Custom config parameters List
The List of custom config parameters that allow changing various settings in config files.
Created: August 2018
Updated: January 2026
Permalink: https://wildix.atlassian.net/wiki/x/Fh3OAQ
- 1 Call Groups
- 2 Legacy Collaboration
- 2.1 Disable file transfer in Legacy Collaboration chat messages
- 2.2 Disable Chat and Post-It functionality in Legacy Collaboration
- 2.3 Change the default Departments tree separator
- 2.4 Display all PBX users on the Colleagues tab
- 2.5 Enable Noise suppression on the PBX level
- 2.6 Enable Relay candidates only for all PBX users
- 2.7 Enable Presence events sending for Webhook integration
- 3 Legacy Collaboration Mobile App
- 4 x-bees
- 5 Devices
- 5.1 Modify devices sync
- 5.2 Modify g729 transcoding for web phone calls to trunks which do not support g711
- 5.3 Modify HD codecs on PBX
- 5.4 Manage wideband codecs usage
- 5.5 Modify presence status monitoring via BLF keys
- 5.6 Modify the behaviour of Colleague BLF keys if a colleague is a Call group member
- 5.7 Modify direct SDES-SRTP
- 5.8 Select a specific GSM gateway
- 5.9 Modify geolocation sharing on PBX
- 5.10 Enable OPUS transcoding on PBX
- 6 Security
- 7 Dialplan
- 8 x-caracal
- 9 Integrations
- 10 Fax Server
- 11 Kite
- 12 Trunks
- 13 Users
- 14 Other
- 14.1 Extend the limit of Voicemails per user
- 14.2 Change type of reconnection to the database in case connection was lost
- 14.3 Avoid no media issue on some iOS versions
- 14.4 Prevent obsolete repeaters from upgrade
- 14.5 Enable transmit_silence parameter in PBX engine configuration
- 14.6 Allow Basic Auth for admin user
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!
To disable it:
Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter:
send_regevent_devstate = noAvailable values: no – the feature is disabled; yes – the feature is enabled.
Reload SIP by running the command:
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.
To override Global Call group settings, you need to edit queues-general-custom.conf which is included in file queues.conf.
Add a custom parameter, for example: autofill = yes (by default, the file queues.conf contains autofill = no parameter)
Run the following command:
callweaver -x "reload"
Global Call groups settings
Global Call group settings are defined and configured in [general] section of the configuration file queues.conf (the path to the file: /rw2/etc/callweaver/queues.conf).
[general] section
The section contains global settings that are applied to all Call groups.
persistentmembers = yes
With persistentmembers enabled, all dynamically added Call group members (via Feature code "Call group management" 97, WebAPI "Call group login" and Contact center feature in Collaboration) are stored in their Call groups and therefore saved after the system restart.
autofill = no
With autofill disabled, a Call group attempts to deliver calls to members in a serial manner. This means only one call is attempted to be distributed to members at a time. Additional callers are not distributed to members until that caller is connected to a member. With autofill enabled, callers are distributed to available agents simultaneously. The parameter allows you to more efficiently distribute calls between Call group members, especially if there are several callers in a queue and several members can accept a call.
General information regarding Call groups: WMS Start Guide
How to set up call distribution in Call groups: Call distribution in Call groups Admin Guide
Legacy Collaboration
Disable file transfer in Legacy Collaboration chat messages
This feature blocks the possibility of transferring files in Legacy Collaboration chat messages for all PBX users.
To disable file transfer, edit the file /rw2/etc/env.custom.ini by adding the following line:
COLLABORATION_FILE_TRANSFER_ALLOW=falseDisable Chat and Post-It functionality in Legacy Collaboration
This feature disables access to Chat and Post-It functionality throughout Collaboration. It removes the Chat tab and disables Chat and Post-It options in Colleagues, Fn keys, History tabs, as well as in Search and via Add button. With disabled chats, users also don't receive Kite chat requests.
To disable Chat and Post-It, edit the config file /rw2/etc/env.custom.ini by adding the following line:
COLLABORATION_CHAT_ALLOW=falseUse case: This feature is useful for teams4Wildix integration, as long as Microsoft Teams offers its own chat. Disabling chat functionality in Legacy Collaboration helps users to avoid confusion between these two systems and use one chat instead of two.
The Colleagues tab in Legacy 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:
DEPARTMENTS_TREE_SEPARATOR=\Documentation: How to configure Departments tree.
