Call can be answered or paused on device controlled from Assistant

XiVO and XiVO UC >= Aldebaran.07 is installed

webrtc user W1

user U1 with device

U1 log in to Assistant

U1 is logged in

Open second browser window

W1 log in to Assistant

W1 is logged in

W1 call U1

U1 answer from Assistant

U1 device is on call

U1 click Hold from Assistant

U1 device is on hold

U1 click Resume

U1 click Hangup

Call end

U1 dial W1 from call history

W1 is ringing

Test new XiVO UC installation on Deneb or newer XiVO

XiVO >= Deneb

Open http://XIVO_IP

Admin page opens

Install XiVO UC by:

• xivouc-installer

You are asked to continue

Confirm

• xivouc installs
• no errors

Dowload the images

  xivocc-dcomp pull

And launch UC:

  xivocc-dcomp up -d

Verify no errors in xivo-db-replic logs

• ERREUR: la relation « callback_list » n'existe pas
• ERREUR: la relation « preferred_callback_period » n'existe pas

If you want to test also Assistant and Monit:

• If you chose to not restart XiVO by UC installer, restart it by:
  xivo-sevice restart all

XiVO restarts

Open http://XIVO_IP

Redirects to https

Error 504 Bad gateway (if Assistant is not running)

Open https://XIVO_IP/admin

Webi opens

Test XiVO UC upgrade

Old version of XiVO with UC installed

(Follow X-917 if you don't have XiVO UC)

Open https://documentation.xivo.solutions/projects/xivo-nextlts/en/latest/installation/xivo/xivouc/xivouc.html#upgrade

or xivouc.rst file in xivo-solutions-doc (if it is newer)

Follow it to upgrade XiVO UC

If needed, also follow the Upgrade Polaris to Aldebaran page

Documentation is complete

All installed without errors

Verify no errors in xivo-db-replic logs

Open https://XIVO_IP/admin

Wait until Monit shows running components

Process docker is running

Containers are in Accessible state

Open https://XIVO_IP

UC Assistant opens

Open https://XIVO_IP/ccagent

403 forbidden

Open https://XIVO_IP/ccmanager

403 forbidden

Open http://XIVO_IP:8070/ccagent

Page is not available

ERR_CONNECTION_REFUSED

(it listens on 127.0.0.1:8070)

Open http://XIVO_IP:8090

xuc status page opens

Open https://XIVO_IP:8443/sample

Webrtc sample page opens

Open https://XIVO_IP/api

Click Available APIs > XiVO Confd

Swagger for XiVO Confd opens

Objectif: test installing Xivo from ISO (64 bit version)

Last modifications: #1497, #1760

Download ISO:

• a) Not published: http://mirror.xivo.solutions/iso/.iso-temp/
• b) Published: http://mirror.xivo.solutions/iso/archives/

Install Xivo on VM using ISO (64 bit version) - see documentation

Xivo is installed

Run the wizard - see documentation

Wizard finisheds without error

You are able to login to Xivo WebUI

Provision two SIP phones (or connect two softphones)

Make call between them 

Call is working

valider install pxe sur le xivo buisiness

1 machine buisiness avec une carte dahdi e1/T1 digium, et un cable T2 croisé reliant les 2 itf de la carte (1-2 vers 4-5)

Boot sur pxe, choisir la version a tester

install fonctionnel, vérifier avec la commande aptitupde install que tout est correctement installé  :

~# aptitude install
Reading package lists... Done
Building dependency tree      
Reading state information... Done
Reading extended state information     
Initializing package states... Done
No packages will be installed, upgraded, or removed.
0 packages upgraded, 0 newly installed, 0 to remove and 0 not upgraded.
Need to get 0B of archives. After unpacking 0B will be used.
Reading package lists... Done
Building dependency tree      
Reading state information... Done
Reading extended state information     
Initializing package states... Done

~#

Doit ne renvoyer aucune erreur. Si des paquets restent a configurer (sauf tasksel et taskel-data), c'est qu'il y a eu un problème avec le packaging.

executer le wizard d'install via la web interface et les pré-requis nécessaires.

se logguer en administrateur

acces page avec info de monitoring

Install latest release of XiVO PBX

Install latest release of XiVO CC on a separate server

Upgrade XiVO PBX

Upgrade XiVOCC

Upgrade XiVO PBX

Upgrade XiVOCC

XiVO CC has configuration:

• Debian 8
• docker-engine installed

Upgrade XiVO PBX

Upgrade XiVOCC

Goal : have the correct Caller ID during transfers with an outgoing call

• 1 outgoing call
  • in the Exten tab of this outoging call
    • the "Caller ID" field is void
• 2 users
• On the xivo, Global parameters -> SIP Protocol, tab default :
  • Trust Rmote-Party-ID : yes
  • Send Remote-Party-ID : PAI

User A calls an external number.

The external callee rings, when he answers, the voice stream works.

User A transfer directly towards user B.

B phone set displays :

• calling number: the number of the external callee that A dialed

User A calls again an external number

A makes an attended transfer towards user B

B phone set displays :

• calling number: the number of the external callee that A dialed

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

• Have no trunk available
• Have two schedules with one open and one close

Try to create a new route in Services/IPBX/Call Management/Outgoing calls

You can click on the link create a trunk in the fieldset destination

Create a trunk to xivo-net (see https://wiki.avencall.com/index.php/Xivonet)

The trunk is correctly created

Create your route to xivo-net with a schedule open

The route is correctly create

Try to do a call toward your route

The call is established

Change the schedule to the closed one

The call is not established

Edit your route

The route is correctly edited

Create a trunk custom in Services/IPBX/Trunk management/Customized then create a new route with this customized trunk

The route is correctly create

Try to delete your trunk

You can't delete your trunk because it has a route associated

Delete your route

Your route is deleted

Delete your trunk

Your trunk is deleted

Be able to

• add
• search
• call

               a phonebook contact

XiVO and CC / UC >= Callisto.13 or Deneb.00

1 agent

Create phonebook contact C1 
(Services > IPBX > IPBX Services > Phonebook)

Enter:

• firstname
• lastname
• display name
• Office > phone = +33445566

Contact was created

Search C1 from UC Assistant by: 

• firstname
• lastname

C1 is found each time

His full name is displayed

Point mouse on C1 name

You see his number reformatted: +33 445 566

Click C1 number

0033445566 is dialing

Repeat from step 2 with CC Agent

The same results as in UC Assistant

See display name for incoming call from a phonebook contact

XiVO and CC / UC >= Callisto.13 or Deneb.00

agent A1 with device

queue Q1 with agent A1

Trunk to xivo-net

Create phonebook contact C1 
(Services > IPBX > IPBX Services > Phonebook)

Enter:

• firstname
• lastname
• display name
• Office > phone = +33445566

Contact was created

Login agent A1 with device to UC Assistant on your xivo

You are logged

Open new tab.

