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

[Örn Arnarson]

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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