Note: The article in English is the reference for the other languages.
- WebRTC gateway 2.3.7 is available on ALE Business Portal since October 4, 2023.
Once the WebRTC Gateway, PBX Sip trunk and Virtual resources (REX on OmniPCX Enterprise, AnyDevice on OXO Connect and OXO Connect Evolution) are configured and your main phone is well associated to your Rainbow account you should have in your Rainbow Web/Desktop client following new menu:
Menu to choose your device to make phone calls (the devices list depends of what you have defined into your profile):
1. Configuration check
from version 1.66 the SSH access is deactivated. If you need to activate/deactivate it you have to run following commands (from version 1.67): mpssh ON|OFF
1.1 Check WebRTC gateway connectivity to Rainbow cloud.
The command mpshow will display the current configuration used by the WebRTC gateway
The command mpcheck will execute some connection tests to Rainbow cloud. The Network configuration and the access to NTP, DNS, Proxy, PBX, TURN server are checked.
The command mpcollect --log will save in addition to the mpcheck results some information about the WebRTC gateway (HW, CPU/RAM usage, installed linux packages, running services, etc...) and the WebRTC gateway logs. The resulting output file is located by default in $HOME/mpcollect.tgz
Please provide this mpcollect.tgz file when you open a support request.
1.2 Config files and Log files:
Once you have executed both scripts mpnetwork and mpconfig you can check the configuration in following file /etc/rainbow-mp.cfg.
This can also be checked with the command mpshow
- The scripts configure the right values into following configuration files:
- Two logs files are generated, one for otlitemediapillargateway and one for janus-gateway-mediapillar:
2. Check WebRTC gateway status and connectivity
2.1 The WebRTC Gateway runs 3 different services for the connection to Rainbow and to the PBX.
2.1.1 - Check their status
you can run following command
sudo service otlitemediapillargateway status
sudo service janus-gateway-mediapillar status
sudo service kamailio status
The "disabled" status from the answer of the status of otlitemediapillargateway is an indication for the autostart of this service and not the status of the service itself
Loaded: loaded (/lib/systemd/system/janus-gateway-mediapillar.service; disabled; vendor preset: enabled)
2.1.2 - Restart the services
sudo service otlitemediapillargateway restart
sudo service janus-gateway-mediapillar restart
sudo service kamailio restart
2.1.3 - Restart the server
2.2 Check that the Rainbow phone numbers of the user are registered into the WebRTC gateway
a) you can run following command:
sudo /usr/sbin/kamctl ul show
b) In case you don't see your device restart the Rainbow agent:
- OmniPCX Enterprise with following command:
dhs3_init –R RAINBOWAGENT
- OXO Connect & OXO Connect Evolution
Rainbow activated button in OMC Rainbow menu
c) and last step is then to restart the Kamailio service from WebRTC Gateway
sudo service kamailio restart
2.3 SIP traces with the mpndump command
Since version 2.3.7 it is recommended to use the command mpndump (on | off | clean) to make sip traces.
When mpndump is run, the command checks whether there is enough disk space to generate a maximum of 8 files of 32 MB each, with a file rotation mechanism. For the disk space check, an additional margin of 512 MB is taken into account to reserve space for a remote upgrade.
In conclusion, a minimum of 768 MB of free disk space is required to start capturing with mpndump.
Files will be saved to /home/rainbow/ with following format ndump-2023-10-06T06-57:04.pcap-0 up to pcap-7.
When the mpndump command starts old files are first removed.
Note that you can close the ssh console, the tcpdump trace capture will not be stopped. When reconnecting you'll see that the tcpdump is still running, To stop the trace capture you'll have to run the command mpndump off.
Once you have collected the files you can remove them with the command mpndump clean.
3. How to report an issue
The proper working of the VoIP Calling functionality depends on 3 elements:
- PBX configuration:
- for the user, configuration of a virtual resource (REX for OmniPCX Enterprise or AnyDevice for OXO Connect & OXO Connect Evolution).
- SIP trunk configuration.
- WebRTC gateway activation for PBX.
- granting a Business or Enterprise license to the user.
- linking the Rainbow Account with the PBX phone extension.
If you have an issue to configure your PBX (network issue to connect to the Rainbow cloud, SIP trunk issue or user configuration) please open a ticket at ALE.WelcomeCenter@al-enterprise.com
If your problem concerns VoIP calling, WebRTC gateway or Rainbow configuration, please contact your support team. Please see the article How Can I Get Support in Rainbow? to find out who you should contact.
When you open a ticket please provide following information:
- Provide the result of the command mpcollect described in §1.1.
- Provide the Rainbow client logs.
- Indicate if the issue is systematic, random or occurred only once and if it can be reproduced.
- Indicate if the issue is for all users or only for one or a few.
- Indicate if a restart of the PBX rainbow agent, WebRTC gateway or Rainbow client solved the issue.
- Indicate the name and email of the user having the issue and the exact date and time the issue occurred.