Page Comparison

To make any changes to config files, access them via SSH as root:

nano-tiny /etc/<sub-directory>/<config file name>

Call Groups

Check registration status of Call group members during call distribution

The feature is enabled by default and it prevents unavailable Call group members (means no registered devices or no push for mobile apps) from receiving calls from a queue. The logic is applied only for Call group calls!

...

• Reload SIP by running the command: 

  Code Block
  callweaver -rx 'sip reload'

  
  

Allow overriding of Global Call groups settings 

The option allows overriding of Global Call groups settings (see the chapter below) and saving custom parameters after each system upgrade.

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Collaboration 

Disable file transfer in Collaboration chat messages 

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COLLABORATION_FILE_TRANSFER_ALLOW=false

Disable Chat and Post-It functionality in Collaboration 

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Change the default Departments tree separator

Anchor

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departmentsastree

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Note:

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departmentsastree

Starting from WMS 5.

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By default, only 1 active Collaboration session is permitted. However, there is an option to allow the second Collaboration tab – in Call Control mode. This can be used, for example, for media devices when connecting via remote desktop. The option is particularly practical for teams4Wildix integration.

To activate this feature, add the following line to the file04.20220309.1, the Colleagues tab in Collaboration shows only the list of Departments and there is possibility to create a multilevel hierarchy of Departments and display them in a tree view. 

When indicating the Department path in WMS -> Users -> Edit User -> Department field, the default separator "/" should be used. For example: Department: UK/ Marketing.

To change the default separator to another character ("\" or "|"), edit the following parameter in the /rw2/etc/env.custom.ini: 

CALLDEPARTMENTS_CONTROLTREE_FEATURE=true

Once the feature is activated, an additional option appears on the Collaboration login window – Call Control only. Tick this checkbox off to log into Collaboration in Call Control mode.  

...

Starting from WMS 5.04.20220309.1, the Colleagues tab in Collaboration shows only the list of Departments and there is possibility to create a multilevel hierarchy of Departments and display them in a tree view. 

When indicating the Department path in WMS -> Users -> Edit User -> Department field, the default separator "/" should be used. For example: Department: UK/ Marketing.

...

SEPARATOR=\

Documentation: How to configure Departments tree.

Display all PBX users on the Colleagues tab 

Anchor
showallcolleagues showallcolleagues

By default, only people added to rosterare shown on the Colleagues tab. To display all users registered on the PBX, add the following line to the /rw2/etc/env.custom.ini file:

COLLABORATION_SHOW_ALL_COLLEAGUES=true

Enable Noise suppression on the PBX level

Starting from WMS 6.04, Collaboration users can enable Noise suppression in Collaboration Settings -> Web Phone. To enable Noise suppression on the PBX level for all users, edit the following parameter in the /rw2/etc/env.custom.ini file:

DEPARTMENTSCOLLABORATION_TREENOISE_SEPARATORSUPPRESSION=\true

Documentation: How to configure Departments tree.

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...

Collaboration Mobile App

Configure which numbers on mobile should be directed to native dialler

Starting from WMS v. 6.02.20230306.1, it is possible to configure that certain numbers dialled via mobile Collaboration app are automatically forwarded to native dialler (call via Mobile network) for all PBX users.  

Add the following parameter to /rw2/etc/env.custom.ini file:

COLLABORATION_SHOWMOBILE_ALLDIRECT_COLLEAGUES=true

Enable Noise suppression on the PBX level

...

CALL=xxx,yyy

 Where xxx, yyy are the extensions/ numbers, calls to which should be automatically forwarded to the native dialler. By default, the field is empty.

Configure Advanced settings in Collaboration app on mobile (Android)

Starting from Android app v. 5.13.05, it is possible to predefine on PBX the values for Advanced settings in Collaboration mobile app (Android). 

For this, set the following parameters in the /rw2/etc/env.custom.ini file:

...

...

Collaboration Mobile App

Configure which numbers on mobile should be directed to native dialler

Starting from WMS v. 6.02.20230306.1, it is possible to configure that certain numbers dialled via mobile Collaboration app are automatically forwarded to native dialler (call via Mobile network) for all PBX users.  

