Tests that multiple PJSIP contacts may ring at the same time and don't overlap each others

Recording saves call records and Configuration manages users that can access it

Agen can login, change styate, see available agents on its queue

Checks SpagoBI and statistics using sample reports

Validate that the call history is generated and available through xuc interfaces

Verify that desktop assistant is available to be downbloaded from web interface and at least runs on Windows

Verify that desktop assistant MSI installer is build and works

Checks that the signed desktop assistant on the mirror is downloadable, signed, can be installed, and launch

Test an outgoing WebRTC call using the latest Stable and Beta release of Chrome under Linux.

Please, write down in the comment box the exact versions of Chrome tested.

Test an outgoing WebRTC call using the latest Stable and Beta release of Chrome under Windows.

Please, write down in the comment box the exact versions of Chrome tested.

This test aims to verify that a webrtc user on mds1 can call and receive internal calls from a phone user on MDS0.  

Calls between external user and webrtc on MDS 

Test new XiVO UC installation on Deneb or newer XiVO

Test XiVO UC upgrade

   upgrade xivo and MDS from izar 

Test to init XiVO with run_script to true

Warning: This test is supposed to be not required as we have the daily install automated test

Objectifs:

1. Une touche de supervision de parking permet de transférer un appel vers un parking
2. Une touche de position de parking permet de récupérer un appel parker dans celui-ci

Objectif : faire un appel sortant à partir d'un appel entrant avec l'application DISA.

Test de non-régression de #4305, #4306.

The goal is to verify that we can use the new global subr internal

Goal : You can create, edit and delete routes in the web interface

Be able to

• add
• search
• call

               a phonebook contact

See display name for incoming call from a phonebook contact

Test that you can update line protocol option for users 

• csv update - webrtc ua sip : update_users.csv
• update_users.csv

User with labels can be updated

• users_voicemail.csv
• users_voicemail.csv

Change user's line type of line with custom option

Verify that custom option are not removed when changing line type

The purpose of this tests is to check that there is no timeout or too long execution time when saving user with many funckeys

this testcase cover the edit label webi page

The purpose of this test is to verify that you are able to add qualifications to the queue.

Test the basic conference feature :

• without PIN
• with user PIN

Test the conference behavior with an admin PIN

Conference behavior with an admin and a waiting room

Test that disabled extension cannot be called:

• line
• user
• group
• queue
• meetme
• incall

Important note

Call from MDS can fail because of https://projects.xivo.solutions/issues/2953

Checks that boss filter works

See https://documentation.xivo.solutions/en/latest/ipbx_configuration/boss_secretary_filter.html for configuration

Si le destinataire est en ligne et rejette mon appel (busy), je peux demander à être rappelé lorsque le destinataire sera à nouveau disponible.

Test swagger

This is mainly to test thath dxtorc is working.

Tests if provisioning of more phones through the webi works

Applying correct configuration should not produce error

Vérifier qu'une sous routine est exécutée correctement.

Vérifier qu'une sous routine est exécutée correctement

Verify that the XDS subroutine is called when a call goes from a MDS to another

Test monitoring on Webi > Services > Monitoring page

Test results of "xivo-service restart all"

• Services are stopped without delays and errors
• Services are started without delays and errors

Start and stop sequence when shutting down or booting is not handled by xivo-service. So it is different.

Note: Services should start even if monit is not running, but some of them fail because db is not running

 Test the extraction of all xivo users as contacts

Be able to join ivr using number from extern

This requires IVR Editor up and running

See 

 - doc is on the way as of 2021.15.05

Be able to duplicate existing IVR flowchart

Checks that the action buttons works as expected

This test checks that you can correctly add, edit and delete voiceprompts (the audio files available to be played in the ivr)

Checks that mass import works

Test the enhanced search feature in UC Assistant.

Check call state if a non standard endpoint like a meet me answers the call.

When in a conference, I should be able to see administrator

When joining a conference, the UC Assistant should be able to display conference participant

When entering a conference, a participant should be able to mute/unmute himself

Check organizer actions on other attendees

To verify that conference events are sent to the correct user even after the beginning of a transfer

(non regression of #2960)

For webrtc users 

The organizer can stop the conference.

Organizers can invite a user to the conference

participants in a conference can start a second call by typing a phone number in the search bar 

Organizers can invite to conference by typing a phone number in the search bar

Tests the search bar display and actions : call, invite to conference and search

When an organizer leaves a conference, the participant states are set to their default state. All muted or deaf participants are not muted or deaf anymore.

Checks that additional parameters works on conference invite

Check that when we are in a conference and we have a call on hold at the same time, we can include the call to the conference by clicking on "include to conference" button ( note that it is not the same action as inviting user to conference" and icons are different)

Be able to send a chat message while other party is offline

Thes tests are aimed to tests either for WebRTC or SIP phones the call forwarding that can be configured in web assistant.

new contact detail in a card instead of an accordion

The goal is to verify that the operator can retrieve a call while he has an ongoing call

This test test that the statistics displayed in the ccmanager can be customized partially or completely

Test that you can hold resume a call when you are already talking with another established one

Test conference from CC Agent

Test Third Party integration in CCAgent by displaying a fake integration in the application.

See Third Party Documentation for more information. 

