Dialing livekit sip endpoint is not working

Asterisk Asterisk SIP
1

If i use older sip module and dial Dial(SIP/+919212345001@*.sip.livekit.cloud) then everything is working, call is being handled by livekit endpoint correctly.

But if I disable the sip module and only use pjsip and dial like: Dial(PJSIP/+919212345001@*.sip.livekit.cloud)

Then I am getting:

Executing [00919228826135@00919228826135:1] Verbose("PJSIP/*-endpoint-0000000d", "2, "Incoming call received"") in new stack
  ==  "Incoming call received"
    -- Executing [*@*:2] Dial("PJSIP/*-endpoint-0000000d", "PJSIP/+919212345001@*.sip.livekit.cloud") in new stack
[Mar  1 12:26:03] ERROR[344773]: chan_pjsip.c:2687 request: Unable to create PJSIP channel - endpoint '*.sip.livekit.cloud' was not found
[Mar  1 12:26:03] NOTICE[344838][C-0000000e]: app_dial.c:2766 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/*-endpoint-0000000d' status is 'CHANUNAVAIL'
2

You need to specify an endpoint, in order to define the codecs and transports to use. See the third example in Dialing PJSIP Channels - Asterisk Documentation

I assume “*” is a redaction.

1 Like
3

Thanks for the help. Working now :+1:

4

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