The List of custom config parameters that allow changing various settings in config files.
Created: August 2018
Updated: April 2021
Permalink: https://confluence.wildix.com/x/0AiIAQ
To make any changes to config files, access them via SSH as root:
nano-tiny /etc/<sub-directory>/<config file name>
Users
2 PBXs in WMS Network, each with its own Active Directory for users
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.
Limitation: the option "Remove existing users which are not received from the backend" does not work in this configuration; do not enable it.
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=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
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=yes
Available values: no – the feature is disabled; yes – the feature is enabled.
Modify HD codecs on PBX
Supported devices:
- Collaboration
- Android / iOS apps
- WorkForce / WelcomeConsole / WP480 r3 / WP490 r3
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,h263p
Run the command:
callweaver -rx "sip reload"
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=no
Available 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=yes
Available values: no - the feature is disabled; yes - the feature is enabled.
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. (available in WMS 5.02) 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:
Available values: 0 - the feature is disabled; 1- the feature is enabled.
Restart kamailio service:
/etc/init.d/kamailio restart
Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter:
skip_cg_members_presence_on_wp=yes
Available 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.
Modify direct SDES-SRTP
Supported devices:
- WorkForce / WelcomeConsole / WP480G r3/ WP490G r3/ START (ex WP410)
- BPI / PRI Media gateways
- FXS Media gateways
Note: The feature is disabled by default in WMS 4.
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.
Select a specific GSM gateway
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
Check registration status of Call group members during call distribution
Note: The feature is disabled by default in WMS 4.
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 = no
Available values: no – the feature is disabled; yes – the feature is enabled.
Reload SIP by running the command:
callweaver -rx'sip reload'
Enable Direct RTP between Kite and Web phone (WMS 5.0X)
Full ICE support for Kite and WebRTC phone:
- endpoints in the same network - media goes directly
- endpoints in different networks and open/ moderate NAT - STUN is used to find the best pair of candidates
- endpoints in different networks, strict NAT - media goes through TURN (on PBX)
The feature is disabled by default. To enable it:
Add the following line to the file /rw2/etc/kamailio/host_specific_custom.cfg:
modparam("pv", "varset", "ice_drtp=i:1")
Available values: "ice_drtp=i:0" - to disable and "ice_drtp=i:1" - to enable.
Modify geolocation sharing on PBX (WMS 5.02)
Specifically this option can be used to block geolocation on Vision/ SuperVision phones since the ACL "Can/cannot - View geolocation via 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: false
Available values: true - location is allowed, false - location is blocked.
Restart the server:
/etc/init.d/ejabberd restart
Notes:
- Users in 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
Use online streaming of mp3 http/ https sources as music on hold (WMS 5.02)
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/mp3-192
where http://stream.104.6rtl.com/rtl-live/mp3-192 is your stream source.
Run the command:
rcallweaver -rx "moh reload"
Add Dialplan application Set -> Music on hold -> radio
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:
register => 144?144:123456:”144″@10.168.0.144:5060~~0.6
Where 0.6 is q-value.
Run the command:
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
rtpkeepalive_mode=udp
Available values: udp | rtp.
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:
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'
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
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)
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
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
- (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: the parameter "BadgeTimeout" is removed from to whoteld_manager_wms.conf.
The new default timeout is 60 seconds. New supported values: 20-600 seconds (min-max).
Note: Information about hotel integration: Hotel Integration - FIAS protocol - Guide.
Integrate custom HTML code to Collaboration by using Google Tag Manager (WMS 5.02)
The feature is available only for Cloud PBXs!
Google Tag Manager allows integrating custom HTML code into Collaboration. 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:
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) -->
Fax Server
Adjust the resolution of outgoing faxes
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:
XResolution=204 YResolution=391
Other widespread resolutions: 204x98, 204x391, 408x391.