Call is not getting routed to asterisk

Asterisk Asterisk SIP
1

I’ve configured opensips as a mid registrar and asterisk as the main registrar. The registration is happening with asterisk. But when making a call, the dialplan is not getting executed.

console log:
Registered SIP ‘201004’ at 10.176.16.131:63954
== WebSocket connection from ‘10.214.0.18:60375’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘8714691684’ at 10.214.0.18:60375
– Unregistered SIP ‘201004’
– Registered SIP ‘201004’ at 10.214.0.18:60375
– Unregistered SIP ‘8714691684’
– Unregistered SIP ‘8714691684’
– Unregistered SIP ‘8714691684’
– Registered SIP ‘8714691684’ at 10.214.0.18:60375
== WebSocket connection from ‘10.176.16.131:63954’ closed
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
> 0x7effb004c0c0 – Strict RTP learning after remote address set to: 14.139.173.219:12979
– Registered SIP ‘6238900031’ at 10.214.0.18:60375
> 0x7effb0056600 – Strict RTP learning after ICE completion

2

If the call will be going through OpenSIPS then you need to do standard troubleshooting. Do a SIP trace in Asterisk and see if the traffic appears, if it doesn’t then the problem is your OpenSIPS configuration.

3

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