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Jan 24, 2022 17:32

Custom config parameters List

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

Created: August 2018

Updated: January 2022

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>

Call Groups

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'

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. 

Collaboration 

Disable file transfer in Collaboration chat messages 

The support starts from WMS 5.03.


This feature blocks the possibility of transferring files in 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=false

Disable Chat and Post-It functionality in Collaboration 

The support starts from WMS 5.03.


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=false

Use case: This feature is useful for teams4Wildix integration, as long as Microsoft Teams offers its own chat. Disabling chat functionality in Collaboration helps users to avoid confusion between these two systems and use one chat instead of two. 

Enable Call Control mode for the second opened Collaboration tab 

The support starts from WMS 5.03.


By default, only 1 active Collaboration session is permitted. However, there is an option to allow the second Collaboration tab – in Call Control mode. This can be used, for example, for media devices when connecting via remote desktop. The option is particularly practical for teams4Wildix integration.

To activate this feature, add the following line to the file /rw2/etc/env.custom.ini

CALL_CONTROL_FEATURE=true

Once the feature is activated, an additional option appears on the Collaboration login window – Call Control only. Tick this checkbox off to log into Collaboration in Call Control mode.  

Use case: User can set "Call control only" mode to be automatically enabled by adding a parameter in the URI (useful when it is needed to share the link among all employees or bookmark the page) How-to: add ?sip=no to PBX domain name ->https://pbxname.wildixin.com/collaboration/?sip=no

Display Departments in a tree view on the Colleagues tab 

The support starts from WMS Beta 5.04.20211227.3

The feature allows to create a multilevel hierarchy of Departments and display them in a tree view on the Colleagues tab in Collaboration.

To enable this option, add the following line in /rw2/etc/env.custom.ini:

DEPARTMENTS_AS_TREE=true

By default, the value of the parameter is “false” and the Colleagues tab in Collaboration displays the list of Groups. When the parameter is set to “true”, the Colleagues tab displays Departments.

Documentation: How to configure Departments tree

Change the default separator (optional)

The support starts from WMS Beta 5.04.20211227.3

By default, the separator “/” is used to indicate the Department path in WMS -> Users -> Edit -> Departments field. 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

Display all PBX users on the Colleagues tab 

The support starts from WMS Beta 5.04.20211227.3

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=true

Current limitation: 

The status of users that are not added to the roster is not displayed.

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:

    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 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

Modify geolocation sharing on PBX

The support starts from 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

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 support starts from 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: 

    callweaver -rx "moh reload"
  • Add Dialplan application Set -> Music on hold -> radio

Note: Starting from WMS Beta 5.04.20220117, 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.

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
  • PostponeRE - in case of using ASA hotel software, this option must be set to "true" to postpone processing of RE (Room equipment) request
  • (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 to /etc/wildix/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 by using Google Tag Manager

The feature is available only for Cloud PBXs!

Supported services:

  • Collaboration (starting from WMS 5.02)
  • x-caracal (starting from WMS 5.03)

Google Tag Manager allows integrating custom HTML code into Collaboration and x-caracal. To integrate the code, 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: 

  1. 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 -->
  2. 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) -->

Feelingstream

Wildix supports integration with the Feelingstream platform for voice recognition and call recordings in stereo format. Feelingstream allows to conduct STT analysis of stereo recordings received from Wildix PBX.

Requirements:

  • Activated Feelingstream account
  • Quality of call recordings set to WAV (you can do this in WMS -> Dialplan -> General Settings)
  • For HW/ VM: mounted storage for recordings

To enable the integration, create the config file /etc/feelingstream.conf with the following parameters: 

host=wildix.feelingstream.com
apikey=fcd4206d-aaab-56cb-bdc8-a33dc4611a4b_ab30ac63-1234-1234-1234-123456789012
bucket=call
lang=en

Where:

  • host  – received from Feelingstream side
  • apikey – generated using Feelingstream web interface
  • bucket – created manually using Feelingstream web interface or API
  • lang – optional parameter for setting the language for recognition. If not indicated, the language is set automatically.

To enable stereo mode on the PBX side and improve the quality of recognition, go to WMS -> Dialplan -> General Settings -> Set dialplan variables and add the following Global Dialplan variable: STEREO_RECORDINGS=yes.

Note: Stereo recordings can be activated and used separately on any PBX, without Feelingstream integration, consult the documentation Custom Global Dialplan Variables List.

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.

Disable Error Correction Mode (ECM) 

Note: The support starts from WMS Beta 5.04.20211214.3. 

ECM is enabled by default. To disable it, add the following line in /etc/callweaver/res_fax_custom.conf:

ecm=no

Set maximum/ minimum fax transmission speed

Note: The support starts from WMS Beta 5.04.20211214.3. 

Maximum transmission speed

To change the maximum transmission speed, modify the following parameter in the /etc/callweaver/res_fax_custom.conf file:

maxrate=14400

The default value is 14400

Possible values: 2400 | 4800 | 7200 | 9600 | 12000 | 14400

Minimum transmission speed

To change the minimum transmission speed, modify the following parameter in the /etc/callweaver/res_fax_custom.conf file:

minrate=2400

The default value is 2400

Possible values: 2400 | 4800 | 7200 | 9600 | 12000 | 14400

Kite

Enable Direct RTP between Kite and Web phone 

The support starts from WMS 5.01.

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.

Deliver Kite chat requests to all Call group members 

The support starts from WMS 5.03.


In some cases, for example in a Call group with Round Robin strategy, Kite chat requests may not be equally distributed within the group. 

To deliver chat requests to all Call group members, edit the config file /rw2/etc/ejabberd/ejabberd_mod_wildix_kite.yml by changing the value of the pickup_strategy parameter:

pickup_strategy: sendall

Available values:

  • onebyone (default) - sends Kite chat requests to Call group members one by one
  • sendall - sends chat requests to all CG members simultaneously. 

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'

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.

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