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The List of custom config parameters that allow changing various settings in config files. Created: August 2018 Updated: October 2023 Permalink: https://wildix.atlassian.net/wiki/x/Fh3OAQ |
Table of Contents |
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To make any changes to config files, access them via SSH as root:
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Note |
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Collaboration
Disable file transfer in Collaboration chat messages
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Change the default Departments tree separator Anchordepartmentsastree departmentsastree
departmentsastree | |
departmentsastree |
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.
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Note |
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The support starts from WMS 5.04.20220309.1 |
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:
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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
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Note |
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Note: Starting from WMS 6.04.20231020.2, it is possible to manage location-based MFA via WMS -> PBX -> Features. Documentation: WMS Settings Menu - Admin Guide |
Dialplan
Exit code 0 from voicemail
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Note |
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Note: Starting from WMS 5.04.20220309.1, in case you have several streams, modify the parameter [radio] to [radio1], [radio2], etc. in the config file and choose the corresponding option ("radio1", "radio2") when setting music on hold in the Dialplan. |
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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.
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Note |
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Note: Stereo recordings can be activated and used separately on any PBX, without Feelingstream integration, consult the documentation Custom Global Dialplan Variables List. |
Fax Server
Adjust the resolution of outgoing faxes
Note |
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Note: The default resolution is 204x196 (fine). |
To adjust the resolution:
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Code Block |
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XResolution=204 YResolution=391 |
Other widespread resolutions: 204x98, 204x391, 408x391.
Disable Error Correction Mode (ECM)
Note |
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Note: The support starts from WMS 5.04.20220309.1. |
ECM is enabled by default. To disable it, add the following line in /etc/callweaver/res_fax_custom.conf:
Code Block |
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ecm=no |
Set maximum/ minimum fax transmission speed
Note |
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Note: The support starts from WMS 5.04.20220309.1. |
Maximum transmission speed
To change the maximum transmission speed, modify the following parameter in the /etc/callweaver/res_fax_custom.conf file:
Code Block |
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maxrate=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:
Code Block |
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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
Note |
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The support starts from WMS 5.03. |
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:
Code Block |
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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 |
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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 |
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rtpkeepalive_mode=udp |
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Run the command:
Code Block |
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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:
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Enable sending presence events to support Microsoft Teams 2-way presence sync Anchor MS Teams presence sync MS Teams presence sync
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 -cr
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
Note |
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Note: The default resolution is 204x196 (fine). |
To adjust the resolution:
- 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)
Note |
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Note: The support starts from WMS 5.04.20220309.1. |
ECM is enabled by default. To disable it, add the following line in /etc/callweaver/res_fax_custom.conf:
Code Block |
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ecm=no |
Set maximum/ minimum fax transmission speed
Note |
---|
Note: The support starts from WMS 5.04.20220309.1. |
Maximum transmission speed
To change the maximum transmission speed, modify the following parameter in the /etc/callweaver/res_fax_custom.conf file:
Code Block |
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maxrate=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:
Code Block |
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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
Note |
---|
The support starts from WMS 5.03. |
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:
Code Block |
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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 by adding the parameter:
Available values: no – the feature is disabled; yes – the feature is enabledCode Block session_expire_header_in_update=yes
register => 144?144:123456:”144″@10.168.0.144:5060~~0.6
Where 0.6 is q-value.
Run the command:
Code Block callweaver -
xrx
'"sip reload
'
Allow calls from users with same public IP address as a static trunk
Note |
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Note: The support starts from WMS 6.01.20221019.4. |
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"
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 below line to the following parameter to the file /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 - disabledcallweaver/sip-general-custom.conf
Code Block rtpkeepalive_mode=udp
Available values: udp | rtp.
Run the command:
Code Block systemctl restart kamailio.service
Users
2 PBXs in WMS Network, each with its own Active Directory for users
Note |
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You need admin access to Active Directory server. |
To make it work, proceed as follows:
- Make import of users via Active Directory on Server PBX
- Access Client PBX and move users from Server PBX to Client PBX
- Enable Active Directory sync on Client PBX: connect as root via SSH to Client PBX and create the file /rw2/etc/ad_connect.conf
- 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.
Note |
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Limitation: the option "Remove existing users which are not received from the backend" does not work in this configuration; do not enable it. |
Other
Extend the limit of Voicemails per user
...
Code Block |
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maxmsg=200 |
Where 200 is the desired number of allowed voicemails.
Note |
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Notes:
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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:
Code Block |
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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
...
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 -cr
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.sockcallweaver -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:
Code Block callweaver -x 'sip reload'
Allow calls from users with same public IP address as a static trunk
Note |
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Note: The support starts from WMS 6.01.20221019.4. |
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:
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 systemctl restart kamailio.service
Users
2 PBXs in WMS Network, each with its own Active Directory for users
Note |
---|
You need admin access to Active Directory server. |
To make it work, proceed as follows:
- Make import of users via Active Directory on Server PBX
- Access Client PBX and move users from Server PBX to Client PBX
- Enable Active Directory sync on Client PBX: connect as root via SSH to Client PBX and create the file /rw2/etc/ad_connect.conf
- 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.
Note |
---|
Limitation: the option "Remove existing users which are not received from the backend" does not work in this configuration; do not enable it. |
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:
Code Block |
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maxmsg=200 |
Where 200 is the desired number of allowed voicemails.
Note |
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Notes:
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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:
Code Block |
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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
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