Below is a list of New Features and Behavior Changes compared to the previous LTS version, Helios (2022.05).
- UC Assistant
- UC App design was updated
- Can listen to the ringtones before selecting one
- Can make a video call from the chat conversation - see Instant Messaging
- Added notification after connecting to the mobile app for the first time, and add mobile-web switch - see Using web and mobile applications together
- Added unpair mobile app
- Make a video call from the chat conversation
- Mobile Application
- New Mobile Application
- Meetingroom sharing link is shorter in order (like
https://<edge>/meet?id=xxxx-yyyy) to be able to visually check it / compare it
- Cannot accept a meeting room invitation if I have already one meeting room ongoing.
- Added a welcome page before entering an external meeting room to choose the camera and microphone.
- Users can select a custom background or blur theirs during the meeting - experimental, see Background Selection
- Asterisk version upgraded to 18.10.1
- SIP channel driver was changed to
res_pjsip- see Asterisk chan_sip to pjsip Migration Guide
- Personal Contact API
- Personal Contact API now validates incoming data. One of (firstname, lastname) and one of (number, mobile, fax) must be filled.
- Meeting room sharing link have a new shortened format
https://<edge>/meet?id=xxxx-yyyyThis link is permanent and doesn’t change if you edit your meetingroom (change name / pin…) Old links are deprecated but still usable.
- Docker update
- Docker-ce version bumped from 5:19.03.13 to 5:20.10.13
- Docker-compose version bumped from 1.27.4 to 1.29.2
- Debian 11: all XiVO components now needs a Debian 11 (Bullseye) base to work.
Asterisk version upgraded to 18.10.1
SIP channel driver was changed to
res_pjsip- see Asterisk chan_sip to pjsip Migration Guide
ctid was moved to a container. Logs will now be found in /var/log/xivo-ctid/ dir instead of /var/log
You MUST review your custom dialplan and replace the usage of the
Answerapplication by calling the xivo-pickup subroutine. Also, for dialplan that ‘ends’ in an application (like Playback or Voicemail …), you should insert a call to the xivo-pickup subroutine before the application :
same = n,Answer()
same = n,Gosub(xivo-pickup,s,1)
same = n,... same = n,Set... same = n,Voicemail()
same = n,Set... same = n,Gosub(xivo-pickup,s,1) same = n,Voicemail()
- A subset of the phone device plugins were moved to a python3 compatible repo - please check: Python3 migration: Phone Device Plugins
This release deprecates:
- LTS Callisto (2019.05): This version is no longer supported. No bug fixes, no security update will be provided for this release.
- Python2: as part of Debian 11 upgrade, all xivo services were moved to python3.
xivo-authLDAP backend was removed
Manual steps for LTS upgrade
Don’t forget to read carefully the specific steps to upgrade from another LTS version
Generic upgrade procedure
Then, follow the generic upgrade procedures:
Izar Bugfixes Versions¶
Components version table¶
Table listing the current version of the components.
Consult the 2022.05.21 (Izar.21) Roadmap.
- #7131 - Desktop Assistant does not always starts (depending the PC load)
Consult the 2022.05.20 (Izar.20) Roadmap.
- #6869 - Update asterisk to 18.17.1 - deb11 - for Izar - to mitigate WS massive closing
- #6873 - Asterisk 18.18.1 on the debian11_ast18 branch (for Izar & Jabbah)
- #6628 - [Edge] CCagent] Dissuasion configuration through Edge with CC Agent is blocked
- #6688 - Window is not focused when launching application already running
- #6954 - Support signing of source code on Izar
- #7093 - Unoffered events are not generated in xc_queue_call
- #6769 - Meetingroom : Add variable ENABLE_BACKGROUND_SELECTION in .env documentation
- #6651 - History requests are sent multiple times to xuc when a call is ongoing
- #6644 - Missing small NaN fix in audio quality for izar and jabbah
- #6550 - The phone status updates are not send to opened Web socket
- #6820 - Trying to connect to multiple unreachable mds leads xuc to stop working
- #6720 - logrotate seems not working for xivo-confgend on main or MDS
- #6632 - [Doc] Complete network flows pre-requisite table with XiVO / Edge-Turn according to diagram
Consult the 2022.05.19 (Izar.19) Roadmap.
