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

[Örn Arnarson]

Sorry for the spam, but there was a typo. I was running ISN09, but the
upgrade was to ISN09u, which I am currently running. That was the
upgrade that caused the interoperability problem with Asterisk that I
mentioned.

On 10/1/07, Örn Arnarson <orn at arnarson.net> wrote:
> Good point. Here goes.
>
> I am running ISN09 (recently upgraded). Actually the upgrade caused a
> lot of problems and now the CS2K has to be datafilled so that the
> Asterisk trunks are Q764 and not Q767, lest the calls fail.
> Additionally the NGSS/SST had to be patched up to date to fix another
> issue.
>
> The NGSS config is pretty straight forward, no fancy options set. In
> this version of * I had to change the following options to make it
> work with this version of Asterisk:
> Use OPTIONS for Heartbeat: No
> Enforce CODEC-Compatibility: No (oddly enough, as the codecs are compatible)
> Accepts Encapsulated ISUP: No
>
> sip.conf entry is like this:
> [Nortel-SIP]
> type=friend
> host=1.1.1.1
> port=5060
> dtmfmode=rfc2833
> canreinvite=no
> disallow=all
> allow=alaw
> allow=ulaw
> context=default
>
> I think most of the other options were left at default, even though I
> don't think that they are crucial.
>
> Best regards,
> Örn
>
> On 10/1/07, Julio Arruda <jarruda-asterisk at jarruda.com> wrote:
> >
> > Just a guess in fact..but..
> > I'm sure others would love to know how is the NGSS (SST now ?) config
> > for this purpose, as well as your sip.conf and etc (one note, you are
> > running SN09 or ISN09 ?
> > Not sure, but this also would help others out there.. :-)
> >
> >
> >
> > Örn Arnarson wrote:
> > > 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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