[asterisk-users] AMD on a SIP trunk...

[BJ Weschke]

 Add an answer() and a playback of 1 second of silence or something else 
to make sure the RTP is nailed up. AMD can/will hang if it has no media 
to analyze.

Carlos Chavez wrote:
> 	We have an Asterisk server with a small outgoing call center.  We use
> AMD and it usually works very well on Zap channels (E1 PRI).  We added a
> couple of SIP trunks to reduce long distance costs but now AMD gets
> stuck when the call goes out through the SIP channels.  Here is an
> example call using a SIP line:
>
>     -- Executing [016566275538 at CC2:1]
> Set("Local/016566275538 at CC2-dad7,2", "CIDTEMP="49875&calllogId=135514"
> <016566275538>") in new stack
>     -- Executing [016566275538 at CC2:2]
> Dial("Local/016566275538 at CC2-dad7,2", "SIP/juarez-60/6275538|25|C") in
> new stack
>     -- Called juarez-60/6275538
>     -- SIP/juarez-60-0892f740 is making progress passing it to
> Local/016566275538 at CC2-dad7,2
>     -- SIP/juarez-60-0892f740 answered Local/016566275538 at CC2-dad7,2
>     -- Executing [406 at CC:1] Answer("Local/016566275538 at CC2-dad7,1", "")
> in new stack
>     -- Executing [406 at CC:2] AMD("Local/016566275538 at CC2-dad7,1", "") in
> new stack
>     -- AMD: Local/016566275538 at CC2-dad7,1 016566275538 (null) (Fmt: 64)
>     -- AMD: initialSilence [3500] greeting [2500] afterGreetingSilence
> [800] totalAnalysisTime [5000] minimumWordLength [100]
> betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold
> [256] 
>
> 	AMD just stops and it takes over a minute until the line is dropped.
> The same number dialed through Zap works without a hitch.  What could be
> the reason?  If I dial the same number without AMD I can talk to the
> other person so I know the SIP line is fine.
>
>
>   
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