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 <[EMAIL PROTECTED]> 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 <[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