Login to xivo Dev UC Assistant (https://xivocc.dev.avencall.com)

• as 1240 / 0000

(This webrtc user must have custom callerid set to +33445566,
if something fails, verify it on xivo Dev)

You are logged as user "webrtc reverse lookup"

Call A1 through xivo-Net (e.g. 0123111020)

A1 has incoming call dialog from:

• C1 full name
• +33445566

A1 phone displays:

• C1 full name +33445566

Call A1 through xivo-Net (e.g. 0123111020)

A1 call control shows:

• C1 full name +33445566

A1 phone displays:

• C1 full name +33445566

Hangup.

Dial +33445566 from history

0033445566 is dialing

Repeat from step 2 with A1 logged in CC Agent

After that, do also the steps bellow.

The same results as with UC Assistant

Call Q1 through xivo-Net

Wait until A1 rings

Open customer history > PATHWAY in CC Agent

You see new entry for Q1

Test that you can import users with WebRTC option activated.

A XIVO with XIVO UC/CC (to be able to do a WebRTC call)

The attached sample CSV file.

Download the CSV sample file and check that the context and extension number will match your test lab.

CSV file ready to be imported

In Services -> IPBX -> IPBX Settings -> Users

Click on the '+' button and select 'Import file'

Select the file previously downloaded.

Click on Save

File is imported without error

Verify the you have two new users :

User1 WebRTC

User2 NoWebRTC

Edit user User1 WebRTC

Go to in tab Line

Click on the line Name

It open the line configuration

In the Line configuration, go to the Advanced tab

You can see the option

webrtc  yes

Now edit User2 NoWebRTC and follow the same path to view its line configuration.

Go to the Advanced tab ot User2 NoWebRTC's line configuration

Verify that the webrtc option is not set at all

Now open the UC Assistant with https://

You get the UC Assistant login page

Log in with

uwebrtc/1234

You are logged in and see the PC icon

(the icon for call forwarding)

Dial *55

It works and the echo test works

Logout and log in with

unowebrtc / 1234

You are logged in and there is the Phone icon

(the icon for call forwarding)

• import_user_with_webrtc.csv
• import_user_with_webrtc.csv

Test that WebRTC users are correctly exported

Create a user WebRTC

(Doc : WebRTC User Configuration)

Go to

Services -> IPBX -> IPBX Settings -> Users

You see the list of users

Click on the '+' button

and select Export to CSV

There is no error and the users are exported to a csv file

Open the CSV file and verify that

- webrtc column is set to 1 for the WebRTC users

- webrtc column is set to <nothing> for non-WebRTC users

Objectif: Lorsqu'un appel est reçu en période fermée, l'action configurée est exécutée.

modifier HORAIRE1:

• modifier la période ouvrée (doit exclure maintenant)
• définir comme action hors période ouvrée par défaut:
  • Destination: Customized
  • Command: Playback(digits/40&digits/2) 

appeler l'utilisateur (extension 102)

On entend le 42.

Appel d'un numéro entrant avec horaire, l'appel suit l'horaire quelque soit la destination choisie pour l'appel entrant.

• Un utilisateur configuré
• Un horaire fermé pendant l'heure du test avec en destination cet utilisateur
• Un numéro entrant avec destination:sound file

Appeler le numéro entrant

le fichier audio configuré dans le numéro entrant joue.

Assigner l'horaire au numéro entrant

Appeler le numéro entrant

L'utilisateur destination de l'horaire sonne

The purpose of this test is to verify that qualification can be create with sub qualifications

Naviagate to Call Center - Qualifications

Click on "+" button

Enter qualification name "qualif1"

Save

Qualification is created

Number of sub qualifications is 0

Click on "+" button

Enter qualification name "qualif2"

Add 2 subqualifications:
- click on small green "+" button to add sub qualification "subqualif1" and "subqualif2"

Save

Qualification is created

Number of sub qualifications is 2

Click on "+" button

Enter qualification name "qualif2"

Qualification is created

Click on "+" button

Enter qualification name "qualif3"

Add 1 subqualification with name "subqualif1"

Qualification is created with subqualifications

The purpose of this tests is to verify that qualification and subqualification names can be edited

Edit qualification

Change its name

Save

The name of the qualification is changed

No errors

Edit qualification with at least 1 sub qualification

Change the name of the sub qualification

Save

Re-open the same qualification

Sub qualification name is changed

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

At least 3 qualifications with some sub qualifications created.

At least 2 queues.

Navigate to Call Center - Queues

Edit queue 1, open tab Qualifications

There are no qualifications assigned to this queue.

There are qualifications available.

Add qualification 1 

Click Save

No errors after saving the form.

Re-open the queue, tab Qualifications

The qualification 1 is assigned to this queue.

Repeat for queue 2 and select the same qualification 1

No errors after saving the form.

Open queue 1 and queue 2

Qualification 1 is assigned to both queues.

Login to the database and verify:

SELECT * FROM queue_qualification;

There are 2 entries.

The ID of the qualification 1 belongs to the IDs of the queues.

Edit queue 1 and un-assign the qualification 1

The qualification is un-assigned.

Verify

SELECT * FROM queue_qualification;

The relationship qualification 1 = queue 1 is removed.

Edit queue 1 and add all qualifications.

Save the form.

No errors after saving the form.

All 3 qualifications are assigned to the queue 1.

Login to the database and verify:

SELECT * FROM queue_qualification;

There are 3 new entries in the table.

