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Summary:ASTERISK-06389: SIP bridging fails with no audio unless...
Reporter:gewfie (gewfie)Labels:
Date Opened:2006-02-21 19:33:46.000-0600Date Closed:2006-05-11 10:00:52
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Transfers
Versions:Frequency of
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Description:I am trying to bridge an incoming SIP call to an outbound SIP call using the same provider.  The only successful attempt is when there is actually some for of audio signal sent on the incoming SIP call (ie; voice or dtmf).

Here are example dial plans that fail (failing being that no audio is passed when Asterisk bridges the calls):

[mobiledivert]
exten => _X.,1,Answer
exten => _X.,2,Dial(${VOIPTRUNK}/03XXXXXXXX) ; number hidden for privacy
exten => _X.,3,Hangup

This is what appears in the log file with the call:
Feb 22 12:37:41 VERBOSE[30845] logger.c:     -- Got SIP response 500 "Internal Server Error" back from 202.61.13.40
Feb 22 12:37:41 VERBOSE[31778] logger.c:     -- SIP/ace1-b0a8 is circuit-busy
Feb 22 12:37:41 DEBUG[31778] chan_sip.c: update_call_counter(03XXXXXXXX) - decrement call limit counter
Feb 22 12:37:41 VERBOSE[31778] logger.c:   == Everyone is busy/congested at this time (1:0/1/0)
Feb 22 12:37:41 DEBUG[31778] app_dial.c: Exiting with DIALSTATUS=CONGESTION.

[mobiledivert]
exten => _X.,1,Answer
exten => _X.,2,Wait(2)
exten => _X.,3,Dial(${VOIP}/03XXXXXXXX)
exten => _X.,4,Hangup

Feb 22 12:41:26 VERBOSE[31788] logger.c:     -- SIP/ace1-3263 answered SIP/039012XXXX-7afa
Feb 22 12:41:26 VERBOSE[31788] logger.c:     -- Attempting native bridge of SIP/039012XXXX-7afa and SIP/ace1-3263
Feb 22 12:41:28 DEBUG[30845] acl.c: ##### Testing 192.168.1.6 with 192.168.1.0
Feb 22 12:41:28 DEBUG[30845] chan_sip.c: Stopping retransmission on '76111fb33d60d0da471920836365944a@192.168.1.3' of Reques$
Feb 22 12:41:39 DEBUG[31788] channel.c: Didn't get a frame from channel: SIP/039012XXXX-7afa
Feb 22 12:41:39 DEBUG[31788] channel.c: Bridge stops bridging channels SIP/039012XXXX-7afa and SIP/ace1-3263
Feb 22 12:41:39 DEBUG[31788] chan_sip.c: update_call_counter(03XXXXXXXX) - decrement call limit counter
Feb 22 12:41:39 DEBUG[31788] app_dial.c: Exiting with DIALSTATUS=ANSWER.

[mobiledivert]
exten => _X.,1,Answer
exten => _X.,2,ResponseTimeout(3)
exten => _X.,3,Dial(${VOIP}/03XXXXXXXX)
exten => _X.,4,Hangup

Results the same as above.

Now on to the dial plan that works and opens the audio on the bridge.

[mobiledivert]
exten => _X.,1,Answer
exten => _X.,2,Read(Temp,vm-password,1)
exten => _X.,3,ResponseTimeout(3)
exten => _X.,4,Dial(${VOIP}/03XXXXXXXX)
exten => _X.,5,Hangup

This doesn't require a digit to be entered in from the user that is initiating the incoming call into the Asterisk server.  As soon as it hits the response timeout it is opening the audio on the bridge when it does the Dial cmd.

Is there a bug to be fixed with this?  Or should I just use the above workaround?  Or is there another workaround that someone knows about?  

I'm happy to do any testing with this that may be required.

-gewfie
Comments:By: Olle Johansson (oej) 2006-03-02 11:52:51.000-0600

You need to make sure that the device sending incoming audio does not use silence suppression. Seems like you get no audio and everyone waits for everyone. Asterisk is timed on incoming audio, so we need all audio, regardless of silence.
By: gewfie (gewfie) 2006-03-02 21:25:00.000-0600

Hi oej,

I have tested this setup with an IAX trunk from another Asterisk server.  So calling into this IAX trunk and then out via the SIP trunk.  Only until I issue the Read and ResponseTimeout does the audio on the bridge open.

I have also tested it by dialing into the SIP trunk and then out through the IAX trunk with the same results as above.  

So not sure if it is a generic issue or sip channel specific.

-gewfie
By: Serge Vecher (serge-v) 2006-05-03 12:01:12

gewfie: there was a feature committed to trunk (http://bugs.digium.com/view.php?id=5374) a while ago that allows Asterisk to have it's own timing on an RTP stream.

Would that be useful in your situation?
By: Serge Vecher (serge-v) 2006-05-11 10:00:52

No response from the reporter. Please reopen the bug if the issue still needs addressing. Thanks.