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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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 > >>>> _______________________________________________ > > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