The ID of the qualification 1, 2 and 3 belongs to the ID of the queue 1.

 id | queue_id | qualification_id 

Conference behavior with an admin and a waiting room

A conference with:

• Name: Testlink conference
• Number: 8000 (or any number in your range)
• Music on hold: default
• User PIN : 1234

Three users U1, U2, U3

In xivo webi, edit conference

in Organizer tab, check the box 'Wating room' (without setting an organizer pin)

save form

Error is raised

If you go in Organizer tab, message is displayed saying that you can't check this option without and admin pin

then add an admin PIN|9876

and check (if not checked) the Waiting room

save form

Form is saved without error

U1 calls conference 8000

U1 is prompted for PIN

U1 enters user PIN 1234

U1 hears 'conf-waitforleader.slin' message

and then is on hold

U2 calls conference 8000

U2 is prompted for PIN

U2 enters user PIN 1234

U2 hears 'conf-waitforleader.slin' message

and then is on hold

(U1 and U2 can't speak to each other, or don't hear each other)

U3 calls conference 8000

U3 is prompted for PIN

U3 enters admin PIN 9876

U1 and U2 hears conf-placeintoconf.slin message

U3 hears join sound

U1, U2 and U3 are in conference

U3 hangup

U1 and U2 hear conf-leaderhasleft.slin

U1 and U2 are on hold

they can't speak to each other or ear each other

A call right can be applied to a user

• A user U1 provisioned on a phone P1
• A user U2 provisioned on a phone P2

Add a call right via Services / IPBX / Call management / Call permissions with :

• General tab -> Authorization: Deny
• General tab -> Extension U2's internal number
• Users tab, select U1

Call right is saved

With U1 call U2

You are told that you don't have the right to call this number

Edit the same call right

in General tab -> Password: add a passord (e.g. 1234)

and save

With U1, call U2

You are told to enter a password

Enter a wrong password

You are told that it is the wrong password

Enter now the correct password

U2 rings

Edit the same call right.

In Users tab, remove U1

Save call right

With U1 call U2

U2 rings

A call right can be applied to a group

i.e.: to all users belonging to the group

• three users U1, U2 and U3
• a group with U1 and u2

Add a call right via Services / IPBX / Call management / Call permissions with :

• General tab -> Authorization: Deny
• General tab -> Extension U3's internal number
• Groups tab, select G1

Save

Call right is saved

With U1 call U3

You are told that you don't have the right to call this number

With U2 call U3

You are told that you don't have the right to call this number

Edit the same call right.

In Groups tab, remove G1

Save call right

With U1 call U3

U3 rings

With U2 call U3

U3 rings

Purpose: The BSFilter function keys can be added to users configured as boss and secretary

No BS filter

Edit user who will be boss

Add new function key

Select type "Filtering Boss - Secretary"

Destination field contains hyperlink "Create callfilter"

Click "Create callfilter"

Create new BS filter

Choose 1 boss and 1 secretary

Filter It is created

• Edit boss
• Add new function key for BS filter
• Edit secretary
• Add new function key for BS filter

• you have to add custom keys and then add the * 37 extension that you get when you edit the callfilter.

Once the phones are reset, the function keys are now functional on the phones.

Edit boss

Edit BS filter function key

Select type "User"

You can choose user in the "Destination" field

Objectif : Un filtrage modifie le temps de sonnerie d'un appel

Avoir un filtre boss secrétaire de configuré.

Dans le filtre, changer le temps de sonnerie du Boss pour 15 sec et utilisez le mode "Boss first then secretaries simultaneously".

Le boss sonne 15 secondes avant que les secrétaires sonnent.

Dans le filtre, utilisez le mode "Secretaries simultaneously" et mettez le "ringing time" a 15 secondes.

Les secrétaires sonnent pendant 15 secondes.

If the destination of a direct transfer does not answer, the caller is sent towards its voicemail.

• Switchboard
• User2, called user with voicemail
• User1, calling user

User1 calls switchboard

Switchboard answers

Switchboard makes a direct transfer (F4) towards User2

User2 rings

User2 doesn't answer

User1 arrives to voicemail of User2 and can leave a message

The switchboard must be able to make calls from the xlet switchboard / directory

• Switchboard S
• User A with a provisioned phone

Click the Call button

Focus is on the xlet directory

Search for user A + enter

The focus is in the result list

• Select A
• Enter

• Your phone picks up 
• The phone of A plays the ringing tone.

All buttons except the "Cancel" button are grayed out.

Call a non-existent extension in the same way.

< Callisto (2019.05)

• A popup (or status bar if automatic reconnection is enabled) indicates that the extension is unreachable

>= Callisto (2019.05)

• No popup, call is routed depending on the route configuration

Call an external number in the same way.

The phone picks up and plays the ringing tone.

Objectifs:

1. A SNOM POPC can answer a call
2. A POPC SNOM can not receive a second call while it is on call

• A POPC configured with a Snom 720.
• The device must be configured with the "switchboard" option enabled.

Answer a call from the POPC.

The phone answers and the call is answered.

Make second call to the extension (without going through the POPC queue).

The second call is rejected.

Objectifs:

1. Un POPC Yealink peut répondre à un appel
2. Un POPC Yealink ne peut pas recevoir un deuxième appel pendant qu'il est en ligne

• un POPC configuré avec un poste Yealink T46G.
• la terminaison doit être configuré avec un plugin "xivo-yealink" et l'option "switchboard" d'activé.

Répondre à un appel à partir du POPC.

Le téléphone décroche et l'appel est répondu.

Émettre deux appels vers le poste (sans passer par la file POPC).

Le deuxième appel est rejeté.

Verify that the initial HA configuration works.

Follow the documentation :
https://documentation.xivo.solutions/en/latest/xivo/high_availability/high_availability.html

One XIVO configured with users etc. that will be the master.

A XIVO without configuration that will be the slave.

Configure the slave IP on the master.
The slave IP must be the VOIP interface IP.

Verify that the cronjob is created on the master :

$ cat /etc/cron.d/xivo-ha-master
0 */1 * * * root /usr/sbin/xivo-master-slave-db-replication <ip slave>
0 */1 * * * root /usr/sbin/xivo-sync

$ ls /etc/cron.d/xivo-ha*
/etc/cron.d/xivo-ha-master

If it was not already done, do a xivo-sync -i

The command must have created a file in /root/.ssh/ named xivo_id_rsa, and must have launched an ssh toward the slave to copy the ssh key.

Configure the master IP on the slave.
The IP address of the master must be the VOIP interface IP.

