Transcoding Problem on SIP calls

Unanswered Question
Feb 16th, 2010
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Hi,


we have a setup as follows:


Location A

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Voice Gateway pair at Location A ( 2811 - SP Services )

SIP server at location A

A stream server behind this SIP server from which we get announcements.

Unified CM 7.0 Publisher and Subscriber

Transcoder and Conf Bridge on 2811 GWs

MTP on UCM

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Location B

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Voice Gateway pair at Location B ( 2811 - SP Services )

SIP server at location B

A stream server behind this SIP server from which we get announcements.

Unified CM 7.0 Subscriber

Transcoder and Conf Bridge on 2811 GWs

MTP on UCM

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On UCM, SIP trunks are configured for all 4 voice gateways.

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I am calling from an IP phone to simulate an incoming call to the environment.

The numbers I am calling are defined on teh SIP server.


If I call the number defined on the location A SIP server, call establishes without any problems.

If I call the number defined on the location B SIP server,  I can hear the announecements, call establishes, but at the end no audio onboth ways.


*** During this call without audio, if I receive a call from someone else ( internal ) and put the first call on hold, when I resume it, I start hearing audio both ways. Sometimes very clear, sometimes with a lot of noise.


I am sure about the routing and firewalls. All the phones and trunks, etc in location A is in DP_A device pool, all the phones and trunks, etc in location B is in DP_B device pool. Their regions are configured so that, in each DP they will be using g.711 and in between device pools, g.729.


If I put everything in the same DP, both scenarios work perfectly.


I atttached the config of one of the voice gateways in location B ( all 4 of them have exactly the same config , I tried two IOS versions without success )


Any ideas ?

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