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Trunk to trunk transfer calls on SIP trunk, no audio

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StubbedmytoeonIPO

Programmer
Dec 19, 2013
86
GB
I'm having a hard time getting my head around this. All calls thru this SIP provider work fine. When a call is transferred to an outside line, no audio in either direction. The trunk to trunk calls always connect, but rarely have audio. I don't know the exact percentage, but some of the calls that connect have two way audio. conference calls are no problem. This is on a 9.0 500v2. Anyone seen anything like this? Thanks.
 
Research the threads on here as this is a common issue and always seems to point to the network set up

acss sme acis sme acss cm 5.2.1 acss cm and cmm acss aura messaging.
 
RTP keepalives is enabled already.

I did search the threads, did not see a post on this. Did see a lot of one way audio. I feel it is the carrier's equipment but have no way to prove it other than putting a switch in between and port mirror with wire shark.

Thanks for your replies.
 
Your provider may have anti-tromboning in place. Wireshark might be your only option.

ACSS - SME
General Geek

 
Biggest drawback with SIP is there is no real demarcation point and so no fault ownership, unless through much time and effort you prove who it is and then manage to convince them of it to :)

 
Okkie26 what did you set your periodic timeout to and did you enable initial keepalives?
 
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