Verify that the cron job is created on the slave :

$ cat /etc/cron.d/xivo-ha-slave
* * * * * root /usr/sbin/xivo-check-master-status <ip master>

$ ls /etc/cron.d/xivo-ha*
/etc/cron.d/xivo-ha-slave

Synchronize your provisionned phones

Configure the XiVO Clients with the IP of the slave server as the secondary IP.

• from master to slave
• from master to CC

• Xivo >= 2019.01.00
• User with provisioned phone
• HA configured and running - use VoIP interface for HA configuration
• DB replic configured on slave - see documentation > High Availability > Interconnection with XiVO CC
• All xivo services are running on master
• asterisk, config_mgt, db_replic and other services are stopped on slave

• configuration from master to slave
• configuration from master to CC
• events from master to CC

On master:
select id from cel order by eventtime desc limit 1;
remember the id

On slave:
select id from cel order by eventtime desc limit 1;
remember the id

On CC:
select id from cel order by eventtime desc limit 1;
remember the id

Call *55

On CC:
select id from cel order by eventtime desc limit 1;

id changed

Table with user "master" was replicated

On slave:
select id from cel order by eventtime desc limit 1;

id did not change

On slave:
select * from userfeatures;
Or refresh user list in webi

Table with user "master" was replicated

• events from slave to CC

On master:
xivo-dcomp stop db_replic
ifdown <VoIP interface>
If you use another interface in HA configuration, disable it also

On slave:
watch xivo-service status

All services start after cca 2-3 minutes

id changed

id changed

Table with user "slave" was NOT replicated

PART 3 - master is UP AGAIN
Test replication of:

• configuration from master to slave
• configuration from master to CC
• events from master to CC

asterisk, config_mgt, db_replic and other services are stopped after cca 1 minute

id changed

Table without users "master" and "slave" was replicated

id did not change

Table without users "master" and "slave" was replicated

Test swagger

XiVO >= 2017.11.06

Edit Configuration > Web Services Access > xivo-agid

Add row 'amid.action.reload.create'

Copy password to clipboard

Save

Form saved

Obtain an authentication token:

Use username: xivo-agid, password from clipboard

https://documentation.xivo.solutions/en/latest/api_sdk/rest_api/embedded_swagger.html#obtain-an-authentication-token

Copy the token to clipboard

Paste the token to inpud field next to Explore button

Token was generated

Click:

1. XiVO Agentd
2. agents
3. GET /agents
4. Try it out

Response Code:

200

Click:

1. XiVO Confd
2. lines
3. GET /lines
4. Try it out

Response Code:

200

Click:

1. XiVO AMId
2. action
3. POST /action/{action}

Enter action value: reload

Click Try it out

Module Reloaded

Response Code:

200

Obtain new authentication token:

Use username xuc, password 0000

Use backend xivo_user

Copy the token to clipboard

Paste the token to inpud field next to Explore button

Token was generated

Click:

1. XiVO Dird
2. personal
3. GET /personal
4. Try it out

Response Code:

200

Verify that SIP Diversion Header is present and contains

- correct reason

- correct callerid

(see #1112 and #1452)

User U1, U2, U3, U4

One outgoing call rule OR1 (towards xivo-net) with the CallerID 0123456789 defined in the Callerid column of the rule in the Exten tab.

SIP debug activated on U3's line : in asterisk CLI

sip set debug peer <U3's sip peer>

U2 activates forward on busy towards U3

• on phone, dial : *23<U3's number>

On XiVO Webi set U2's Simultaneous calls to 1

Open asterisk CLI

U4 calls U2

U2 answers

Then, U1 calls U2

As U2 is busy, U3 rings

In asterisk CLI verify that the SIP INVITE sent to U3 contains this Header :

Diversion: "U2" <sip:1002@10.32.5.7>;reason=user-busy

Everyone hangup

U2 activates forward on no-answer towards U3

• on phone, dial : *22<U3's number>

Open asterisk CLI

U1 calls U2

After U2's ring timeout, U3 rings

In asterisk CLI verify that the SIP INVITE sent to U3 contains this Header :

Diversion: "U2" <sip:1002@10.32.5.7>;reason=no-answer

Everyone hangup

U2 activates unconditionnal forward towards U3

• on phone, dial : *21<U3's number>

Open asterisk CLI

U1 calls U2

As U2 is forwarded unconditionally, U3 rings

In asterisk CLI verify that the SIP INVITE sent to U3 contains this Header :

Diversion: "U2" <sip:1002@10.32.5.7>;reason=unconditional

Everyone hangup

U2 activates unconditionnal forwards towards external number (which will go through outgoing call rule OR1)

• on phone, dial : *21<external number>

Open asterisk CLI

Activate sip debug on SIP trunk that will be used with OR1, like :

    sip set debug peer trunk-maq-eal

Verify that U1 has no Outgoing callerID configured

U1 calls U2

As U2 is forwarded unconditionally, external user rings

In asterisk CLI verify that the SIP INVITE sent to trunk contains this Header :

Diversion: "0123456789" <sip:0123456789@10.32.5.7>;reason=unconditional

Everyone hangup

On U2 deactivate all call forwards :

• on phone dial : *20

In Webi, reset U2's Simultaneous calls parameter to the default value (2)

Verify the correct caller id when call is forwarded externally

(see #1477)

External user E
Internal users U1, U2
External (mobile) user M

One outgoing call rule OR1 (towards xivo-net) with the CallerID 0123456789 defined in the Callerid column of the rule in the Exten tab.

Set U2's Outgoing Caller ID to Customize 

• set to 2121002

U2 activates unconditional forward towards M

• on phone, dial : *21<M>

E calls U2

E is forwarded towards M

Verify that M sees 2121002 as caller id

Everyone hangup

U1 calls U2

U1 is forwarded to M

Verify that M sees 2121002 as caller id

Everyone hangup

Set U2's Outgoing Caller ID to Default

Repeat steps 2-5

Verify that M sees 0123456789 as caller id

note: you may need to re-provision the phone during this step

Set U2's Outgoing Caller ID to Customize

• set to 2121002

Delete (sip) line of U2, create a new one and assign it to U2

Repeat steps 2-6

Verify the user U1 with no rights to call mobile phone numbers can call towards mobile number because U2 has the right to

(see #1478)

External user 1 E1 with number M1 eg. 01230041304
External user 2 E2 with number M2 eg. 01230041306
External user 3 E3

Create the following rules

In callisto.01 U2 must be provisioned on the same MDS where is the trunk to xivo-net (see bug #2494). 