By default, only people added to roster are 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=trueCurrent limitation:
The status of users that are not added to the roster is not displayed.
Enable Noise suppression on the PBX level
Starting from WMS 6.04, Legacy Collaboration users can enable Noise suppression in Legacy 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:
COLLABORATION_NOISE_SUPPRESSION=trueNote: The settings are applied to all PBX users except for those, who enabled/ disabled Noise suppression in Legacy Collaboration settings.
Enable Relay candidates only for all PBX users
The support starts from WMS 6.06.20240606.1
The option "Relay candidates only” allows to avoid a rare issue when there may be no audio during the first call after PC restart. When enabled, the option switches the user to WebRTC relay (TCP) candidates configuration. By default, the feature is disabled. To enable the option for all PBX users, add the following parameter to /rw2/etc/env.custom.ini file:
COLLABORATION_WEBRTC_RELAYCANDIDATESONLY=trueFor the correct work of presence events of Webhook integration it is necessary to enable sending of presence events to Analytics on PBXs with UC subscription plan (for x-bees subscription, it is enabled by default). For this, follow the steps described below:
Create a directory /etc/systemd/system/pbx-data-engine.service.d
# mkdir /etc/systemd/system/pbx-data-engine.service.dCreate a file /etc/systemd/system/pbx-data-engine.service.d/override.conf and add the following lines
# vi /etc/systemd/system/pbx-data-engine.service.d/override.conf [Service] ExecStart= ExecStart=/usr/sbin/pbx_data_engine.py --daemon --mode calls presenceReload systemd and restart the service to apply the changes
# systemctl daemon-reload # systemctl restart pbx-data-engineRun the following command to check what events are sent to Analytics
echo mode | socat - UNIX-CONNECT:/var/run/data_engine/data_engine.sock
Legacy 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 Legacy Collaboration app are automatically forwarded to native dialler (call via Mobile network) for all PBX users.
Note: The support of the feature on mobile starts from Android Legacy Collaboration app v. 5.11; iOS Legacy Collaboration app v. 8.11.
Add the following parameter to /rw2/etc/env.custom.ini file:
COLLABORATION_MOBILE_DIRECT_CALL=xxx,yyyWhere 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 Legacy 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 Legacy Collaboration mobile app (Android).
Note: The support starts from WMS 6.04.20230724.1.
For this, set the following parameters in the /rw2/etc/env.custom.ini file:
Setting on mobile | Parameter | Default value | Possible values |
|---|---|---|---|
Disable SIP mode | COLLABORATION_MOBILE_DISABLE_SIP_MODE | false | true | false |
Enable WebRTC | COLLABORATION_MOBILE_ENABLE_WEBRTC | false | true | false |
Firewall bypass (beta) | COLLABORATION_MOBILE_FIREWALL_BYPASS | off | auto | on | off |
Outgoing calls via Wildix | COLLABORATION_MOBILE_OUTGOING_CALLS_VIA_WILDIX | false | true | false |
Run in background | COLLABORATION_MOBILE_RUN_IN_BACKGROUND | false | true | false |
User status when talking on mobile | COLLABORATION_MOBILE_USER_STATUS_WHEN_TALKING_ON_MOBILE | false | true | false |
Native Calls Support | COLLABORATION_MOBILE_NATIVE_CALLS_SUPPORT | false | true | false |
Mic gain | COLLABORATION_MOBILE_MIC_GAIN | 0 | -4..12 |
Activate DND mode | COLLABORATION_MOBILE_ACTIVATE_DND_MODE | false | true | false |
More about advanced settings on mobile: Appendix 3: Advanced Settings
x-bees
By default, a call to 911 is redirected to a native dialler. To disable redirection for 911 and enable it for custom number(s), add the following parameter to the /rw2/etc/env.custom.ini file:
COLLABORATION_MOBILE_DIRECT_CALL=number1,number2Where “number”1, “number2” are the numbers that should be redirected to a native dialler.
Note:
The support starts from x-bees app v. 1.95.1
If the parameter "COLLABORATION_MOBILE_DIRECT_CALL" contains an empty value, the default logic is used: a call to 911 goes through a native dialler.
Devices
Modify devices sync
Important: By default, the sync is enabled. It is necessary to disable it in case of FAILOVER scenario on Failover PBXs!
