Webrtc Phone legacy sip not working
Hi, the Webrtc phone does not work properly with legacy sip, when I created an extent as webrtc using legacy sip the extent can not be registered in asterisk using the webrtc phone because the plugin is trying to register as pjsip. with pjsip work great but in my scenario, I must use legacy SIP
WebSocket connection from 'xx.xx.xx.xx:34672' for protocol 'sip' accepted using version '13'
[2022-02-21 23:20:03] NOTICE[18319]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'REGISTER' from '"204" <sip:[email protected]>' failed for 'xx.xx.xx.xx:34672' (callid: aabfeeg7m6b1slvgb3c1o0) - No matching endpoint found
[2022-02-21 23:20:03] NOTICE[18319]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'REGISTER' from '"204" <sip:[email protected]>' failed for 'xx.xx.xx.xx:34672' (callid: aabfeeg7m6b1slvgb3c1o0) - No matching endpoint found
[2022-02-21 23:20:03] NOTICE[18319]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'REGISTER' from '"204" <sip:[email protected]>' failed for 'xx.xx.xx.xx:34672' (callid: aabfeeg7m6b1slvgb3c1o0) - Failed to authenticate
Any help on how to resolve it will be appreciated.
Comments
Webrtc phones knows nothing about channel drivers. Your problem is in your Asterisk configuration.
There is a setting in sip.conf to enable or disable websocket support. If you want sip websocket to go to chan_sip you will have to set that to true:
websocket_enabled = trueBest regards,
Thank you for your help that information put me in the right direction, I found the solution.
This is my working environment
Issabel4 PBX
centos 7
As described in my first post when I created a webrtc extent as chan_sip, and try to use that extent in with the webrtc phone, the PBX does not register the extent, even after I enable websocket_enabled option in the sip advance setting, we need also to enable these options
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
those options are not present on the GUI we can add to sip_general_custom.conf
again thank for your Nicolas
Glad it worked.
Latest issabel version comes with all available webrtc settings for modification (but it is still marked as beta)
Best regards,