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

[Örn Arnarson]

Julio,

It seems you had something going there; I disallowed ISUP messages on
the SIP-T server and now I have two way audio.

Thanks a lot for your help!

Best regards,
Örn

On 10/1/07, Örn Arnarson <orn at arnarson.net> wrote:
> You are right, the remote server is a SIP-T.
>
> I haven't had any problems connecting it to regular SIP servers
> thusfar though. Also like I mentioned, I don't have this one-way RTP
> problem with an earlier version of Asterisk.
>
> Thanks for your reply,
> Örn
>
> On 10/1/07, Julio Arruda <jarruda-asterisk at jarruda.com> wrote:
> > 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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