Add the following parameter to /rw2/etc/env.custom.ini file:

COLLABORATION_MOBILE_DIRECT_CALL=xxx,yyy

 Where xxx, yyy are the extensions/ numbers, calls to which should be automatically forwarded to the native dialler. By default, the field is empty.

Configure Advanced settings in Collaboration app on mobile (Android)

Starting from Android app v. 5.13.05, it is possible to configure which Advanced settings should be set in Collaboration mobile app (Android). 

For this, set the following parameters in the /rw2/etc/env.custom.ini file: 

...

More about advanced settings on mobile: Appendix 3: Advanced Settings

Devices

Modify devices sync

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More about advanced settings on mobile: Appendix 3: Advanced Settings

Devices

Modify devices sync

...

• Add the following parameters to the config file /rw2/etc/pbx/device_sync.conf 

  Code Block
  disable_sync_portal=1 disable_sync_manual=1

  Available values: 1 – sync is disabled; 0 – sync is enabled.

Modify g729 transcoding for web phone calls to trunks which do not support g711

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• Run the command: 

  Code Block
  callweaver -rx "sip reload"

  
  

Manage wideband codecs usage

Starting from WMS 5.04.20220309.1, there is custom provisioning parameter WideNarrowbandUse, which allows to manage wideband codecs usage. By default, the parameter is disabled and wideband codecs are forced for all devices. When the parameter is enabled (WideNarrowbandUse=yes), the previous behaviour remains: all devices use wideband codec priority or narrowband codec priority and remote codecs priority or local codecs.

...

• Run the command: 

  Code Block
  callweaver -rx "sip reload"

  
  

Modify the behaviour of Colleague BLF keys if a colleague is a Call group member

The feature is disabled by default . There are two options for its configuration:

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Select a specific GSM gateway 

Anchor
selectgsmgateway selectgsmgateway

The option allows setting a specific GSM gateway for SMS sending for each separate user:

• Edit the config file /etc/wildix/smsd-route.conf by specifying user extension and MAC address of GSM gateway, for example:

  Code Block
  101,9c7533014b00 102,9c7533014b00 103,9c7533014b00 104,9c7533014b01

  
  

Modify geolocation sharing on geolocation sharing on PBX

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Security

...

Configure FXS to recognize pulse dialing

It is possible to configure SSL connection for sending CDR data to external MySQL server. 

Upload the following three files to the PBX:

...

Make changes in the file /etc/callweaver/cdr_mysql.conf by adding/ editing the following rows: 

ssl_cert = /etc/callweaver/certs/client-cert.pem
ssl_key = /etc/callweaver/certs/client-key.pem
ssl_ca = /etc/callweaver/certs/ca.pem

Run the following command:

callweaver -x'reload cdr_mysql.so'

In the file /rw2/etc/ejabberd/ejabberd_mod_mam.yml, edit the following line:

sql_ssl: true

Enable location-based MFA 

...

The feature is disabled by default. To activate it, set the following parameter in /rw2/etc/env.custom.ini file:

MFA_SERVICE_ENABLE=true 

Dialplan

Exit code 0 from voicemail 

Support for exit code 0 from voicemail allowing caller to speak with an operator was added.

How to use:

• Add the letter ‘o’ as called number to the Dialplan context (that is where the “0” key sends the caller)

How to enable:

• Add the parameter operator=yes to the file voicemail.conf. It allows sender to hit 0 before/ after/ during leaving a voicemail to reach an operator

Use online streaming of mp3 http/ https sources as music on hold 

The feature is disabled by default. When enabled, it allows setting online streaming of mp3 http/ https sources as music on hold.

To enable:

• Add the following parameters to the file /rw2/etc/callweaver/musiconhold-stream.conf:

  Code Block
  [radio] mode=custom application=/usr/sbin/cw_play_http_stream.sh http://stream.104.6rtl.com/rtl-live/mp3-192

  where http://stream.104.6rtl.com/rtl-live/mp3-192 is your stream source.

Run the command: 

callweaver -rx "moh reload"

...

Add Dialplan application Set -> Music on hold -> radio

x-caracal

Change time range limit for reports

Starting from WMS 5.04, depending on the PBX load, some limits of allowed days range apply when generating x-caracal reports. Learn more in x-caracal documentation.