- #6601 - CCagent do not resize to correct size at startup of desktop assistant
- #5950 - When in a meeting room, if I receive another meeting room invitation I can’t quit my current MR to join the new invitation
#6577 - Packet loss warning does not work with Chrome 110 and higher
#6589 - Opus is not activated for WebRTC users for Incoming call IZAR
Behavior change We now ensure Opus codec is selected for calls towards WebRTC users. Given A a user. Given W a WebRTC user. When A calls W, we ensure
opuscodec will be used on W’s channel (whatever the codec offered by A).
This implies potentially more transcoding load on asterisk. Previously A’s codec was preferred (presumably G.711 A-Law) and then implying less transcoding by asterisk.
#6591 - Ice negociation timeout not working in scenarios where two calls are presented in quick succession
- #6518 - Fix the memory leak when a user is connecting to xuc
- #6387 - [Doc] UCAddon - wrong debian version in doc
- #6382 - Fix python3 for the greffon SPAXX-ATAXX
- #6469 - [S] - Edge - Call Qualification export API is accessible without authentication
Consult the 2022.05.17 (Izar.17) Roadmap.
- #5417 - OBS - BTIP - SIP Trunk validation - In SDP owner is changed upon re-INVITE
- #6110 - Asterisk crash in res_rtp_asterisk.c:3165
- #6228 - Invalid asterisk patch xivo_early_cli_channels
- #6245 - Autologin by token seems broken
- #6175 - Duplicates in stat_agent_periodic when recompiling from db_replic
- #6274 - UC assistant - Search results are not all displayed on frontend side
- #6314 - [Doc] Roaming agent does not work with 2 webrtc lines - Relogging a wertc agent with default line on other webrtc line fails
- #6315 - [Doc] Agent on pause is set back to ready status after refreshing page
- #5638 - Polycom plugin update for provisionning
- #6252 - Simplify Kamailio debug - log SIP call id
Consult the 2022.05.16 (Izar.16) Roadmap.
- #6134 - [C] - Missing notification on a function key during a double call
- #6100 - XiVO Desktop Application not shown properly under startup tab
- #6046 - XuC - If XuC is started while DNS (for rabbitmq) is unavailable, then it never gets back working
- #6172 - If a user logs in with the XiVO Client (!) and change its forward/dnd it breaks dnd/forward status on UC assistant
- #6074 - C - Paging not working in PJSIP
- #6187 - [Web-i] - Remove the …annoying… tooltip in provisioning/general/ that leads to misconfiguration of provd
Consult the 2022.05.14 (Izar.14) Roadmap.
#6144 - Fix mobile app connection
Behavior change The
LineConfigWebsocket CTI event contains a new field
sipPortwhich contains the SIP port of the SIP proxy.
Consult the 2022.05.13 (Izar.13) Roadmap.
- #5893 - ‘*’ DTMF fails from UC Assistant
- #6096 - xucmgt-call qualification starting countdown when call is hanged up
- #6023 - UC assistant - PIN removal for personal meeting room is not handled properly in form
- #5903 - Desktop tray icon is sometimes wrong due to race condition between missed call and chat
- #5969 - Loss of performance when having around +20 favorites contact
#5915 - Since PJSIP, Asterisk charts in WebIare not rendered properly
#6052 - [C] - Webi list user XSRF token issue
Behavior change Found in version 2021.09 too
Consult the 2022.05.12 (Izar.12) Roadmap.
- #5928 - [C] PJSIP “Insecure” option in the SIP Trunk configuration
- #5937 - xivo-full-stats restart indefinitely if CEL with appdata contains chars different than [a-zA-Z]
- #5901 - Add log in xuc with the connection type when a user is log in
- #5530 - Integration of new mobile application waiting message
- #5876 - As a user I want to have an error displayed when trying to login on MobileApp if XiVO is not properly configured for it
- #5927 - Update route to increased missed call for a user to be more RESTfull compliant
- #5945 - XDS - agid doesn’t start on MDS (python build problem)
Consult the 2022.05.11 (Izar.11) Roadmap.
- #5350 - Sometimes call is automatically hangup when answered by mobile application
- #5702 - Mobile application documentation
- #5769 - [C] - Recording - Cannot download access logs when it is too big
- #5902 - Handle call history when user is WebAppAndMobileApp (follow up of #5350)
#5336 - As mobile app user I want to correctly see missed calls on my webapp and mobileapp
Behavior change The number of missed call is now shared across all devices.