U2 activates unconditional forward towards M1 - 01230041304

• on phone, dial : *21<M1>

U2 calls E1

E1 is ringing

U1 calls E1

U1 hears "Sorry, you don't have the rights to call this number"

U1 calls U2

E1 is ringing (with #1478)

U3 calls U2

E1 is ringing

E3 calls U2 via its DID

E1 is ringing (with #1478)

U2 deactivates unconditional forward

• on phone, dial : *20

U2 activates unconditional forward towards M2 - 01230041306

• on phone, dial : *21<M2>

U2 calls E2

U2 hears "Sorry, you don't have the rights to call this number"

U1 calls U2

U1 hears "Sorry, you don't have the rights to call this number"

U3 calls E2

E2 is ringing

U3 calls U2

U1 hears "Sorry, you don't have the rights to call this number" (with #1478)

E3 calls U2 via its DID

E3 hears "Sorry, you don't have the rights to call this number"

Agent A1 calls an internal user, and then hangup after user answered.

Agent A1 logged in and in Ready state

Agent A1 calls internal user U1

Agent A1 state is Dialing

U1 answers

Agent A1 state is On call - outgoing - (acd false)

A1 hangs up

Agent A1 state is Ready

Check agent states on acd outgoing calls

and check callerid

A user U1 with custom outgoing callerid

An agent A1 member of an outbound queue (queue name has to start by out, see documentation)

Agent A1 logged on U1

Calls must be started from CC Agent (not from device)

A1 dials an outbound acd call (with 6 or more numbers)

A1 hangup

A1 state is dialing -> Ready

Call control shows that "outbound" is called 

A1 dials an outbound acd call

Destination answers

A1 hangs up

On xivo, run the command watch -n 0.1 --color "asterisk -rx 'queue show outbound'"

Dialing -> Outbound acd call -> A1 ready 

On xivo, you should see the call number changed with now C:1 (instead of C:0 when no calls were passed yet).

A1 dials and oubound call

A1 pause

Destination answers

A1 hangs up

Dialing -> Outbound acd call -> A1 paused

On xivo, you should see the call number increased.

A1 ready

A1 dials an outbound call

Destination answers

A1 pause

A1 hangs up

Dialing -> Outbound acd call -> A1 paused

On xivo, you should see the call number increased.

A1 on pause,

A1 dials an outbound call

Destination answers

A1 hangs up

Pause -> Dialing -> Outbound call NON acd  -> A1 paused

This call should be a direct call as the agent is NotReady on ACD. On xivo, the call number shouldn't increase.

A1 ready

A1 place an outbound call

Destination answers

Destination sees customized Outoging caller id of U1

Agent A1 in Ready state

Agent A1 is logged in queue Q1

Call enters the queue Q1

Agent A1 state is Ringing

Agent A1 answers call

Agent A1 state is On call - incoming - acd true

Caller hangs up

Agent A1 state is Ready

Test agent states with every (or almost) transfer cases with a Yealink phoneset.

There is a common pattern for these tests :

 a call which answered on A1's phone set which he transfers to A2 (either directly or via another queue, either with the phone set ot via the ccagent)

You will need :

One user U1

Q1 without wrapup

Q2 with wrapup

Two agents A1 and A2 that can be logged in these queues.

Being logged to CCManager and CCagent

A1 is logged on Yealink phoneset

A1 logged in Q1

A2 logged in Q1

DESTXFER is Q1

A1 uses its phone set for the actions

A1 is Ready

A2 is Ready

U1 calls Q1

A1 is Ringing

A1 answers

A1 is on ACD call

A1 init att. transfer to DESTXFER

A2 is ringing

A2 answers

A2 On call (incoming or ACD)

A1 complete transfer

A1 ready

A2 On call (incoming or ACD)

A2 hangup

A2 is Ready

A1 logged in Q1

A2 logged in Q1

DESTXFER is Q1

Reproduce steps 2 to 7 but with A1 using the agent application to transfer

States must be correct

A1 logged in Q1

A2 logged in Q1

DESTXFER is Q1

Reproduce steps 2 to 7 but with A1 (number associated to line) using

• agent application to init transfer
• phone set to complete transfer

States must be correct

A1 logged in Q1

A2 logged in Q1

DESTXFER is A2  (number associated to line)

Reproduce steps 2 to 7 but with A1 using phone set

States must be correct

A1 logged in Q1

A2 logged in Q1

DESTXFER is A2  (number associated to line)

Reproduce steps 2 to 7 but with A1 using

• agent application to init transfer
• phone set to complete transfer

States must be correct

Redo all the test with :

A1 logged in Q2 (queue with wrapup)

A2 logged in Q2 (queue with wrapup)

And replace DESTXFER Q1 by DESTXFER Q2 when appropriate

States must be correct

(note that you must have a Wrapup state between On call and Ready)

A2 may not be in wrapup in some cases, see #1908

Test agent states with every (or almost) transfer cases with a WebRTC user.

There is a common pattern for these tests :

 a call which is answered on A1's webrtc line which he transfers to A2 (either directly or via another queue via the ccagent)

You will need :

One user U1

Q1 without wrapup

Q2 with wrapup

Two agents A1 and A2 that can be logged in these queues.