To set custom limit (for example 50 days), add the following parameter to /var/www/x-caracal/.env file:

MAX_ALLOWED_DAY_RANGE=50

Modify Inactivity Timer

By default, after 10 minutes of inactivity, x-caracal page needs to be refreshed. Starting from WMS 6.03, it is possible to modify its inactivity timeout.

To modify the timeout, add the following parameter to /var/www/x-caracal/.env file:

INACTIVITY_TIMEOUT_MINUTES = 120 

• Where 120 is the timeout in minutes (enter the preferred value)

The Maximum value is 600 (10 hours). In case you set the inactivity timeout to 0, it gets disabled. 

Integrations

Hotel PMS

WHoteld package supports some custom configuration parameters that can be changed by editing the file /etc/wildix/whoteld_manager_custom.conf.

List of the available parameters:

...

• Full - allows all synchronizations
• Forbid - denies any data synchronizations
• Lite - denies synchronization requests to FIAS/ XOpen interfaces

...

• std - accepts DND events only from FIAS interface
• extended - allows handling DND events also from PBX and XOpen interfaces

...

Supported values: 2 min - 6 min - 10 min (min - default - max) 

Integrate custom HTML code by using Google Tag Manager

Google Tag Manager allows integrating custom HTML code into Collaboration and x-caracal. To integrate the code, take the following steps

• Create a new config file template.conf including GTM ID in the file /rw2/etc/pbx/, for example, use the command: 

  Code Block
  echo 'gtm=GTM-12345678' > template.conf

  where 123445678 is Google Tag Manager ID.

•  After implementation, the following sections appear in the code: 

after the <HEAD> section:

<!-- Google Tag Manager -->
<script>(function(w,d,s,l,i){w[l]=w[l]||[];w[l].push({'gtm.start':
new Date().getTime(),event:'gtm.js'});var f=d.getElementsByTagName(s)[0],
j=d.createElement(s),dl=l!='dataLayer'?'&l='+l:'';j.async=true;j.src=
'https://www.googletagmanager.com/gtm.js?id='+i+dl;f.parentNode.insertBefore(j,f);
})(window,document,'script','dataLayer','GTM-XXXX');</script>
<!-- End Google Tag Manager -->

after the <BODY> section: 

<!-- Google Tag Manager (noscript) -->
<noscript><iframe src="https://www.googletagmanager.com/ns.html?id=GTM-XXXX"
height="0" width="0" style="display:none;visibility:hidden"></iframe></noscript>
<!-- End Google Tag Manager (noscript) -->

Feelingstream

Wildix supports integration with the Feelingstream platform for voice recognition and call recordings in stereo format. Feelingstream allows to conduct STT analysis of stereo recordings received from Wildix PBX.

Requirements:

• Activated Feelingstream account
• Quality of call recordings set to WAV (you can do this in WMS -> Dialplan -> General Settings)
• For HW/ VM: mounted storage for recordings

To enable the integration, create the config file /etc/feelingstream.conf with the following parameters: 

host=wildix.feelingstream.com
apikey=fcd4206d-aaab-56cb-bdc8-a33dc4611a4b_ab30ac63-1234-1234-1234-123456789012
bucket=call
lang=en

Where:

• host  – received from Feelingstream side
• apikey – generated using Feelingstream web interface
• bucket – created manually using Feelingstream web interface or API
• lang – optional parameter for setting the language for recognition. If not indicated, the language is set automatically.

To enable stereo mode on the PBX side and improve the quality of recognition, go to WMS -> Dialplan -> General Settings -> Set dialplan variables and add the following Global Dialplan variable: STEREO_RECORDINGS=yes.

...

to configure FXS to recognize pulse dialing to avoid an issue when phones using pulse dialing only could not make calls. For this, add the following parameter to the /rw2/etc/provision.conf file, indicating the MAC address of the necessary FXS device(s) and minimum and maximum flash hook time:

 UseFlashHookTime={"9C750430060a": {"min": 150,"max": 400},"9C71153200DC": {"min": 300,"max": 500}}

You can set as many devices as you need. 

Security

Enable SSL connection to remote MySQL server

It is possible to configure SSL connection for sending CDR data to external MySQL server. 

1. Upload the following three files to the PBX:
   

   Note
   Note: The files should be generated by server CA file on MySQL server. For MySQL v. 5.7, you can refer to the following instruction.

