[asterisk-users] Odd one way RTP on SIP to SIP calls

[Julio Arruda]

Is this a SIP connection or a SIP-T one? Not sure (don't have access to 
my previous life docs :-), but this seems to be a Session Server Trunks 
doing SIP-T, not sure is the configuration you want...Have you tried to 
contact their support ?
PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't 
remember seeing in plain SIP calls, so that is why I suspect is 
configured as a SIP-T.

Örn Arnarson wrote:
> Hi everyone,
> 
> I'm having an odd problem with one way RTP on SIP to SIP calls.
> I have two SIP servers, one is an Asterisk and the remote SIP server
> is a Nortel SIP server.
> 
> When a call comes to the Nortel server through the PSTN and is routed
> to the Asterisk, audio is fine. Two way RTP and no problems. When a
> SIP client registered on the Nortel server calls the Asterisk, the
> Asterisk doesn't seem to send any RTP.
> 
> As far as I can tell, there isn't anything wrong with the call setup.
> 
> show core version shows:
> Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on
> 2007-05-17 06:39:34 UTC
> 
> SIP and RTP debugging on Asterisk shows this:
> http://www.arnarson.net/~orn/calldebug.txt
> 
> On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by
> root @ build.trixbox.org on a i686 running Linux on 2007-04-25
> 19:59:21 UTC) on the same network (same subnet and physical location)
> as the 1.4.4 this problem does not exist. There is no RTP problem when
> SIP clients registered on Nortel call.
> 
> If anyone could help or suggest anything it would be greatly appreciated.
> 
> Best regards,
> Örn
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