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

Carlos Chavez cursor at telecomabmex.com
Tue Feb 26 12:24:04 CST 2008


	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 México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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