   • ca.pem
   • client-key.pem

   • client-cert.pem
     
     

2. Place the files into the directory /etc/callweaver/certs (the directory should be created beforehand) or any other directory on the PBX.

3. Make changes in the file /etc/callweaver/cdr_mysql.conf by adding/ editing the following rows: 
   

   Code Block
   ssl_cert = /etc/callweaver/certs/client-cert.pem ssl_key = /etc/callweaver/certs/client-key.pem ssl_ca = /etc/callweaver/certs/ca.pem

   
   

4. Run the following command:
   

   Code Block
   callweaver -x'reload cdr_mysql.so'

   
   

5. In the file /rw2/etc/ejabberd/ejabberd_mod_mam.yml, edit the following line:
   

   Code Block
   sql_ssl: true

   
   

Enable location-based MFA 

It is possible to enable the location-based multi-factor authentication that requests user to confirm IP address via email when logging in to Collaboration or using basic authorization from a new IP address.

The feature is disabled by default. To activate it, set the following parameter in /rw2/etc/env.custom.ini file:

MFA_SERVICE_ENABLE=true 

Dialplan

Exit code 0 from voicemail 

Support for exit code 0 from voicemail allowing caller to speak with an operator was added.

How to use:

• Add the letter ‘o’ as called number to the Dialplan context (that is where the “0” key sends the caller)

How to enable:

• Add the parameter operator=yes to the file voicemail.conf. It allows sender to hit 0 before/ after/ during leaving a voicemail to reach an operator

Use online streaming of mp3 http/ https sources as music on hold 

The feature is disabled by default. When enabled, it allows setting online streaming of mp3 http/ https sources as music on hold.

To enable:

• Add the following parameters to the file /rw2/etc/callweaver/musiconhold-stream.conf:

  Code Block
  [radio] mode=custom application=/usr/sbin/cw_play_http_stream.sh http://stream.104.6rtl.com/rtl-live/mp3-192

  where http://stream.104.6rtl.com/rtl-live/mp3-192 is your stream source.

• Run the command: 

  Code Block
  callweaver -rx "moh reload"

  
  

• Add Dialplan application Set -> Music on hold -> radio

x-caracal

Change time range limit for reports

Starting from WMS 5.04, depending on the PBX load, some limits of allowed days range apply when generating x-caracal reports. Learn more in x-caracal documentation.

To set custom limit (for example 50 days), add the following parameter to /var/www/x-caracal/.env file:

MAX_ALLOWED_DAY_RANGE=50

Modify Inactivity Timer

By default, after 10 minutes of inactivity, x-caracal page needs to be refreshed. Starting from WMS 6.03, it is possible to modify its inactivity timeout.

To modify the timeout, add the following parameter to /var/www/x-caracal/.env file:

INACTIVITY_TIMEOUT_MINUTES = 120 

• Where 120 is the timeout in minutes (enter the preferred value)

The Maximum value is 600 (10 hours). In case you set the inactivity timeout to 0, it gets disabled. 

Integrations

Hotel PMS

WHoteld package supports some custom configuration parameters that can be changed by editing the file /etc/wildix/whoteld_manager_custom.conf.

List of the available parameters:

• ReSyncType is used to modify data synchronization procedure. The following values are available:
  • Full - allows all synchronizations
  • Forbid - denies any data synchronizations
  • Lite - denies synchronization requests to FIAS/ XOpen interfaces
• DnDBehaviour -  setup for the DND event processing. The following values are available:
  
  • std - accepts DND events only from FIAS interface
  • extended - allows handling DND events also from PBX and XOpen interfaces
• PostponeRE- in case of using ASA hotel software, this option must be set to "true" to postpone processing of RE (Room equipment) request
• (removed to another file. See information below) BadgeTimeout - timeout for waiting on the badge programming response 

• Supported values: 2 min - 6 min - 10 min (min - default - max) 

  Note
  Note: the parameter "BadgeTimeout" is removed to /etc/wildix/whoteld_manager_wms.conf. The new default timeout is 60 seconds. New supported values: 20-600 seconds (min-max).

  
  

Integrate custom HTML code by using Google Tag Manager

Google Tag Manager allows integrating custom HTML code into Collaboration and x-caracal. To integrate the code, take the following steps

• Create a new config file template.conf including GTM ID in the file /rw2/etc/pbx/, for example, use the command: 

  Code Block
  echo 'gtm=GTM-12345678' > template.conf

  where 123445678 is Google Tag Manager ID.

•  After implementation, the following sections appear in the code: 

1. after the <HEAD> section:

   Code Block

   <!-- Google Tag Manager --> <script>(function(w,d,s,l,i){w[l]=w[l]||[];w[l].push({'gtm.start': new Date().getTime(),event:'gtm.js'});var f=d.getElementsByTagName(s)[0], j=d.createElement(s),dl=l!='dataLayer'?'&l='+l:'';j.async=true;j.src= 'https://www.googletagmanager.com/gtm.js?id='+i+dl;f.parentNode.insertBefore(j,f); })(window,document,'script','dataLayer','GTM-XXXX');</script> <!-- End Google Tag Manager -->

   
   

2. after the <BODY> section: 

   Code Block
   <!-- Google Tag Manager (noscript) --> <noscript><iframe src="https://www.googletagmanager.com/ns.html?id=GTM-XXXX" height="0" width="0" style="display:none;visibility:hidden"></iframe></noscript> <!-- End Google Tag Manager (noscript) -->

   
   

Feelingstream

Wildix supports integration with the Feelingstream platform for voice recognition and call recordings in stereo format. Feelingstream allows to conduct STT analysis of stereo recordings received from Wildix PBX.

Requirements:

• Activated Feelingstream account
• Quality of call recordings set to WAV (you can do this in WMS -> Dialplan -> General Settings)
• For HW/ VM: mounted storage for recordings

To enable the integration, create the config file /etc/feelingstream.conf with the following parameters: 

host=wildix.feelingstream.com
apikey=fcd4206d-aaab-56cb-bdc8-a33dc4611a4b_ab30ac63-1234-1234-1234-123456789012
bucket=call
lang=en

Where:

• host  – received from Feelingstream side
• apikey – generated using Feelingstream web interface
• bucket – created manually using Feelingstream web interface or API
• lang – optional parameter for setting the language for recognition. If not indicated, the language is set automatically.

To enable stereo mode on the PBX side and improve the quality of recognition, go to WMS -> Dialplan -> General Settings -> Set dialplan variables and add the following Global Dialplan variable: STEREO_RECORDINGS=yes.

Enable sending presence events to support Microsoft Teams 2-way presence sync 

Anchor
MS Teams presence sync MS Teams presence sync

For the correct user presence sync, it is necessary to enable sending of presence events to Analytics on PBXs with UC licenses (for x-bees licenses, it is enabled by default). For this, follow the steps described below:

• Create a directory /etc/systemd/system/pbx-data-engine.service.d 

  Code Block
  # mkdir /etc/systemd/system/pbx-data-engine.service.d

  
  

• Create a file /etc/systemd/system/pbx-data-engine.service.d/override.conf and add the following lines

  Code Block
  # vi /etc/systemd/system/pbx-data-engine.service.d/override.conf [Service] ExecStart= ExecStart=/usr/sbin/pbx_data_engine.py --daemon --mode calls presence

  
  

• Reload systemd and restart the service to apply the changes

  Code Block
  # systemctl daemon-reload # systemctl restart pbx-data-engine

  
  

• Run the following command to check what events are sent to Analytics

  Code Block
  echo mode | socat - UNIX-CONNECT:/var/run/data_engine/data_engine.sock

  
  

Fax Server

Adjust the resolution of outgoing faxes 

...

• Edit the config file /rw2/etc/faxglobal.conf by specifying the desired horizontal and vertical resolution of the image in pixels per inch. For example:

  
  

  Code Block
  XResolution=204 YResolution=391

  Other widespread resolutions: 204x98, 204x391, 408x391.