#5665 - Can’t relogin when we get the new version message on login page
#5698 - User with just email and no Number still have the chevron displayed near Email Icon
#5858 - UC assistant : bugs on call history page
#5865 - Update xucmgt to use missed_call from user preferences instead of computing it.
#5881 - Call history buttons appear out of the div
- #5586 - XDS - inter mds trunk fails to authenticate (PJSIP)
- #5708 - Outcall route in Webi are not displayed in order of select
- #5719 - [Webi - Device] Configuration model changes when we put a device in autoprov mode
- #5729 - monit generate high disk latency and high cpu usage (Izar)
- #5762 - Boss/Secretary filter does not work with PJSIP
- #5817 - Boss/Secretary filter “Ringing time” field missing in specific scenario
- #5843 - IVR uploads new audio file in place of an existing error.
- #5864 - Store the number of missed calls in user preferences
- #5885 - PJSIP - Wrong option mapping for directmedia = nonat in sip config
Consult the 2022.05.10 (Izar.10) Roadmap.
Components updated: xivo-confd, xivo-web-interface, xucmgt
- #5554 - Chat : on the ucassistant, when I receive a link in the chat, the link is not clickable
#5582 - Third party application is not displayed in desktop mode
#5684 - Calls pickup with (*8) extension fails with Cti.Dial
Behavior change *8 is not anymore supported by asterisk core feature but xivo-feature. Pay attention that migration sets it back to *8
- #5483 - Calling using keyboard on switchboard and cc agent does not work - Izar
- #5626 - When I search for a user without line number, I still see the dialing icon
Consult the 2022.05.09 (Izar.09) Roadmap.
Components updated: xivo-confgend, xivo-config, xivo-dird, xivocc-installer, xucmgt, xucserver
- #5439 - XDS - prevent loops between MDS (dialplan)
- #5378 - Jitsi on top of third party application
- #5250 - As an english user I want to see the mobile app association gif in english
- #5127 - PJSIP - Missing SIPCALLID sometimes in PhoneEvents
- #5229 - PJSIP - Missing call tracking when agent makes call from his deskphone (Snom or Yealink)
#5236 - PJSIP - Verify and adapt some sip options for pjsip
#5469 - Nginx bridged mode - make work phonebook search and search via phone [Izar]
#5486 - Sorting of accented characters [Izar]
Behavior change Lookup results are now ordered in way to prefer more relevant results (see default_json):
- exact matches of whole word
- exact matches at beginning of word
- exact matches anywhere
- matches similar to term
#5534 - [IVR] Include a xivo-pickup prior the call to Graphical IVR AGI
#5553 - XDS / PJSIP - Be able to call an outgoing call via another MDS
Consult the 2022.05.08 (Izar.08) Roadmap.
Components updated: xucmgt
- #5449 - Favorites are not sorted properly [Izar]
Consult the 2022.05.07 (Izar.07) Roadmap.
Components updated: asterisk, recording-server, xivo-config, xivo-dird, xivo-web-interface, xucmgt, xucserver
- #5196 - Astersik - crash in rtp_engine.c:565 (asterisk 18.10.1)
- #5402 - Asterisk crash with forked Invite
- #5437 - Build asterisk 18.10.0-1 with patch #5196
- #5366 - Cannot complete attended transfer from mobile when both UC and mobile are used
- #5410 - Mobile App - Dialplan does not loop correctly over contact list
- #5210 - Translate Recording Server Interface in English
- #5372 - Cannot make webrtc audio calls with Chrome 103 - Izar
- #5343 - PJSIP - xucserver - function isSipAttTransfer works only with chan_sip
- #5276 - Directory lookup: search result order given by server is not followed by frontend (Izar)
- #5333 - Mobile application: 2 registers on the Xivo side
- #5352 - Labels - when more than 150 labels, labels selection component in user form is broken
- #5430 - agent login (*30) does not work from UC app (with Cti.Dial)
Consult the Izar.06 Roadmap.
