
"id": 2,
"uuid": "15585b75-1d75-45b1-8678-520d1210ec59",
"name": "xivo-221",
"host": "192.168.29.221",
"remainingSlots": 280
},
"intervals": [
{
"start": "3990000",
"end": "3999999"
},
{
"start": "39990000",
"end": "39999999"
}
],
"presentedNumber": "inbNo"
},
"firstName": "Sous sol Logistique",
"lastName": "CLF 88:40 P3",
"internalNumber": "6260",
"externalNumber": null,
"mail": null,
"ctiLogin": null,
"ctiPassword": null,
"provisioningNumber": "114133"
}
Create a new user.
POST /api/1.0/users
Payload parameters :
entityCId
(String)- Entity combinedId the user will be attached to
templateId
(Integer)- Line template to apply to the user
firstName
(String)- First name of the user
lastName
(String)- Last name of the user
internalNumber
(String)- Internal phone number of the user
ctiLogin
(String) Optional- CTI login of the user
ctiPassword
(String) Optional- CTI password of the user
The following methods allow you to operate on the line templates used to create users.
List all the templates available.
GET /api/1.0/templates
[
{
"id": 1,
"name": "Modèle 220",
"peerSipName": "auto",
"routedInbound": false,
"callerIdMode": "incomingNo",
"ringingTime": 30,
"voiceMailEnabled": false,
"voiceMailNumberMode": "short_number",
"xivos": [
1
],
"entities": [
"default@8f159082-4b25-48b3-afec-1873491a60be"
]
}
]
Get a template by its id
.
GET /api/1.0/templates/$id
{
"id": 1,
"name": "Modèle 220",
"peerSipName": "auto",
"routedInbound": false,
"callerIdMode": "incomingNo",
"ringingTime": 30,
"voiceMailEnabled": false,
"voiceMailNumberMode": "short_number",
"xivos": [
1
],
"entities": [
"default@8f159082-4b25-48b3-afec-1873491a60be"
]
}
Create a new template.
POST /api/1.0/templates
Payload parameters :
name
(String)- Name of the template
xivos
(Array of Integer)- List of XiVOs ids the template will be available to
entities
(Array of String)- List of entities combinedIds the template will be available to
peerSipName
(String)- Possible values are
auto
ormodel
ringingTime
(Integer)- Number of seconds before incoming call is rejected
routedInbound
(Boolean)- Whether or not the phone can be called from the outside
routedInboundPrefix
(String) Compulsory ifroutedInbound
istrue
- SDA prefix to call the phone
callerIdMode
(String)- Option specifying what number is displayed on outgoing call. Possible values are :
incomingNo
: use the SDA prefixanonymous
: masked callcustom
: a custom number
customCallerId
(String) Compulsory ifcallerIdMode
iscustom
- Custom number to display on outgoing call
voiceMailEnabled
(Boolean)- Whether or not to enable the voice mail
voiceMailNumberMode
(Boolean)- Option specifying what number is used to call the voice mail. Possible values are :
short_number
: use the default short numbercustom
: a custom number
voiceMailCustomNumber
(String) Compulsory ifvoiceMailNumberMode
iscustom
- Custom number to call the voice mail
voiceMailSendEmail
(Boolean)- Whether or not to send an email when a new message is left
The following methods allow you to operate on the administrators of the XCI.
List all the administrators present.
GET /api/1.0/administrators
{
"items": [
{
"id": 1,
"login": "admin",
"name": "",
"password": "+\/\/rIncoyp\/Ai\/8l3xSEeSY+P+x4uNle7cHkL6rpPS3ucgr2EAJIqnQbsIpSGwHj",
"superAdmin": true,
"ldap": false,
"entities": [
]
}
]
}
Get an administrator by its id
.
GET /api/1.0/administrators/$id
{
"id": 1,
"login": "admin",
"name": "",
"password": "+\/\/rIncoyp\/Ai\/8l3xSEeSY+P+x4uNle7cHkL6rpPS3ucgr2EAJIqnQbsIpSGwHj",
"superAdmin": true,
"ldap": false,
"entities": [
]
}
Create a new administrator.
POST /api/1.0/administrators
Payload parameters :
login
(String)- Login of the administrator
name
(String)- Displayed name of the administrator
ldap
(Boolean)- Whether or not to use the LDAP authentication configured in
application.conf
password
(String) Compulsory ifldap
isfalse
- Password used by the administrator to login
superAdmin
(Boolean)- Whether or not this administrator is a super-administrator. Super-administrators can manage everything in XCI.
entityIds
(Array of Integer)- List of entities this administrator has the rights to manage
Edit an administrator. See Create administrator for fields details.
PUT /api/1.0/administrators/$id
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
from urllib.parse import urlencode
from urllib.request import Request, urlopen
import json, sys
class XCIApiExample:
base_url = None
cookie = None
def __init__(self, base_url, login, password):
self.base_url = base_url
self.make_login(login, password)
def make_login(self, login, password):
data = {"login": login, "password": password}
response = self.make_post_request("/login", data)
self.cookie = response.info()["Set-Cookie"]
def get_entities(self):
response = self.make_get_request("/entities")
return self.handle_json_response(response)
def get_available_numbers(self, entity):
response = self.make_get_request("/entities/" + entity["combinedId"] + "/available_numbers")
return self.handle_json_response(response)
def create_line_template(self, data):
self.make_post_request("/templates", data)
def get_line_templates(self):
response = self.make_get_request("/templates")
return self.handle_json_response(response)
def create_user(self, data):
self.make_post_request("/users", data)
def make_get_request(self, method):
request = Request(self.base_url + method, headers = {"Cookie": self.cookie})
response = urlopen(request)
return response
def make_post_request(self, method, data):
header = {"Content-Type": "application/json", "Cookie": self.cookie if self.cookie else ""}
request = Request(self.base_url + method, json.dumps(data).encode(), header)
response = urlopen(request)
return response
def handle_json_response(self, response):
return json.loads(response.read().decode())
# Initialize API
api_example = XCIApiExample("http://192.168.29.103:9001/api/1.0", "admin", "superpass")
# Get an entity and its XiVO
entities = api_example.get_entities()["items"]
if (len(entities) == 0):
sys.exit("There isn't any XiVO configured yet or they don't have any entity !")
else:
entity = entities[1]
xivo = entity["xivo"]
print("Selected entity \"%s\" in XiVO \"%s\""%(entity["name"], xivo["name"]))
# Create a line template
template_data = {
"name": "My line template",
"xivos": [xivo["id"]],
"entities": [entity["combinedId"]],
"peerSipName": "auto",
"ringingTime": 30,
"routedInbound": False,
"callerIdMode": "anonymous",
"voiceMailEnabled": False
}
api_example.create_line_template(template_data)
line_template = api_example.get_line_templates()[0]
print("New line template created")
# Create a user
user_data = {
"entityCId": entity["combinedId"],
"templateId": line_template["id"],
"firstName": "Alice",
"lastName": "In Wonderland",
"internalNumber": api_example.get_available_numbers(entity)["items"][0]
}
api_example.create_user(user_data)
print("New user created")