Disable Error Correction Mode (ECM) 

ECM is enabled by default. To disable it, add the following line in /etc/callweaver/res_fax_custom.conf:

ecm=no

Set maximum/ minimum fax transmission speed

Maximum transmission speed

To change the maximum transmission speed, modify the following parameter in the /etc/callweaver/res_fax_custom.conf file:

maxrate=14400=14400

The default value is 14400

Possible values: 2400 | 4800 | 7200 | 9600 | 12000 | 14400

Minimum transmission speed

To change the minimum transmission speed, modify the following parameter in the /etc/callweaver/res_fax_custom.conf file:

minrate=2400

The default value is 144002400

Possible values: 2400 | 4800 | 7200 | 9600 | 12000 | 14400

Minimum transmission speed

To change the minimum transmission speed, modify the following parameter in the /etc/callweaver/res_fax_custom.conf file:

minrate=2400

The default value is 2400

Possible values: 2400 | 4800 | 7200 | 9600 | 12000 | 14400

Kite

Deliver Kite chat requests to all Call group members 

In some cases, Kite chat requests may not be equally distributed within the group. 

To deliver chat requests to all Call group members simultaneously, edit the config file /rw2/etc/ejabberd/ejabberd_mod_wildix_kite.yml by changing the value of the pickup_strategy parameter:

pickup_strategy: sendall

Available values:

• onebyone (default) - sends Kite chat requests to Call group members one by one
• sendall - sends chat requests to all CG members simultaneously. 

Trunks

Enable Q-value (serial forking) for trunk registration

To enable q-value (serial forking) parameter via custom register string:

...

Copy registration line for a trunk from /etc/callweaver/sip-registration.conf (Example: register => 144?144:123456:”144″@10.168.0.144:5060)

...

Kite

Deliver Kite chat requests to all Call group members 

In some cases, Kite chat requests may not be equally distributed within the group. 

To deliver chat requests to all Call group members simultaneously, edit the config file /rw2/etc/ejabberd/ejabberd_mod_wildix_kite.yml by changing the value of the pickup_strategy parameter:

pickup_strategy: sendall

Available values:

• onebyone (default) - sends Kite chat requests to Call group members one by one
• sendall - sends chat requests to all CG members simultaneously. 

Trunks

Enable Q-value (serial forking) for trunk registration

To enable q-value (serial forking) parameter via custom register string:

• Copy registration line for a trunk from /etc/callweaver/sip-registration.conf (Example: register => 144?144:123456:”144″@10.168.0.144:5060)

• Uncheck Enable registration option in Trunk Settings (WMS -> Trunks)

• Add a new line into /etc/callweaver/sip-general-custom.conf: 

  Code Block
  register => 144?144:123456:”144″@10.168.0.144:5060~~0.6

  Where 0.6 is q-value.

• Run the command: 

  Code Block
  callweaver -rx "sip reload"

  
  

Modify sending keep-alive packets via UDP packets to keep RTP ports opened

The option improves symmetric RTP/ NAT by allowing keep-alive packets to be sent via UDP packets for PBXs located behind NAT. 

To send UDP packets (by default, RTP packets are sent), proceed with the following:

• Add the following parameter to the file /etc/callweaver/sip-general-custom.conf 

  Code Block
  rtpkeepalive_mode=udp

  Available values: udp | rtp.
  
  

• Run the command:
  

  Code Block
  callweaver -rx "sip reload"

  
  

Include the session-expires timer in UPDATE message

The feature is disabled by default. It is applied for some specific carriers when calls can drop due to the missing session-expires timer in UPDATE messages.

If you encounter such issue, follow these steps to include the session-expires timer:

• Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter: 

  Code Block
  register => 144?144:123456:”144″@10.168.0.144:5060~~0.6

  Where 0.6 is q-value

  session_expire_header_in_update=yes

  Available values: no – the feature is disabled; yes – the feature is enabled.

• Run the command: 

  Code Block
  callweaver -

rx

• x

"

• 'sip reload

"

Modify sending keep-alive packets via UDP packets to keep RTP ports opened

The option improves symmetric RTP/ NAT by allowing keep-alive packets to be sent via UDP packets for PBXs located behind NAT. 

...

• '

  
  

Allow calls from users with same public IP address as a static trunk

The option is disabled by default. To enable it and allow calls from users that have the same public IP address as a static trunk, do the following:

• Add the following parameter to the file /etc/callweaver/sip-general-custom.conf 

  Code Block
  rtpkeepalive_mode=udp

  Available values: udp | rtpbelow line to /etc/kamailio/cfg.d/host_specific_custom.cfg file: 
  

  Code Block
  modparam("pv", "varset", "allow_users_and_trunks_behind_same_nat=i:1")

  Available values: 1 - enabled; 0 - disabled.
  
