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Info

The List of custom config parameters that allow changing various settings in config files.

Created: August 2018

Updated: February 2020July 2020

Permalink: https://confluence.wildix.com/x/0AiIAQ

Table of Contents

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:

  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.

Note

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

Warning

Important: By default, the sync is enabled. It is necessary to disable it in case of FAILOVER scenario on Failover PBXsPBXs!

To disable the sync:

  • Add parameters disable_sync_portal=1  and disable_sync_manual=1 to the config file file /rw2/etc/pbx/device_sync.conf
    Available values: 1 – sync is disabled; 0 – sync is enabled

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Modify g729 transcoding for web phone calls to trunks which do not support

...

g711

Warning

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

Note: Feature is supported only on PBXs with modern CPU or Cloud.


To enable g729 transcoding, edit the file edit the file /etc/callweaver/sip-general-custom.conf:

  • Add the parameter g729_transcoding=yes

...

Modify HD codecs on PBX 

Note

Notes:

  • Currently supported devices: Collaboration , / Android v. 4.01.12 , / iOS v. 7.1.35284 apps, WorkForce , / WelcomeConsole v. 68.145.2.1 , WP480 r3 v. 63 63.145.8.111 , WP490 r3 v. 67.145.8.75 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: 
Code Block
disallow=all
allow=alaw:20,ulaw:20,g729:20,vp8,h264,h263,h263p

...

Code Block
callweaver -rx "sip reload" 

...

Modify presence status monitoring via BLF keys

Note

Detailed information about the feature: Presence status monitoring.


The feature is enabled by default starting from WMS version 4version 4.01.44251.31. To disable it:

  • Edit the file /etc/callweaver/sip-general-custom.conf by adding the following lines: 

...

Code Block
callweaver -rx "sip reload" 

...

Modify direct SDES-SRTP  
Anchor
sdessrtp
sdessrtp

Available starting from WMS v. 4.03.44914.04.
Note

Supported devices:

  • BPI/ PRI Media gateways
  • WP480G r3/ WP490G r3/ WorkForce, WelcomeConsole
Note

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:1" - to enable
    • sdes_srtp=i:0" - to disable
Code Block
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:10")

*restart SIP proxy via console (/etc/init.d/kamailio restart)

  • (only for BRI/ PRI mgw) Add a custom provisioning parameter SRTPForceUse=yes to [wildixgw] section of the file /rw2/etc/provision.conf

  • Send the new configuration to devices via Configure / Sync device option in WMS -> Devices


Note

During ongoing calls, a red lock on a phone's screen indicates that Direct SDES-SRTP is established.


Select a specific GSM gateway 
Anchor
selectgsmgateway
selectgsmgateway

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: 

    Code Block
    nano-tiny /etc/wildix/smsd-route.conf
  • Specify user extension and MAC address of GSM gateway, for example:

    Code Block
    101,9c7533014b00
    102,9c7533014b00
    103,9c7533014b00
    104,9c7533014b01

Check registration status of Call group members during call distribution

Note

Available starting from WMS v. 4 4.03.45071.08.


The feature is disabled by default. When 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 file etc/callweaver/sip-general-custom.conf by adding the line: 

    Code Block
    send_regevent_devstate = yes

    To disable, add  add the line: send_regevent_devstate = no


  • Reload SIP by running the command: 

    Code Block
    callweaver -rx'sip reload'

Enable Direct RTP between Kite and Web phone (WMS 5.0)

Note

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

Code Block
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  register => 144?144:123456:”144″@10.168.0.144:5060~~0.6 (where 0.6 is q-value)
  • Run the command: 
Code Block
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 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  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 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 in file queues.conf.

  • Access queues-general-custom.conf via SSH as root: 

    Code Block
    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 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 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. 

Note

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

      Note: Starting from WMS Beta Version 3.88.42955.49, the parameter "BadgeTimeout" is removed from whoteldwhoteld_manager_custom.conf to whoteld_manager_custom.conf.

      The new default timeout is 60 seconds. New supported values: 20-600 seconds (min-max).

Note

Note: Information about hotel integration: Hotel Integration - FIAS protocol - Guide.

Fax Server

Adjust the resolution of outgoing faxes 

Note

Note: The  The default resolution is 204x196 (fine).


To adjust the resolution:

  • Edit the config file file /rw2/etc/faxglobal.conf by  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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