Components updated: edge-kamailio, xivo-web-interface, xucmgt, xucserver
- #5259 - Stop ringing on assistant when the call was answered on mobile
- #5275 - When dialing from UC, we should not make ring Mobile APP
- #5243 - Improve web-interface labels with filtering and better display for long lists
- #5260 - We should see a message in user configuration page if you have no labels
- #5254 - Cannot make ring UC and mobile app at the same time
Consult the Izar.05 Roadmap.
Components updated: config-mgt, edge-kamailio, ivr-editor, xivo-acceptance, xivo-config, xivo-db-replication, xivo-install-cd, xivo-web-interface, xivo-ws, xucmgt, xucserver
- #4803 - Windows 11 - Desktop Assistant prevents the use of “$” character in other app
- #5153 - Cannot do attended transfer when agent takes control of phone without cti user
- #5101 - Configmgt : dissociate a user account from the mobile app
- #5214 - Update user preference value for mobile and webapp
- #4629 - Desktop Assistant - Support Windows 11
- #5156 - When I have an incoming video conférence I would like to have a system notification that allows me to have an incoming video conference
- #5193 - Mobile app switch doesn’t get displayed or not displayed on the fly
- #5186 - Missing labels tables in xivo_stats replication
- #5223 - Video conference - modal display of incoming video conference in the foreground until an action is chosen.
- #5007 - As a user, I no longer wish to use the mobile application
- #5099 - Front end : dissociate a user account from the mobile app
- #5100 - Xuc : dissociate a user account from the mobile app
- #5105 - As a UA user I want to be warned if I try to login on the MobileApp that it is incompatible
- #5134 - Empty entry in the history cause js errors in the console
- #5215 - Evolution call line - hang up with a background via a red circle white phone
- #5187 - Lineconfig on mobile app association / disassociation should be trigered by userfeature table update
- #4608 - Debian 11 “Bullseye” : Update ISO Build
- #4836 - Update logo in the documentation for Helios and above
- #4941 - As a User I want to be able to chose whether the UC App or the Mobile App or both will ring
- #4954 - IVR - As an administrator I want to duplicate an existing svi
- #4984 - IVR - Be able to exit from the IVR in a number@context
- #5035 - PJSIP - Follow up
- #5083 - PJSIP - Update documentation
- #5112 - Dialplan/AGI - be able to call MobileApp or WebApp or both depending on the PREFERRED_DEVICE
- #5145 - Release Helios.11
- #5185 - Webi glitches
- #5189 - Devices synchronisation through xivo-provd-cli not working
- #5190 - IVR - Unable to delete existing IVR
Consult the Izar.04 Roadmap.
Components updated: config-mgt, xivo-full-stats, xivo-provd-client, xivo-provisioning, xivo-upgrade, xucmgt, xucserver
- #5116 - When muting in CC Agent, mic icon should be unstriked (same behavior as UC app)
- #5088 - Ringing device is set to blank if the previously selected device is disconnected on Desktop Assistant startup
- #4880 - [C] - Missing answer time for consultation call (queue call with transfer to another queue) in call_on_queue table
- #5047 - As a User, when I’m logged in UC App, I want to see a notification pop-up in app after my 1st login to the mobile application
- #5121 - Call failure notification does not update the called number when calling from search bar.
- #5168 - As a non-webrtc user I should not be able to see the ringing device menu
- #5122 - dual ringback tone when calling internal user phone from webrtc
- #5158 - PJSIP / Provisioning - Create in PJSIP the current autoprov peer
- #4597 - Debian 11 “Bullseye” : Add Xivo to PXE
Consult the Izar.03 Roadmap.
Components updated: xivo-config
Consult the Izar.02 Roadmap.
Components updated: xivo, xivo-agentd, xivo-db, xivo-lib-python, xivo-outcall, xivocc-installer
- #4836 - Update logo in the documentation for Helios and above
- #5113 - UC Addon yml file misses XIVO_SIPDRV env variable for xuc
- #5114 - Clean install doesn’t work
- #5119 - [PJSIP] Unable to dial external number using phone device
- #5120 - Can’t join/leave queue anymore in CCAgent
- #5123 - Upgrade from Freya does not work: migration script does not work in python3
- #5135 - Python3 - xivo-agentd-cli does not work
Consult the Izar 2022.05 Roadmap.