Being logged to CCManager and CCagent

A1 is logged on WebRTC line

A1 logged in Q1

A2 logged in Q1

DESTXFER is Q1

A1 uses its ccagent for the actions (A1 is logged on webrtc line)

A1 is Ready

A2 is Ready

U1 calls Q1

A1 is Ringing

A1 answers

A1 is on ACD call

A1 init att. transfer to DESTXFER

A2 is ringing

A2 answers

A2 On call (incoming or ACD)

A1 complete transfer

A1 ready

A2 On call (incoming or ACD)

A2 hangup

A2 is Ready

A1 logged in Q1

A2 logged in Q1

DESTXFER is A2  (number associated to line)

Reproduce steps 2 to 7 A1 using its ccagent for the actions (A1 is logged on webrtc line)

States must be correct

Redo all the test with :

A1 logged in Q2 (queue with wrapup)

A2 logged in Q2 (queue with wrapup)

And replace DESTXFER Q1 by DESTXFER Q2 when appropriate

States must be correct

(note that you must have a Wrapup state between On call and Ready)

A2 may not be in wrapup in some cases, see #1908

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

Configuration: Login as "avencall" with password "superpass" using https://XIVO_IP/configmgt URL

You should be logged in and see list of users

Configuration: Type a letter into search field on the right side

Drop-down list of XiVO users appears

Configuration: Through fingerboard open configuration

Login as "avencall" with password "superpass"

You should be logged in and see list of users

Configuration: Type a letter into search field on the right side

Drop-down list of XiVO users appears

Configuration: Select a user that has XivoCC login and password defined
(in Xivo configuration > IPBX settings > Users > XiVO Client section)

Add him (with "+" button)

Select Superviseur profile, assign him to a queue and confirm

The user appears in the list of users

Recording: Try login as user that has login and password defined in Xivo configuration, but he is not in CC Configuration

User is recognized, but can't login

Message "Aucun droit d'accès n'existe pour vous" appears

Call a queue, answer, talk to the microphone

Recording: Login as the newly added user

You are logged in

Recording: Open the first tab - Liste des enregistrements

You should see list of call records

Recording: Check the most recent call records (on top)

Last record should be recent

Records should have different length

If the topmost record has zero length, refresh the page

Recording: Download a recent record

Wav file is downloaded

Recording: Open the file in default media player

Test record is being played
(although calls from loadtester may not contain any sound)

Configuration: Refresh the page to see if you are still logged in. Login if not

Delete the newly added user

The user disappears from the list of users

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

Agents A1 and A2, both assigned to Q1, not logged in to the agent interface 

Agent A1  put the username and password and click to the button "Se connecter"

Agent A1 is logged in the Agent interface. A1 state is Ready (green icon).

Agent A1 is logged.

Agent A1 sees his queue and calls in queue. A1 state is Ready (green icon).

In CCManager agent view check value of statistic "Tot. Pause"
Agent A1 change his state Ready into Pause.

A1 state is Paused (red icon).

Agent A1 change his state Pause back into Ready.

A1 state is Ready (green icon).

In CCManager verify that statistic "Tot. Pause" was incremented.

Agent A2 is logged.

In Activities tab move cursor on queue Q1 on Status column and verify that number of Avail. agents increased

Logout A2

In Activities tab move cursor on queue Q1 on Status column and verify that number of Avail. agents decreased

Checks the Kibana panels

Xivo and XivoCC installed, at least one agent in a queue and some calls passed

Open Kibana in a browser

Kibana welcome page is shown

Modify the URL following this pattern:

http://XIVOCC_IP/kibana/#/dashboard/file/queues.json

You should see some calls in following widgets:

Checks SpagoBI and statistics using sample reports

Xivo and XivoCC installed, at least one agent in a queue, some calls passed

Sample reports installed following the procedure described in the documentattion:

https://documentation.xivo.solutions/en/latest/xivocc/install_and_config/installation/installation.html#spagobi

Open the SpagoBI and log in using biadmin/biadmin

You should be logged in

Open the Documents development item from the sidebar  (directory icon, the name is shown in a tooltip), then go to system directoty and open the data status report by clicking on it

You should get a rapport called Etat de la base des statistiques

in this rapport you have to check that dates for Dernier queuelog, Dernier cel, Call data and Call on queue correspond to the last call passed on the xivo and for the other tables the time should be at more half an hour before "now", means if you generated the report at 10.10, the time should be 9.30 or 9.45

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

Xivo and XivoCC installed, some calls passed by user U1

Login to the xivo assistant as user U1

You should be logged in

Open the call history

You should see calls passed by user U1 with numbers translated to names for xivo users.

Manager can login, see queues, states of agents, calls and other activity at call center.

Xivo and XivoCC installed

Agents A1 and A2 in queue Q1

Xivo user U1

A1 login to CC Agent

A1 is logged in

A2 login to XiVO CC Manager (from second browser)

A2 is logged in

Select Vue Globale

You should see queue list with assigned agents

A1 and A2 are ready

U1 calls a queue

Agents' states should change to ringing

Select Vue FA, wait 7 seconds

Longest waiting time is being incremented

A1 answers

Longest waiting time is 0

Select Vue Globale

A1 is on phone, A2 is available

U1 hangs up

A1 and A2 are ready

Listen Agent and test DTMF listening action using CCManager and CCAgent

A supervisor logged in CCManager

An agent loggee in CCAgent

Agent is on call

Supervisor start listening to the agent call using the toolbox in CCManager

Supervisor listens to the agent, agent does not hear supervisor

Listen indicator in the CCAgent lights in red.

Supervisor press 5

Agent hears the supervisor

Supervisor press 6

Client and agent hear the supervisor

Supervisor press 4

Nobody hears the supervisor

Supervisor hangs up

Agent is not listened anymore, the listen indicator in ccAgent goes back to grey.

Test if manager can see Base config. column and it refreshes when settings are changed.

Documentation: https://documentation.xivo.solutions/en/latest/xivocc/features/ccmanager/ccmanager.html#ccmanager-base-configuration-filter

Xivo and XivoCC installed

Queues Q1 and Q2

Agents A1 and A2 are in both queues

Open Agent view (tab) in CCManager

You see both agents

You see Base config. column with 'n/a' values

Edit A1

Set Penalty for both queues to 0

Check that Default column is grey

If column Default is not grey, remove agent from both queues and save.

Save

Then re-add agent to both queues in Xivo web interface.

A1 shoukld be in Q1 and Q2 with penalty to 0 and without any Base configuration.

