The List of custom config parameters that allow changing various settings in config files. Created: August 2018 Updated: May 2024 Permalink: https://wildix.atlassian.net/wiki/x/Fh3OAQ |
To make any changes to config files, access them via SSH as root:
nano-tiny /etc/<sub-directory>/<config file name> |
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' |
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.
Run the following command:
callweaver -x "reload" |
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.
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.
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.
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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 |
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. |
Starting from WMS 5.04.20220309.1, the Colleagues tab in Collaboration shows only the list of Departments and there is possibility to create a multilevel hierarchy of Departments and display them in a tree view.
When indicating the Department path in WMS -> Users -> Edit User -> Department field, the default separator "/" should be used. 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.
The support starts from WMS 5.04.20220309.1 |
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. |
Starting from WMS 6.04, Collaboration users can enable Noise suppression in Collaboration Settings -> Web Phone. To enable Noise suppression on the PBX level for all users, edit the following parameter in the /rw2/etc/env.custom.ini file:
COLLABORATION_NOISE_SUPPRESSION=true |
Note: The settings are applied to all PBX users except for those, who enabled/ disabled Noise suppression in Collaboration settings. |
Starting from WMS v. 6.02.20230306.1, it is possible to configure that certain numbers dialled via mobile Collaboration app are automatically forwarded to native dialler (call via Mobile network) for all PBX users.
Note: The support of the feature on mobile starts from Android Collaboration app v. 5.11; iOS Collaboration app v. 8.11. |
Add the following parameter to /rw2/etc/env.custom.ini file:
COLLABORATION_MOBILE_DIRECT_CALL=xxx,yyy |
Where xxx, yyy are the extensions/ numbers, calls to which should be automatically forwarded to the native dialler. By default, the field is empty.
Starting from Android app v. 5.13.05, it is possible to predefine on PBX the values for Advanced settings in Collaboration mobile app (Android).
Note: The support starts from WMS Beta 6.04.20230724.1. |
For this, set the following parameters in the /rw2/etc/env.custom.ini file:
Setting on mobile | Parameter | Default value | Possible values |
---|---|---|---|
Disable SIP mode | COLLABORATION_MOBILE_DISABLE_SIP_MODE | false | true | false |
Enable WebRTC | COLLABORATION_MOBILE_ENABLE_WEBRTC | false | true | false |
Firewall bypass (beta) | COLLABORATION_MOBILE_FIREWALL_BYPASS | off | auto | on | off |
Outgoing calls via Wildix | COLLABORATION_MOBILE_OUTGOING_CALLS_VIA_WILDIX | false | true | false |
Run in background | COLLABORATION_MOBILE_RUN_IN_BACKGROUND | false | true | false |
User status when talking on mobile | COLLABORATION_MOBILE_USER_STATUS_WHEN_TALKING_ON_MOBILE | false | true | false |
Native Calls Support | COLLABORATION_MOBILE_NATIVE_CALLS_SUPPORT | false | true | false |
Mic gain | COLLABORATION_MOBILE_MIC_GAIN | 0 | -4..12 |
Activate DND mode | COLLABORATION_MOBILE_ACTIVATE_DND_MODE | false | true | false |
More about advanced settings on mobile: Appendix 3: Advanced Settings
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.
Important:
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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.
Run the command:
callweaver -rx "sip reload" |
Supported devices:
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" |
Starting from WMS 5.04.20220309.1, there is custom provisioning parameter WideNarrowbandUse, which allows to manage wideband codecs usage. By default, the parameter is disabled and wideband codecs are forced for all devices. When the parameter is enabled (WideNarrowbandUse=yes), the previous behaviour remains: all devices use wideband codec priority or narrowband codec priority and remote codecs priority or local codecs.
In case of WideNarrowbandUse=yes, you need to add relevant lines to /etc/callweaver/sip-wideband.conf to set the parameters
To Enable wideband codec usage for all networks:
forcewideband=yes disableautowideband=yes |
To Enable wideband codec usage in LAN:
forcewideband=no disableautowideband=no |
To set Networks where force usage of wideband codecs (i.e. for network 10.100.6.0/16):
forcewideband=no disableautowideband=yes wideband=10.100.6.0/16 |
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" |
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.
Run the command:
callweaver -rx "sip reload" |
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.
3. Avoid an issue in which WP phones of Call group members continue to ring after the call was answered
Such an issue may occur in case of large amount of call group members with "Colleague" BLF buttons configured.
Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter:
skip_cg_members_early_state_presence=yes |
Available values: no - the feature is disabled; yes - the feature is enabled. When the parameter is enabled, sending PUBLISH messages is skipped.
2. Reload PBX engine:
callweaver -rx "sip reload" |
Supported devices:
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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
During ongoing calls, a lock on a phone's screen indicates that Direct SDES-SRTP is established. |
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 |
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:
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Note: The support starts from WMS 6.06.20240425.1. |
It is possible to configure FXS to recognize pulse dialing to avoid an issue when phones using pulse dialing only could not make calls. For this, add the following parameter to the /rw2/etc/provision.conf file, indicating the MAC address of the necessary FXS device(s) and minimum and maximum flash hook time:
UseFlashHookTime={"9C750430060a": {"min": 150,"max": 400},"9C71153200DC": {"min": 300,"max": 500}} |
You can set as many devices as you need.
Note:
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It is possible to configure SSL connection for sending CDR data to external MySQL server.
