Bogdan, Thanks for responding. I am using vitelity for my upstream; I will send them a ticket. If they fail to act, do you have any suggestions...switch carriers? any config change?
Thanks again, Brad On Tue, Feb 26, 2013 at 7:58 AM, Bogdan-Andrei Iancu <bog...@opensips.org>wrote: > ** > Hi Brad, > > Thinks are a bit more complicated, it seems.... > > In the INVITE your opensips sends to 64.....93 IP, you have the Contact > with 192.168.1.21 (priv IP of asterisk). > > When you receive the BYE from 64.....93 IP, the Route hdrs are ok (the 2 > hdrs added by opensips to reflect the interface exchange), but the RURI is > wrong - it must be the contact from the INVITE you sent, but it seems to be > the IP of your opensips - this makes opensips to do act as strict router > and not like a loose router....and routing gets broken. > > So, the 64.....93 party or some other behind it, screw up the Contact in > the your INVITE and this alters the in-dialog requests - you should check > with the upstream guys. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 02/25/2013 04:36 PM, brad smith wrote: > > I just tested an outbound call (Asterisk originate) without bridging and > get the same '404 not here' if that helps. > > Thanks again, > Brad > > > On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu <vladp...@opensips.org> wrote: > >> Hello, >> >> Seems the incoming BYE does not have any Route headers, and the >> loose_route() function returns false. >> >> Since you have dialog support in your script, try >> >> if (has_totag()) { >> # sequential request withing a dialog should >> # take the path determined by record-routing >> if (loose_route() || match_dialog()) { >> >> >> This way you will force matching of dialog sequential requests that have >> no Route headers. >> >> Best Regards, >> >> Vlad Paiu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> >> On 02/24/2013 02:57 AM, brad smith wrote: >> >> Hello, >> >> I am currently running opensips 1.8.1 no tls. It is >> multi-homed with a public and private address. >> I have a asterisk >> 1.8.19 in the lan that is connected to opensips via lan >> address. >> >> >> *issue* >> A caller calls in >> and then I place an outbound call and finally bridge the two >> calls. >> This works as >> expected, except when the outbound caller hangs up first the >> BYE never gets back to Asterisk. >> I can see the BYE >> reach OpenSips but a '404 not here' is returned to the ISP. >> >> >> >> >> sip trace https://gist.github.com/5009662 >> >> >> opensips.cfg https://gist.github.com/5009704 >> >> >> >> >> >> >> thanks for your time. >> >> >> _______________________________________________ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > _______________________________________________ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
_______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users