Repeat previous step with A2

Select both agents by checkbox

Open drop down menu

Click Create base configuration

‘Create base configuration’ dialog window appears

Add Q1 with penalty 1

Add Q2 with penalty 2

Save

Dialog window closes

Base config. column values are 'Different'

Edit A1

Click Set current

Base configuration is apllied to Penalty column

Save

Dialog window closes

Base config. column value for A1 is 'Ok'

Base config. column value for A2 remains 'Different'

Aim is to validate behavior of queue recording switch option

At least 3 queues Q1, Q2, Q3

One agent (supervisor) that have access to CC manager and recording access

Connect to XiVO PBX and check that all queues have recording set to

• Activate: Not activated
• Mode: Not recorded

All queues should be set to not recorded

Connect to CC Manager, check that no recording option are displayed on the first 3 menus (global view, group view and queue view) and then logout

No switch appears in left menu

and no visual indicator next to queue name in Global view

Edit in XiVO PBX to add recording mode "Recorded" to Q1 but not activated and then login again in ccmanager

Check in CC manager that recording option switch appear in left menu in off mode

Click on switch button in CC manager to enable recording.

Button must be switched on and should have the same status on all first three menus.

Visual indicator next to queue name must be present and in the correct state

Check on XiVO PBX that recording is activated on Q1.

Checkbox is enabled

Edit in XiVO PBX Q2 and Q3 to set recording mode to 'Recorded on demand' but let them not activated. Check switch Status on CC manager

Switch status must be in intermediate position as all queues are not activated.

Visual indicator next to queue name must be present and in the correct state

Click on switch to activate them all. Check on XiVO the queues recording state

Switch must be set on. In XiVO all three queues are activated

Click again on switch.

Switch must be set off. In xivo all queues are deactivated

Stop config-mgt docker container, stay connected with your agent in CC manager, click in menu to change view

An error message must be displayed at top in CC manager to notify that config server is ko.

Switch must be hidden.

The goal is to verify that an administrator or a supervisor with "dissuasion" right can :

- list,

- change,

the dissuasion destination of a queue from CC Manager and that it actually works :-)

Two queues :

• Q1 - support
• Q2 - sales

A user U1 to call the queues

An agent A1 logged in Q2

One administrator or supervisor to access the ccManager

Configure the sound files:

Note: to be displayed the sound file MUST BE prefixed with the queue name. Below we will use the names support for Q1 and sales for Q2. Replace them accordingly to use your queue's name.

Go to the sound directory:

cd /var/lib/xivo/sounds/playback

Create real file for queue Q1 (support) :

cp /usr/share/asterisk/sounds/en_US/tt-monkeys.wav support_monkeys.wav

Remove files for queue Q2 (sales) :

rm sales_*

Give correct rights:

xivo-fix-paths-rights

Verify queue configuration

In webi go to IPBX -> Call Center -> Queues

Edit Q1, go to Advanced tab and verify that Join an empty queue parameter has the Unavailable in the items selected

Configure the queuedissuasion default queue:

In xivo add in your custom.env

QUEUEDISSUASION_DEFAULT_QUEUE=sales

Recreate the containers (configmgt)

xivo-dcomp up -d

Give the right to the manager to access the dissuasion

Login to the config-mgt interface : 

User : avencall 

Password : superpass

Edit agent A1 ->Profile : supervisor 

     - Update dissuasions : checked  

     - Queues : Q1 and Q2 checked 

     - Groups : At least one group checked (can be default)

Log in to the CCManager with the supervisor profile set up previously (with access to dissuasion). 

On the top right corner, select Q1 and Q2. 

You are logged in. 

Q1 and Q2 are selected and displayed on the global view. 

On the global view, make sure that :

- No agent is logged in Q1

- One agent A1 is logged in Q2.

On the global view, click on the name of Q1 (Support).  

A modal appears with some information about the queue and a queue configuration tab. 

On the queue configuration tab, click on the title Failed destination and select the monkeys sound file.

The dropdown closes

U1 calls Q1

U1 listens to the monkeys. 

From the failed destination dropdown, select Q2 (sales).

U1 calls Q1 

A1 rings (from Q2)

On the config-mgt interface : 

Edit A1 to remove their access to dissuasion

-> profile Supervisor 

Update dissuasion : UNCHECKED 

Q1 and Q2 checked 

At least one group checked. 

Refresh the CCManager page

The supervisor is connected to the CCManager interface. 

Click on the name of Q1 (Support).

The modal appears with only one page displaying some queue details and no configuration tab. The dissuasion is not displayed. 

On the config-mgt interface : 

Edit A1 to give them access to dissuasion

-> profile Administrator 

Refresh the CCManager page

Click on the name of Q1

The configuration tab is available again and you can see and change the dissuasion. 

Be able to take process a callback and close the callback

An agent and a manager, callbacks enabled, a queue with callbacks available (import callbacks from callback sample file)

Process 4 callbacks and choose a different status (except reschedule)

Enter a comment

Callbacks are not to be processed in CCAgent

Callbacks are marked clotured in CCManager

Check tickets are correctly updated in CCManager by downloading the file

Agent can search users by name.

• while not on call, search result offers to call user
• while on call, search result offers to transfers (direct and attended) to user

• agent A logged into CCAgent
• users U1, U2

1. Make user U1 available
2. Type first letters of U1's name + Enter into search box

Search result appears with uset U1 (and maybe other users)

Mouse over U1's status

"Call" action is displayed

Click "Call" action

U1 is called by A

U1 accepts

There is ongoing call between a and U1

Type first letters of U2's name + Enter into search box

Search result apperars with U2 (and maybe other users)

Mouse over U2's status

"Call" action appears

Click "Call"

U2 rings

U2 accepts

there is call between A and U2, u1's call is on hold

U2 hang up

there is call on hold between A and U1 (state On call)

Type first letters of U2's name + Enter into search box

Search result apperars with U2 (and maybe other users)

Call U2

"Transfer" actions appears in callbox panel

Click "Transfer"

U2 rings, A is ready with no calls (state Ready)

Validate webRTC integration on the agent interface

Have a user with associated agent with webrtc line, a queue with this agent connected, with autopause agent false

Login with the user with webrtc line

Login success

Make an ougoing call

You have the ringback tone and once answered the callee

Put the call on hold

The call is on hold

Resume the call

The call is resumed

Hangup the call by the button on the interface

The call is terminated

Make a call to a queue where the agent is connected

The call is distributed to the agent.

Reject the call

The call is rejected.