Upload the following three files to the PBX:
Note: The files should be generated by server CA file on MySQL server. For MySQL v. 5.7, you can refer to the following instruction. |
client-cert.pem
Make changes in the file /etc/callweaver/cdr_mysql.conf by adding/ editing the following rows:
ssl_cert = /etc/callweaver/certs/client-cert.pem ssl_key = /etc/callweaver/certs/client-key.pem ssl_ca = /etc/callweaver/certs/ca.pem |
Run the following command:
callweaver -x'reload cdr_mysql.so' |
In the file /rw2/etc/ejabberd/ejabberd_mod_mam.yml, edit the following line:
sql_ssl: true |
The support starts from WMS 6.03. |
It is possible to enable the location-based multi-factor authentication that requests user to confirm IP address via email when logging in to Collaboration or using basic authorization from a new IP address.
The feature is disabled by default. To activate it, set the following parameter in /rw2/etc/env.custom.ini file:
MFA_SERVICE_ENABLE=true |
Note: Starting from WMS 6.04.20231020.2, it is possible to manage location-based MFA via WMS -> PBX -> Features. Documentation: WMS Settings Menu - Admin Guide |
Support for exit code 0 from voicemail allowing caller to speak with an operator was added.
How to use:
How to enable:
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 5.04.20220309.1, 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. |
Starting from WMS 5.04, depending on the PBX load, some limits of allowed days range apply when generating x-caracal reports. Learn more in x-caracal documentation.
To set custom limit (for example 50 days), add the following parameter to /var/www/x-caracal/.env file:
MAX_ALLOWED_DAY_RANGE=50 |
By default, after 10 minutes of inactivity, x-caracal page needs to be refreshed. Starting from WMS 6.03, it is possible to modify its inactivity timeout.
To modify the timeout, add the following parameter to /var/www/x-caracal/.env file:
INACTIVITY_TIMEOUT_MINUTES = 120 |
The Maximum value is 600 (10 hours). In case you set the inactivity timeout to 0, it gets disabled.
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:
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. |
The feature is available only for Cloud PBXs! Supported services:
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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:
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) --> |
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:
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:
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. |
For the correct user presence sync, it is necessary to enable sending of presence events to Analytics on PBXs with UC licenses (for x-bees licenses, it is enabled by default). For this, follow the steps described below:
Create a directory /etc/systemd/system/pbx-data-engine.service.d
# mkdir /etc/systemd/system/pbx-data-engine.service.d |
Create a file /etc/systemd/system/pbx-data-engine.service.d/override.conf and add the following lines
# vi /etc/systemd/system/pbx-data-engine.service.d/override.conf [Service] ExecStart= ExecStart=/usr/sbin/pbx_data_engine.py --daemon --mode calls presence |
Reload systemd and restart the service to apply the changes
# systemctl daemon-reload # systemctl restart pbx-data-engine |
Run the following command to check what events are sent to Analytics
echo mode | socat - UNIX-CONNECT:/var/run/data_engine/data_engine.sock |
Note: The default resolution is 204x196 (fine). |
To adjust the resolution:
XResolution=204 YResolution=391 |
Other widespread resolutions: 204x98, 204x391, 408x391.
Note: The support starts from WMS 5.04.20220309.1. |
ECM is enabled by default. To disable it, add the following line in /etc/callweaver/res_fax_custom.conf:
ecm=no |
Note: The support starts from WMS 5.04.20220309.1. |
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
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
The support starts from WMS 5.03. |
In some cases, Kite chat requests may not be equally distributed within the group.
To deliver chat requests to all Call group members simultaneously, 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:
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)
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" |
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.
Run the command:
callweaver -rx "sip reload" |
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' |
Note: The support starts from WMS 6.01.20221019.4. |
The option is disabled by default. To enable it and allow calls from users that have the same public IP address as a static trunk, do the following:
Add the below line to /etc/kamailio/cfg.d/host_specific_custom.cfg file:
modparam("pv", "varset", "allow_users_and_trunks_behind_same_nat=i:1") |
Available values: 1 - enabled; 0 - disabled.
Run the command:
systemctl restart kamailio.service |
You need admin access to Active Directory server. |
To make it work, proceed as follows:
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. |
By default, the maximum number of Voicemails per user is 100. Starting from WMS 6.01.20220621.2, it is possible to extend the number. For this, change the value of the following parameter in the /etc/callweaver/voicemail-custom.conf file:
maxmsg=200 |
Where 200 is the desired number of allowed voicemails.
Notes:
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It is possible to change type of reconnection to the database in case the connection was lost. For this, add the following custom parameter to /etc/callweaver/cdr_mysql.conf configuration file:
reconnect_mode=ping |
The “reconnect_mode” parameter has the following values:
Note: The support starts from WMS 6.05.20240119.1. |
On some iOS versions, there is an occasional issue of in which there can be no media after answering a call, in case the phone was locked and it is an API call by an external CRM or a call from Collaboration with Any selected as a device. If you encounter such an issue, to avoid it, you can add the following line to the /rw2/etc/env.custom.ini file:
ORIGINATE_VIA_ANY_DEVICE_TRANSFER=false |
Starting from W-AIR firmware v.07.30.xx.xx and higher, old Wildix repeaters are no longer supported and need additional configuration to stay on previous firmwares to continue working. For old repeaters to work, you need to add the following parameter to the /rw2/etc/provision.conf file:
SupportObsoleteRepeater = yes |
The default value is “no”. When the parameter is enabled, the base stations automatically switch to the previous firmware on which the obsolete repeaters are supported (v. 06.xx.xx.xx)
Note: To differentiate between obsolete and supported repeaters, check out the document How to Identify Obsolete Wildix Repeaters |