hello ,
am facing an issue with the outbound calls , when ever the agent start outbound calls the other party phone will start ringing after 20-30 second ! .
please let me know how can i troubleshot such cases . what can cause such case ?
Thanks in advance .
Setup :
PBX -------> digium400 gateway -------> ISP
My first stop would be the asterisk console; asterisk -r from the command line, then watch what the server does when you place the call.
I’m having the same issue. Did you ever get a resolution?
Here is what I’m seeing in the CLI:
[Oct 28 15:59:17] – Executing [911231231234@unrestrictedInternational:3] Dial(“SIP/3759-00000000”, “SIP/11231231234@207.227.123.123,300”) in new stack
[Oct 28 15:59:17] == Using SIP RTP CoS mark 5
[Oct 28 15:59:37] – Called SIP/11231231234@207.227.123.123
[Oct 28 15:59:38] – SIP/207.227.123.123-00000001 answered SIP/3759-00000000
[Oct 28 15:59:38] – Channel SIP/207.227.123.123-00000001 joined ‘simple_bridge’ basic-bridge <543f8a70-f084-4ad0-bb85-7032b544cc09>
[Oct 28 15:59:38] – Channel SIP/3759-00000000 joined ‘simple_bridge’ basic-bridge <543f8a70-f084-4ad0-bb85-7032b544cc09>
[Oct 28 15:59:40] – Channel SIP/3759-00000000 left ‘native_rtp’ basic-bridge <543f8a70-f084-4ad0-bb85-7032b544cc09>
[Oct 28 15:59:40] – Channel SIP/207.227.236.19-00000001 left ‘native_rtp’ basic-bridge <543f8a70-f084-4ad0-bb85-7032b544cc09>
[Oct 28 15:59:40] == Spawn extension (unrestrictedInternational, 911231231234, 3) exited non-zero on ‘SIP/3759-00000000’
As you can see after the Dial command is sent there is a twenty second wait before the “Called SIP” line. I checked with my provider (Level3), and they don’t see anything from me until 05:59:37
Anyone have any thoughts?
DNS
I would suggest turning SIP debug on (CLI> sip set debug on) and watching the conversation with L3. “core set debug 30” may give you some more info, also. Looks like something is failing over since the call eventually completes.