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 > _______________________________________________ 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
