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[cursor at telecomabmex...]

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.
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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[bweschke at gmail.com]

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: