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 Services ‣ IPBX ‣ IPBX configuration ‣ Configuration files add a new file containing the following dialplan:

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.


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.

  1. In the Web Interface and under Services ‣ IPBX ‣ IPBX configuration ‣ Configuration files add a new file named fax-detection.conf containing the following dialplan:

    ;; Fax Detection
    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}))
  2. In the file /etc/xivo/asterisk/xivo_globals.conf set the global user subroutine to pre-user-global-faxdetection : this subroutine will be executed each time a user is called:

    XIVO_PRESUBR_GLOBAL_USER = pre-user-global-faxdetection
  3. 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
                +------+                            +-----+


  | 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.

  1. Follow the Berofos general configuration (firmware, IP, login/password) described in the the Berofos Installation and Configuration page.

  2. When done, apply these specific parameters to the berofos:

    bnfos --set scenario=1   -h -u admin:berofos
    bnfos --set mode=1       -h -u admin:berofos
    bnfos --set modedef=1    -h -u admin:berofos
    bnfos --set wdog=1       -h -u admin:berofos
    bnfos --set wdogdef=1    -h -u admin:berofos
    bnfos --set wdogitime=60 -h -u admin:berofos
  3. Add the following script /usr/local/sbin/berofos-workaround:

    # Script workaround for berofos integration with a XiVO in front of PABX
    res=$(/usr/sbin/service asterisk status)
    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
        /usr/bin/logger "$0 - Asterisk is not running"
  4. Add execution rights to script:

    chmod +x /usr/local/sbin/berofos-workaround
  5. Create a cron to launch the script every minutes /etc/cron.d/berofos-cron-workaround:

    # Workaround to berofos integration
    */1 * * * * root /usr/local/sbin/berofos-workaround

Upgrading from XiVO 1.2.3

  1. There is an issue with xivo-libsccp and pf-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
    Errors were encountered while processing:
    E: Sub-process /usr/bin/dpkg returned an error code (1)
  2. You have to remove /var/lib/dpkg/info/xivo-libsccp.conffiles:

    rm /var/lib/dpkg/info/xivo-libsccp.conffiles
  3. You have to edit /var/lib/dpkg/info/xivo-libsccp.list and remove the following line:

  4. and remove /etc/asterisk/sccp.conf:

    rm /etc/asterisk/sccp.conf
  5. 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:


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:

  1. 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
    systemctl daemon-reload
  2. 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


Procedure was removed since bug was fixed in asterisk version shipped in 2017.LTS1 (2017.03)

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 system

  • locale to display information about the current locale of your shell

  • grep ^lc_ /etc/postgresql/9.4/main/postgresql.conf to see the locale configuration of your database cluster

  • sudo -u postgres psql -l to see the locale of your databases

  • the /etc/locale.gen file and the associated locale-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


You can use any locale with XiVO as long as it uses an UTF-8 encoding.

Database cluster is not starting

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 running locale-gen

  • or modify the /etc/postgresql/9.4/main/postgresql.conf file to set the various lc_* options to a locale that is available on your system

Once this is done, restart your database cluster.

Can’t connect to the database

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:

Error during the upgrade

Then you are mostly in one of the cases described above. Check your log file.

Error while restoring a database backup

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, running locale-gen and restarting your database cluster

  • or 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

Error during master-slave replication

Then the slave database is most likely not using an UTF-8 encoding. You’ll need to recreate the database using a different locale

Changing the locale (LC_COLLATE and LC_CTYPE) of the database

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 the asterisk 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'
ALTER DATABASE asterisk_newlocale RENAME TO asterisk;
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


  • http.conf - asterisk’s webserver must accept connection from outside, the listen address must be updated, for the sake of simplicity let’s use, you can also pick an address of one of the network interfaces:

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
; RTP start and RTP end configure start and end addresses
; Defaults are rtpstart=5000 and rtpend=31000
; Whether to enable or disable UDP checksums on RTP traffic
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)

The configuration is reloaded by module reload

  • 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: 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

Call Permission and Transfers

Some Call Permission issues may occur in case of call transfers. For example:

  • Given user U1 with call permissions C1,

  • Given user U2 with another call permissions set C2,

  • When U1 calls U2 and transfers it somewhere

  • Then, depending on the type of transfer it will take the call permissions C1 or C2.

Current behavior is descrbided in bug 1944.