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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


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:

  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.

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 direct SDES-SRTP  

Supported devices:

  • WorkForce / WelcomeConsole / WP480G r3/ WP490G r3
  • BPI / PRI 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)")
    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 green 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 5.0)

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")

    Available values: "ice_drtp=i:0" - to disable and "ice_drtp=i:1" - to enable.

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.

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 filqueues.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. 

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.

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.


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