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The List of custom config parameters that allow changing various settings in config files. Created: August 2018 Updated: September 2021 Permalink: https://confluence.wildix.com/x/0AiIAQ |
Table of Contents |
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To make any changes to config files, access them via SSH as root:
Code Block |
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nano-tiny /etc/<sub-directory>/<config file name> |
Users
2 PBXs in WMS Network, each with its own Active Directory for users
Note |
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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.
Note |
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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 |
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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
Code Block 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
Warning |
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Important:
|
Note |
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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:
Code Block g729_transcoding=yes
Available values: no – the feature is disabled; yes – the feature is enabled.
Modify HD codecs on PBX
Note |
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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:
Code Block |
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disallow=all
allow=alaw:20,ulaw:20,g729:20,vp8,h264,h263,h263p |
Run the command:
Code Block |
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callweaver -rx "sip reload" |
Modify presence status monitoring via BLF keys
Note |
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Detailed information about the feature: Presence status monitoring. |
...
Edit the file /etc/callweaver/sip-general-custom.conf by adding the following parameter:
Code Block full_presence=no
Available values: no – the feature is disabled; yes – the feature is enabled.
Run the command:
Code Block |
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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:
Code Block |
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send_cg_member_ring_notify=yes |
...
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:
Html |
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modparam("pv", "varset", "skip_cg_members_presence_on_wp=i:1") |
...
Restart kamailio service:
Code Block |
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/etc/init.d/kamailio restart |
Edit the file /etc/callweaver/sip-general-custom.conf by adding the parameter:
Code Block |
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skip_cg_members_presence_on_wp=yes |
...
Restart callweaver service:
Code Block |
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callweaver -rx "sip reload" |
If the feature is enabled, the default devices mask is:
Code Block |
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modparam("pv", "varset", "device_presence_skip_event_dialog=s:(Wildix WP490GR[3|4])") |
...
Note |
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Supported devices:
|
...
Call Groups
Check registration status of Call group members during call distribution
Note |
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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:
Code Block send_regevent_devstate = no
Available values: no – the feature is disabled; yes – the feature is enabled.
Reload SIP by running the command:
Code Block 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 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.
Note |
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|
Collaboration
Disable file transfer in Collaboration chat messages
Note |
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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:
Code Block |
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COLLABORATION_FILE_TRANSFER_ALLOW=false |
Enable Call Control mode for the second opened Collaboration tab
Note |
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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.
To activate this feature, add the following line to the file/rw2/etc/env.custom.ini:
Code Block |
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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.
Note |
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Use cases
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Devices
Modify devices sync
Warning |
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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
Code Block 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
Warning |
---|
Important:
|
Note |
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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:
Code Block g729_transcoding=yes
Available values: no – the feature is disabled; yes – the feature is enabled.
Modify HD codecs on PBX
Note |
---|
Supported devices:
The feature also works for PBXs in WMS Network. |
The feature is enabled by default. To disable it:
Edit the file /etc/kamailio/cfg.d/host_specific_custom.cfgcallweaver/sip-general-custom.conf by adding the following parameters:
Available values: "sdes_srtp=i:0" - to disable and "sdes_srtp=i:1" - to enableCode Block 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")
disallow=all allow=alaw:20,ulaw:20,g729:20,vp8,h264,h263,h263p
Run the command:
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. To disable it:
Edit the file /etc/callweaver/sip-general-custom.conf by adding the following parameter:
Code Block full_presence=no
Available values: no – the feature is disabled; yes – the feature is enabled.
Run the command:
Code Block /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
Note |
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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:
Code Block |
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101,9c7533014b00
102,9c7533014b00
103,9c7533014b00
104,9c7533014b01 |
Check registration status of Call group members during call distribution
Note |
---|
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
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:
Code Block 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:
Html 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:
Code Block /etc/init.d/kamailio restart
Edit the file /etc/callweaver/sip-general-custom.
conf conf by adding the parameter:
Available Code Block skip_cg_members_presence_on_wp=yes
Available values:no
– - the feature is disabled; yes
– - the feature is enabled.
send_regevent_devstate = no
Reload SIP by running the commandRestart callweaver service:
Code Block callweaver -rx' "sip reload'
Enable Direct RTP between Kite and Web phone (WMS 5.0X)
Note |
---|
Full ICE support for Kite and WebRTC phone:
|
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")
Available values: "ice_drtp=i:0" - to disable and "ice_drtp=i:1" - to enable.
Modify geolocation sharing on PBX (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:
Code Block allow_location: false
Available values: true - location is allowed, false - location is blocked.
Restart the server:
Code Block |
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/etc/init.d/ejabberd restart |
Note |
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Notes:
|
Use online streaming of mp3 http/ https sources as music on hold (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:
Code Block [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:
Code Block |
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rcallweaver -rx "moh reload" |
...