Validate that variables are propagated from the incoming channel to agent channel

Test badge icon when calls are waiting in a queue or multiple queues.

When currently logged in as an agent on a phone number, you cannot login again on another phone.

Agent can take the phone of another currently logged in agent

On the login page of the ccagent (in the browser), I can download the desktop assistant by clicking on the windows button. 
The Linux-Debian button should lead to the documentation. 

When the window is too small, a message appears for the user to extend it. 

Check call line on ccagent and mute/unmute icons for phone and webrtc users

Test that an agent can take the control of users having a line but no cti credentials

Verify the behavior of the CC Agent's Statistics Carrousel as described in the documentation (https://documentation.xivo.solutions/projects/xivo-nextlts/en/latest/contact_center/ccagent/ccagent.html#statistics)

Display versions of xivocc containers and images

Warning : for XiVO >= Maia with docker-compose v2, the display has changed from xivocc_xuc_1 to xivocc-xuc-1. For testing on previous lts, keep that in mind.

Test if Xivo CC runs with customized variables

When Xuc connection is lost while using Assistant with SIP phone , user is redirected to login page. 

When Xuc connection is lost while using Assistant with WebRtc , user is shown error dialog and after clicking logout is user redirected to login page (also terminating any current call).

WebRTC endpoint can have only two calls

Test dtmf keypad to close when called user hangs-up.

Test webrtc conference hold/resume 

This testlink checks that the webrtc doesnt explode if somebody with special letters in his name calls the webrtc user

Test that PhoneEvent Failure is seen in websocket when Cti.dials fails

When A calls W1 (a webrtc user)

Test that whatever the caller's codec, we propose first the Opus codec to W1

Tests Recording API to listen recorded calls 

Purpose of this test is to verify that recording files are purged before than the "number of weeks to keep recording"

Transfer a call to another user after the destination answered the call.

Ensure bug #1498 is fixed. Upon startup, xuc should be able to load active calls.

This test is a chek-up to validate that we are correctly feeding reporting stats database with correct values when doing consultation calls (queue calls tranfered to another queue after agent destination answered)

Test that we can recompute statistics from "stats-recompile.sh' script avaible in /usr/bin of XiVOO CC/UC environement

Test Real time calculated Queue statistic generated by xuc - see note about custom tresholds in documentation.

Test CTI.Sheet and popupUrl in CCAgent by displaying attached data to the call and open new page for url if defined.

See Sheet documentation and popupUrl documentation for more information. 

Test that a UA user can connect to UC Assistant, be recognized as a WebRTC user, and call

The goal is verify that the current (ongoing) call can be send directly to the hold queue

This test checks that microphone permission error ask the user to switch to the phone mod or beeing logout from the application

Checks if *8 works for UA users

Verify that xuc retries connection on start to rabbitmq

Checks that xuc disallow everybody that is not cti user to open CTI websockets

This test ensure the xuc connection to AMI is retry and is behaving well with multiples mds unreachable.

Test Single sign-on feature with keycloak Server with a version >= 18

Test Single sign-on feature with Azure Server

Test Single sign-on feature with Kerberos

Test Single sign-on feature with CAS Server

Test Single sign-on feature with keycloak Server 10.2

  The goal is :

•     Display all search results ( Ldap users+ xivo users) even if they were looking the same.
•     Search results (Ldap - xivo users)   are sorted in proper way.

test ansc login procedure

We want to check whether the request POST /api/2.0/dial/toqueue works as expected.

• Postman configuration : dialtoqueue.json
• dialtoqueue.json

We want to see the list of default ACLs for a CTI user

• Postman configuration : postmanconfiguration.json
• postmanconfiguration.json

Making sure the electron window can stretch to full screen during a video call.

this test will make sure that the desktop assistant displays correctly on every resolution and also with custom zoom factor

check if the display works on windows 10 when the user has a different zoom on each of his screen, for example an user with a thinkpad laptop zoomed at 125% and a 1080p screen with standard 100% zoom

This test is to check if we can correctly disable the download buttons that are on the login page

If I manually change the width and height of my desktop app, the width will go back to default after closing the video

Check that the contextual menu works on electron

Ensure that we can zoom in, zoom out, and reset the zoom on a desktop app.

Verify that incoming call to the user on any MDS is correctly routed.

Ensure routing between MDS is working when forwarding a user to another one 

The purpose is to verify that user on MDS1 is allowed or denied to call another user on MDSx

The purpose of this test is to verify that internal caller id is passed through the dialplan

Calls between webrtc users on different MDS

I can call a Group/Queue/Meetme configured on XiVO Main

Test that recordings directory is sync both ways between Main and MDS.

• mohpiano.doc
• mohpiano.doc

Transfer of device from msin to MDS has to work properly.

The purpose of this test is to check you can edit your stun/turn server configuration and propagate it to the asterisk

Test that we can override default 5060 port or completey ignore it

Be able to join meeting room using number from mds

  When i call a non existing Meeting Room number (say **1234) i should fallback to a message that says 'invalid number' and then hangup

Test the enhanced search feature: a mix of "fulltext search", "simple like search" and "fuzzy search"

Test that we can set / update a token for mobile push notification for mobile application users.

Checks that users with mobile app can use group calls

Test if mobileApp wakes up when we call mobileApp user on his "locked"  mobile