Components updated: asterisk, config-mgt, edge-kamailio, edge-nginx, recording-server, xivo-acceptance, xivo-agid, xivo-confgend, xivo-config, xivo-dao, xivo-db, xivo-jicofo-jitsi, xivo-jigasi-jitsi, xivo-jvb-jitsi, xivo-meetingrooms, xivo-prosody-jitsi, xivo-provd-plugins, xivo-provd-plugins-addons, xivo-provisioning, xivo-sysconfd, xivo-upgrade, xivo-utils, xivo-web-interface, xivo-web-jitsi, xivo-ws, xucmgt, xucserver
- #5003 - Prepare patches for asterisk 18.10.1
- #5044 - Re-enable pjsip notify
- #5068 - PJSIP - Update fail2ban regex
- #5049 - [Config-Mgt] Add meetingroom API to create and get Alias
- #5072 - Configmgt - create user preference when adding mobile push token
- #5008 - As a User I want to see a notification in my UC App after my 1st login to the mobile application
- #4752 - Send push notification on incoming call if user has a Firebase ID registered
- #4884 - Caller name is not displayed properly on the switchboard interface
- #4562 - Conference room - As a user I can blur or change my video background
- #4622 - Meeting Room - As a user I want to have a MR sharing link short enough in order to be able to visually check it / compare it
- #4643 - Meeting rooms: Disable ‘accept’ button when receiving a meeting room invite if I’m already in a meeting room
- #4691 - Meeting Room - Mozilla Firefox
- #5026 - Meeting Room - Update to latest release
- #5090 - Meetingroom - Screensharing quality - Make simulcast disable by deafiult
- #5104 - Meeting Room - Docker container logs are not rotated
- #4106 - Add a warning when using webRTC on unsupported browser
- #4861 - As a user I want to be able to listen to the ringtone I choose
- #5069 - XucMgt - Add popup with mobile usage informations
- #5074 - Front-end : display switch UC App and Mobile App
- #5098 - Contact initials break if contact name starts by something else than letters
#4980 - Be able to connect to OpenId and LDAP with a username containing an @
#5050 - [XUC] Use new Config-mgt meetingroom Alias API
Behavior change Meeting room sharing link have a new shortened format :
This link is permanent and doesn’t change if you edit your meetingroom (change name / pin…)
Old links are deprecated but still usable.
#5070 - Cti.js - Add cti user preference command and handler
#5071 - Xuc - Handle user preference apis
#5092 - [Pjsip] Make agent state work with PJSIP
#5095 - Xuc - Add a property to indicate if the user has the mobile app in the lineConfig
#4654 - [Meeting Rooms] Provide an IVR to allow joining (any existing) Meeting Room in Audio mode from outside (through a did)
#4901 - [Python 2 to 3] Remove remains of python 2 compatibility
#4902 - Python 2 to 3 conversion - follow up
#4948 - Provd-related issues wrapper
#5021 - Documenation for install of edge and jitsi should be upgraded for bullseye
#5022 - Automatic modification of the login field.
Behavior change Login field is not automatically converted to lowercase letters after a modification in the admin web interface.
#5036 - [Pjsip] history does not work properly because different name of interface
#5038 - [Pjsip] - Webi configuration should reload the correct module (chan_sip or pjsip)
#5040 - [Pjsip] - Call between MDS
#5043 - MDS upgrade fails on Asterisk files
#5048 - [DB] Add alias column in meetingroom table
#5052 - Db creation fails
#5053 - Upgrade might fail when rebuilding dahdi for old kernels
#5073 - PJSIP - Audio volume control does not work (missing SIPCALLID)
#5075 - PJSIP - the OPTIONS message should only be sent if qualify=yes is set on the peer
#5077 - PJSIP - User status are not present
#5078 - PJSIP - Add queue/group user member with correct channel type
#5079 - PJSIP - UA user - Can’t call UA user on its deskphone (SIPPEER function does not exist)
#5082 - PJSIP - Finish the dialplan (SipAddHeader etc.)
#5097 - Asterisk key generation might fail
#5102 - [PJSIP] Audio only call to meeting room
#5107 - PJSIP - Mobile App - Rais the number of max_contact for webrtc peers
- #4906 - Move xivo plugins in python3
- #4985 - [Pjsip] Autoprov requires user auto create and context
- #5093 - Python3 - provd - Still some str/byte decode problem Error with the is_sensitive_filename function
- #5096 - PJSIP - Yealink phones are banned during 10min after being provisioned