Wait till call arrives once more to the agent, answer the call

The call is answered

Hangup the call by the button

The call is terminated

Login with the user with web rtc line on HTTP protocol (thus so without SSL)

Login is refused with an error

Login agent with a fixed phone

Login success

Customer call a queue and exit by pressing a key, when key pressed call is directed to a dialplan which simulates IVR

When customer calls back, customer history has to be displayed (Bug https://projects.xivo.solutions/issues/1223)

A caller

An agent logged in a queue Q1

A queue Q2 with exit with key configured :

• Menu service->IPX->Configuration FIle add a file exit_with_pound with the folowing contents

[exit_with_pound]
exten = #,1,NoOp(## User exited by pressing pound ##)
same = n,SayDigits(765)
same = n,hangup()

Add an internal context exit_with_pound

In queue advanced configuration add this context as an exit context

When calling the queue should be able to get out the queue by pressing the # key and hear message "7..6..5"

Caller calls Q2 and press the # key

message 7..6..5

Caller calls Q1, agent rings

Agent should see the customer call history and context in CCAgent

The goal is to verify that you can :

- list,

- change,

the dissuasion destination of a queue from CC Agent and that it actually works :-)

Two queues :

• Q1 - support
• Q2 - sales

One agent A1 belonging to Q1

One agent A2 belonging to Q2

A user U1 to call the queues

Configure the sound files:

Note: to be displayed the sound file MUST BE prefixed with the queue name. Below we will use the names support for Q1 and sales for Q2. Replace them accordingly to use your queue's name.

Go to the sound directory:

cd /var/lib/xivo/sounds/playback

Create real files for queue Q1 (support) :

cp /usr/share/asterisk/sounds/en_US/tt-monkeys.wav support_monkeys.wav

cp /usr/share/asterisk/sounds/en_US/tt-weasels.wav support_weasels.wav

Remove files for queue Q2 (sales) :

rm sales_*

Give correct rights:

xivo-fix-paths-rights

Verify queue configuration

In webi go to IPBX -> Call Center -> Queues

Edit Q1, go to Advanced tab and verify that Join an empty queue parameter has the Unavailable in the items selected

Configure the queuedissuasion default queue:

In xivo add in your custom.env

QUEUEDISSUASION_DEFAULT_QUEUE=sales

Recreate the containers (configmgt)

xivo-dcomp up -d

Give the right to agent to access the dissuasion

Login to the config-mgt interface : 

User : avencall 

Password : superpass

Edit agent A1 ->Profile : supervisor 

     - Update dissuasions : checked  

     - Queues : Q1, Q2 and Q3 checked 

     - Groups : At least one group checked (can be default)

A1 clicks on the eject icon on Q1 and select monkeys sound file

The dropdown closes

A1 logs out from CC Agent

There must be no other agent logged in

U1 calls Q1

U1 listen to the monkeys

U1 hangup

A1 logs in CC Agent again

U1 calls Q1

A1 rings

U1 hangup

A1 clicks on Q1 eject button and selects queue Q2

A1 logs out

There must be no other agent logged in Q1

U1 calls Q1

A2 rings (from queue Q2)

U1 hangup

A1 logs in CC Agent again

Click eject icon on Q1 and select weasels sound file

A1 logs out

There must be no other agent logged in Q1

U1 calls Q1

U1 listens to weasels

The goal is to verify that the dissuasion button is displayed only for the agents who are administrators, or in some cases, supervisors. It shouldn't be displayed in the following cases : 

- for agents with a teacher profile

- for supervisors who don't have the right to manage ECD 

You need : 

Agents A1, A2, A3

Login to the config-mgt interface : 

User : avencall 

Password : superpass

Edit agent A1 ->Profile : Administrator

Edit agent A2 ->Profile : Teacher 

     - Queues : Q1, Q2 and Q3 checked 

     - Groups : At least one group checked (can be default)

Edit agent A3 ->Profile : supervisor 

     - Update dissuasions : Unchecked 

     - Queues : Q1, Q2 and Q3 checked 

     - Groups : At least one group checked (can be default)

Agent A1 logs in ccAgent 

In the activity table, A1 can see the Failed Destination button and can manage the dissuasion. 

A1 logs off.

A2 logs in ccAgent

In the activity tab, A2 cannot see the Failed Destination button.  

A2 logs off. 

A3 logs in ccAgent. 

In the activity tab, A3 can see Q1, Q2 and Q3 but cannot see the Failed Destination button. 

On the config-mgt interface : 

Edit A3 to give them access to dissuasion

-> profile Supervisor 

Update dissuasion : checked 

Q1 Q2 and Q3 checked 

At least one group checked. 

A3 logs off from ccAgent and logs in again. 

In the activity tab A3 can see Q1, Q2 and Q3, can also see the Failed Destination button and can change the dissuasion.   

Should be able to pause / unpause recording outgoing calls using the web agent interface

Agent is using the outbound acd call facility (Logged into an outbound queue)

Xivo-Recording installed,

Web Agent installed and working agent logged on

Recording server installed (xivocc/install_and_config/installation/installation.html#recording) and configured (xivocc/configuration/recording/recording.html)

Agent call external destination  using recording not paused on outgoing call

(using subroutine xivocc-outcall-recording)

Agent answers

Pause recording button should be displayed on the web agent

Say a number suite 0 1 2 3 etc

Pause recording using the button, restarts recording

End the call

Using the recording server, download and listen the record, should be paused

Call external destination using recording paused on outgoing call

Make sure the outgoing call is an acd call (color)

(using subroutine xivocc-outcall-recording-paused)

Agent answers

Start recording button should be displayed on the web agent

Say a number suite 0 1 2 3 etc

Start recording using the button, pause recording

End the call

Using the recording server, download and listen the record, should be paused

Should be able to pause / unpause recording incomig calls using the web agent interfac

Xivo-Recording installed,

Web Agent installed and working agent logged on

Recording server installed (xivocc/install_and_config/installation/installation.html#recording) and configured (xivocc/configuration/recording/recording.html)

Call agent  using recording not paused on incoming call

(using subroutine xivocc-incall-recording)

Agent answers

Pause recording button should be displayed on the web agent

Say a number suite 0 1 2 3 etc

Pause recording using the button, restarts recording

End the call

Using the recording server, download and listen the record, should be paused

Call agent  using recording paused on incoming call

(using subroutine xivocc-incall-recording-paused)

Agent answers

Start recording button should be displayed on the web agent

Say a number suite 0 1 2 3 etc

Start recording using the button, pause recording

End the call

Using the recording server, download and listen the record, should be paused