The List of custom config parameters that allow changing various settings in config files.
Created: August 2018
Updated: July 2020
Permalink: https://confluence.wildix.com/x/0AiIAQ
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 parameters disable_sync_portal=1 and disable_sync_manual=1 to the config file /rw2/etc/pbx/device_sync.conf
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:
- Add the parameter g729_transcoding=yes
Modify HD codecs on PBX
Notes:
- Currently supported devices: Collaboration / Android v. 4.01.12 / iOS v. 7.1.35284 / WorkForce / WelcomeConsole v. 68.145.2.1 / WP480 r3 v. 63.145.8.111 / WP490 r3 v. 67.145.8.75
- The feature works for PBXs in WMS Network
The feature is enabled by default starting from WMS version 4.01.44329.36. To disable it:
- Edit the file /etc/callweaver/sip-general-custom.conf by adding the following lines:
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 starting from WMS version 4.01.44251.31. To disable it:
- Edit the file /etc/callweaver/sip-general-custom.conf by adding the following lines:
full_presence=no
Run the command:
callweaver -rx "sip reload"
Modify direct SDES-SRTP
Supported devices:
- BPI/ PRI Media gateways
- WP480G r3/ WP490G r3/ WorkForce, WelcomeConsole
The feature in:
- WMS 5.0 - enabled by default
- WMS 4.0 - disabled by default
To modify:
Edit the file /etc/kamailio/cfg.d/host_specific_custom.cfg by adding the following lines:
Available values:- sdes_srtp=i:0" - to disable
- sdes_srtp=i:1" - to enable
modparam("pv", "varset", "device_caps_sdes_srtp=s:(Wildix WP4[8|9]0GR[3|4])|(Wildix .*BRI)|(Wildix Wildix W0[1-2]PRI)") modparam("pv", "varset", "sdes_srtp=i:0")
*restart SIP proxy via console (/etc/init.d/kamailio restart)
(only for BRI/ PRI mgw) Add a custom provisioning parameter SRTPForceUse=no (to disable) / SRTPForceUse=yes (to enable) 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 red lock on a phone's screen indicates that Direct SDES-SRTP is established.
Select a specific GSM gateway
The option allows selecting a specific GSM gateway for SMS sending for each separate user (available starting from WMS v. 4.03.44914.04):
Access smsd-route.conf file via SSH as root:
nano-tiny /etc/wildix/smsd-route.conf
Specify 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
The feature in:
- WMS 5.0X - enabled by default
- WMS 4.0X - disabled by default
When the feature is enabled, unavailable users (means no registered devices or no push for mobile apps), do not receive calls from a queue. The logic is applied only for Call group calls.
To enable it:
Edit the file /etc/callweaver/sip-general-custom.conf by adding the line:
send_regevent_devstate = yes
To disable, add the line: send_regevent_devstate = no
Reload SIP by running the command:
callweaver -rx'sip reload'
Enable Direct RTP between Kite and Web phone (WMS 5.0)
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")
Trunks
Enable Q-value (serial forking) for trunk registration
To enable q-value (serial forking) parameter via custom register string:
Copy registration line for some 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 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 parameter rtpkeepalive_mode=udp (available values [udp|rtp]) to the file etc/callweaver/sip-general-custom.conf
Dialplan
Exit code 0 from voicemail
Starting from WMS Stable Version 3.88.41762.32, 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.
Access queues-general-custom.conf via SSH as root:
nano-tiny /rw2/etc/callweaver/queues-general-custom.conf
- Add a custom parameter, for example, autofill = yes (by default, the file queues.conf contains autofill = no parameter)
- Save changes
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: Starting from WMS Beta Version 3.88.42955.49, the parameter "BadgeTimeout" is removed from whoteld_manager_custom.conf to whoteld_manager_custom.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.
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