  

• Run the command: 
  

  Code Block
  callweaversystemctl -rx "sip reload"

Include the session-expires timer in UPDATE message

The feature is disabled by default. It is applied for some specific carriers when calls can drop due to the missing session-expires timer in UPDATE messages.

If you encounter such issue, follow these steps to include the session-expires timer:

• Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter: 

  Code Block
  session_expire_header_in_update=yes

  Available values: no – the feature is disabled; yes – the feature is enabled.

Run the command: 

callweaver -x 'sip reload'

Allow calls from users with same public IP address as a static trunk

The option is disabled by default. To enable it and allow calls from users that have the same public IP address as a static trunk, do the following:

Add the below line to /etc/kamailio/cfg.d/host_specific_custom.cfg file: 

modparam("pv", "varset", "allow_users_and_trunks_behind_same_nat=i:1")

...

Run the command: 

systemctl restart kamailio.service

Users

2 PBXs in WMS Network, each with its own Active Directory for users

To make it work, proceed as follows:

1. Make import of users via Active Directory on Server PBX
2. Access Client PBX and move users from Server PBX to Client PBX
3. Enable Active Directory sync on Client PBX: connect as root via SSH to Client PBX and create the file /rw2/etc/ad_connect.conf
4. Copy the contents of the file ad_connect.conf from Server PBX to Client PBX

Result: Single Sign-On for Active Directory works for users on Client PBX.

Other 

Extend the limit of Voicemails per user

...

maxmsg=200

Where 200 is the desired number of allowed voicemails.

Change type of reconnection to the database in case connection was lost 

It is possible to change type of reconnection to the database in case the connection was lost. For this, add the following custom parameter to /etc/callweaver/cdr_mysql.conf configuration file:

reconnect_mode=ping

The “reconnect_mode” parameter has the following values:

• init: reinitializes the connection. This value is used by default. 
• ping: restores the connection by sending a ping request

  restart kamailio.service

  
  

Users

2 PBXs in WMS Network, each with its own Active Directory for users

To make it work, proceed as follows:

1. Make import of users via Active Directory on Server PBX
2. Access Client PBX and move users from Server PBX to Client PBX
3. Enable Active Directory sync on Client PBX: connect as root via SSH to Client PBX and create the file /rw2/etc/ad_connect.conf
4. Copy the contents of the file ad_connect.conf from Server PBX to Client PBX

Result: Single Sign-On for Active Directory works for users on Client PBX.

Other 

Extend the limit of Voicemails per user

By default, the maximum number of Voicemails per user is 100. Starting from WMS 6.01.20220621.2, it is possible to extend the number. For this, change the value of the following parameter in the/etc/callweaver/voicemail-custom.conf file:

maxmsg=200

Where 200 is the desired number of allowed voicemails.

Change type of reconnection to the database in case connection was lost 

It is possible to change type of reconnection to the database in case the connection was lost. For this, add the following custom parameter to /etc/callweaver/cdr_mysql.conf configuration file:

reconnect_mode=ping

The “reconnect_mode” parameter has the following values:

• init: reinitializes the connection. This value is used by default. 
• ping: restores the connection by sending a ping request
  

  
  

Avoid no media issue on some iOS versions

On some iOS versions, there is an occasional issue of in which there can be no media after answering a call, in case the phone was locked and it is an API call by an external CRM or a call from Collaboration with Any selected as a device. If you encounter such an issue, to avoid it, you can add the following line to the /rw2/etc/env.custom.ini file:

ORIGINATE_VIA_ANY_DEVICE_TRANSFER=false

Prevent obsolete repeaters from upgrade

Starting from W-AIR firmware v.07.30.xx.xx and higher, old Wildix repeaters are no longer supported and need additional configuration to stay on previous firmwares to continue working. For old repeaters to work, you need to add the following parameter to the /rw2/etc/provision.conf file:

SupportObsoleteRepeater = yes 

The default value is “no”. When the parameter is enabled, the base stations automatically switch to the previous firmware on which the obsolete repeaters are supported (v. 06.xx.xx.xx)  

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