To disable the sync:
Add the following parameters to the config file /rw2/etc/pbx/device_sync.conf
disable_sync_portal=1 disable_sync_manual=1Available values: 1 – sync is disabled; 0 – sync is enabled.
Modify g729 transcoding for web phone calls to trunks which do not support g711
Important:
It’s not recommended to enable this feature as it reduces call quality and generates useless load on CPU!
It must be enabled only if the operator doesn’t support g711a/u for some calls
It can generate CPU overload and problems if too many calls use it; in this case it is recommended to use another operator which supports all the needed codecs ( g711a / g711u / g729)
Note: Feature is supported only on PBXs with modern CPU or Cloud.
To enable g729 transcoding:
Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter:
g729_transcoding=yesAvailable values: no – the feature is disabled; yes – the feature is enabled.
Run the command:
callweaver -rx "sip reload"
Modify HD codecs on PBX
Supported devices:
Collaboration
Android / iOS apps
WorkForce / WelcomeConsole / WP480 r3 / WP490 r3 / WorkForce 5 / ForcePro 5 / WelcomeConsole 5 / Start 5 / SuperVision 5
The feature also works for PBXs in WMS Network.
The feature is enabled by default. To disable it:
Edit the file /etc/callweaver/sip-general-custom.conf by adding the following parameters:
disallow=all allow=alaw:20,ulaw:20,g729:20,vp8,h264,h263,h263pRun the command:
callweaver -rx "sip reload"
Manage wideband codecs usage
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.
In case of WideNarrowbandUse=yes, you need to add relevant lines to /etc/callweaver/sip-wideband.conf to set the parameters
To Enable wideband codec usage for all networks:
forcewideband=yes disableautowideband=yesTo Enable wideband codec usage in LAN:
forcewideband=no disableautowideband=noTo set Networks where force usage of wideband codecs (i.e. for network 10.100.6.0/16):
forcewideband=no disableautowideband=yes wideband=10.100.6.0/16
Modify presence status monitoring via BLF keys
Detailed information about the feature: Presence status monitoring.
The feature is enabled by default. To disable it:
Edit the file /etc/callweaver/sip-general-custom.conf by adding the following parameter:
full_presence=noAvailable values: no – the feature is disabled; yes – the feature is enabled.
Run the command:
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:
1. Disable updates for "ringing" status for "Call All 10" strategy. To enable such behavior:
Edit the file /etc/callweaver/sip-general-custom.conf by adding the following parameter:
send_cg_member_ring_notify=yesAvailable values: no - the feature is disabled; yes - the feature is enabled.
Run the command:
callweaver -rx "sip reload"
Once it is enabled, Colleague BLF keys do not switch to an active state for incoming calls to Call Group before call answer (only Call group BLF turns on).
2. Disable updates for all early statuses including "ringing", "check who is calling", "cancelled" etc for "Call All 10/32" strategies (check the separate Article regarding the feature). To enable the behaviour:
Edit the file /etc/kamailio/cfg.d/host_specific_custom.cfg by adding the parameter:
modparam("pv", "varset", "skip_cg_members_presence_on_wp=i:1")Available values: 0 - the feature is disabled; 1- the feature is enabled.
Restart kamailio service:
/etc/init.d/kamailio restartEdit the file /etc/callweaver/sip-general-custom.conf by adding the parameter:
skip_cg_members_presence_on_wp=yesAvailable values: no - the feature is disabled; yes - the feature is enabled.
Restart callweaver service:
callweaver -rx "sip reload"
If the feature is enabled, the default devices mask is:
modparam("pv", "varset", "device_presence_skip_event_dialog=s:(Wildix WP490GR[3|4])")It can be changed in the config file /etc/kamailio/cfg.d/host_specific_custom.cfg.
3. Avoid an issue in which WP / WP 5 phones of Call group members continue to ring after the call was answered
Such an issue may occur in case of large amount of call group members with "Colleague" BLF buttons configured.
Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter:
skip_cg_members_early_state_presence=yesAvailable values: no - the feature is disabled; yes - the feature is enabled. When the parameter is enabled, sending PUBLISH messages is skipped.