Add Dialplan application Set -> Music on hold -> radio
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:
Code Block register => 144?144:123456:”144″@10.168.0.144:5060~~0.6
Where 0.6 is q-value.
Run the command:
Code Block |
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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
Code Block 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:
Code Block session_expire_header_in_update=yes
Available values: no – the feature is disabled; yes – the feature is enabled.
"
Run the command:
Code Block |
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callweaver -x 'sip reload' |
If the feature is enabled, the default devices mask is:
Code Block |
---|
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
Anchor | ||||
---|---|---|---|---|
|
Note |
---|
Supported devices:
|
Note |
---|
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:
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)|(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:
Code Block /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
Note |
---|
During ongoing calls, a lock on a phone's screen indicates that Direct SDES-SRTP is established. |
Select a specific GSM gateway Anchor selectgsmgateway selectgsmgateway
selectgsmgateway | |
selectgsmgateway |
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:
Code Block 101,9c7533014b00 102,9c7533014b00 103,9c7533014b00 104,9c7533014b01
Modify geolocation sharing on PBX
Note |
---|
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:
Code Block allow_location: false
Available values: true - location is allowed, false - location is blocked.
Restart the server:
Code Block /etc/init.d/ejabberd restart
Note |
---|
Notes:
|
Dialplan
Exit code 0 from voicemail
...
- 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
...
Use online streaming of mp3 http/ https sources as music on hold
Note |
---|
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
...
:
Code Block [
...
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.
Note |
---|
|
Collaboration (WMS 5.03)
Disable file transfer in Collaboration chat messages
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:
Code Block |
---|
COLLABORATION_FILE_TRANSFER_ALLOW=false |
Enable Call Control mode for the second opened Collaboration tab
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.
...
Code Block |
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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.
Note |
---|
Note: You can have only 1 active session in Call Control mode. If you open a second tab in Call Control mode, the previous one is terminated. |
Kite
Deliver Kite chat requests to all Call group members
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:
Code Block |
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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.
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:
...
- Full - allows all synchronizations
- Forbid - denies any data synchronizations
- Lite - denies synchronization requests to FIAS/ XOpen interfaces
...
- std - accepts DND events only from FIAS interface
- extended - allows handling DND events also from PBX and XOpen interfaces
...
Supported values: 2 min - 6 min - 10 min (min - default - max)
Note |
---|
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 |
---|
Note: Information about hotel integration: Hotel Integration - FIAS protocol - Guide. |
Integrate custom HTML code to Collaboration by using Google Tag Manager (WMS 5.02)
Warning |
---|
The feature is available only for Cloud PBXs! |
Google Tag Manager allows integrating custom HTML code into Collaboration. 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:
Code Block 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:
Code Block |
---|
<!-- 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:
Code Block |
---|
<!-- 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) --> |
Fax Server
Adjust the resolution of outgoing faxes
Note |
---|
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:
Other widespread resolutions: 204x98, 204x391, 408x391Code Block XResolution=204 YResolution=391
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:
Code Block rcallweaver -rx "moh reload"
Add Dialplan application Set -> Music on hold -> radio
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: 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 |
---|
Note: Information about hotel integration: Hotel Integration - FIAS protocol - Guide. |
Integrate custom HTML code to Collaboration by using Google Tag Manager
Warning |
---|
The support starts from WMS 5.02. The feature is available only for Cloud PBXs! |
Google Tag Manager allows integrating custom HTML code into Collaboration. 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:
Code Block 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:
Code Block <!-- 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:
Code Block <!-- 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) -->
Fax Server
Adjust the resolution of outgoing faxes
Note |
---|
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:
Code Block XResolution=204 YResolution=391
Other widespread resolutions: 204x98, 204x391, 408x391.
Kite
Enable Direct RTP between Kite and Web phone
Note |
---|
The support starts from WMS 5.01. |
Note |
---|
Full ICE support for Kite and WebRTC phone:
|
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")
Available values: "ice_drtp=i:0" - to disable and "ice_drtp=i:1" - to enable.
Deliver Kite chat requests to all Call group members
Note |
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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:
Code Block |
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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:
Code Block 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 default, RTP packets are sent), proceed with the following:
Add the following parameter to the file etc/callweaver/sip-general-custom.conf
Code Block 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:
Code Block session_expire_header_in_update=yes
Available values: no – the feature is disabled; yes – the feature is enabled.
Run the command:
Code Block callweaver -x 'sip reload'
Users
2 PBXs in WMS Network, each with its own Active Directory for users
Note |
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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.
Note |
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Limitation: the option "Remove existing users which are not received from the backend" does not work in this configuration; do not enable it. |