2. Reload PBX engine:
callweaver -rx "sip reload"Supported devices:
WorkForce / WelcomeConsole / WP480G r3/ WP490G r3/ Start / WorkForce 5 / ForcePro 5 / WelcomeConsole 5 / Start 5 / SuperVision 5
BRI / PRI Media gateways (EOL)
FXS Media gateways
W-PA/ W-PA 2 (starting from 6.10.20250428.1)
The feature is enabled by default. To disable it:
Edit the file /etc/kamailio/cfg.d/host_specific_custom.cfg by adding the following parameters:
modparam("pv", "varset", "device_caps_sdes_srtp=s:(Wildix WP4[8|9]0GR[3|4])|(Wildix .*BRI)|(Wildix Wildix W0[1-2]PRI)|(Wildix W0[2|4]FXSR3)|(Wildix Wildix 3000)|(Wildix WP410R2)") modparam("pv", "varset", "sdes_srtp=i:0")Available values: "sdes_srtp=i:0" - to disable and "sdes_srtp=i:1" - to enable.
Run the command:
/etc/init.d/kamailio restart(only for BRI/ PRI mgw) Add a custom provisioning parameter SRTPForceUse to [wildixgw] section of the file /rw2/etc/provision.conf
Send the new configuration to devices via Configure / Sync device option in WMS -> Devices
During ongoing calls, a lock on a phone's screen indicates that Direct SDES-SRTP is established.
There is also another option to disable SDES-SRTP on WP 5 phones. To disable it, follow the steps below:
By default, the feature is enabled
Edit the file /rw2/etc/provision.conf by adding the following parameters:
[wildixp5]
SRTPForceUse=noEdit the file /etc/kamailio/cfg.d/host_specific_custom.cfg by adding the following parameters:
modparam("pv", "varset", "device_caps_sdes_srtp=s:(Wildix-WXQA) | (Wildix WP4[8|9]0GR[3|4])") Restart kamailio service by running the command:
systemctl restart kamailio Send the new configuration to devices via Configure / Sync device option in WMS -> Devices
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:
101,9c7533014b00 102,9c7533014b00 103,9c7533014b00 104,9c7533014b01
Modify geolocation sharing on PBX
Specifically this option can be used to block geolocation on Vision (EOL)/ SuperVision (EOL) phones since the ACL "Can/cannot - View geolocation via Legacy Collaboration - Group" can't be applied.
To block geolocation sharing:
Edit the file /rw2/etc/ejabberd/ejabberd_mod_wildix_presence.yml by changing the following parameter to false:
allow_location: falseAvailable values: true - location is allowed, false - location is blocked.
Restart the server:
/etc/init.d/ejabberd restart
Notes:
Users in Legacy Collaboration can still view their own statuses
If geolocation is blocked only on PBX A and not blocked on PBX B in WMS Network, users from PBX A are still able to view geolocation of PBX B users
Enable OPUS transcoding on PBX
Note:
The feature is available starting from WMS 7.05.20251016.1
By default, the option is disabled.
This feature allows you to improve call audio quality by enabling OPUS transcoding on the PBX in case of network issues or when G711/G729 codecs are used.
To enable it, add the following parameter to the /etc/kamailio/cfg.d/host_specific_custom.cfg file:
modparam("pv", "varset", "opus_transcoding=i:1")Run the following command to apply the changes:
callweaver -rx 'sip reload'Security
Enable SSL connection to remote MySQL server
It is possible to configure SSL connection for sending CDR data to external MySQL server.
Upload the following three files to the PBX:
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
Place the files into the directory /etc/callweaver/certs (the directory should be created beforehand) or any other directory on 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.pemRun 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: trueEnable location-based MFA (WMS 6.03)
The support starts from WMS 6.03.
The location-based multi-factor authentication is the feature that requests user to confirm IP address via email when logging in to Legacy Collaboration or using basic authorization from a new IP address. On WMS 6.04.20231020.2 or higher, it is possible to manage location-based MFA via WMS -> PBX -> Features (documentation: WMS Settings Menu - Admin Guide).
On WMS 6.03, you can enable location-based MFA by setting the following parameter in /rw2/etc/env.custom.ini file:
MFA_SERVICE_ENABLE=true Note: On WMS 7, location-based MFA is enabled by default.
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:
[radio] mode=custom application=/usr/sbin/cw_play_http_stream.sh http://stream.104.6rtl.com/rtl-live/mp3-192where 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
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.
x-caracal
Change time range limit for reports
Depending on the PBX load, some limits of allowed days range apply when generating x-caracal reports. Learn more in x-caracal documentation.