Hi Jeff, in the failed case, OpenSIPS thinks that the previous hop was a strict router (this is why you have the after_strict). This detection is done based on the RURI - it looks if the domain part of RURI is a local domain or not. This test includes: - test against all listen IPs - test against all aliases (from script) - test against the domains from the "domains" module.
is the ww.xx.yy.82 somehow listed in the "domains" module ? Regards, Bogdan Jeff Pyle wrote: > Ok, debugs: > > On the reINVITE that fails: > > DBG:core:grep_sock_info: checking if host==us: 11==11 && [ww.xx.yy.82] > == [ww.xx.yy.83] > DBG:core:grep_sock_info: checking if port 5060 matches port 5060 > DBG:rr:after_strict: Next hop: > 'sip:[email protected]:5060;lr=on;ftag=as3137fec5;did=b92.be47e975' > is loose router > > On the reINVITE that works: > > DBG:core:grep_sock_info: checking if host==us: 12==11 && [ff.gg.hh.94] > == [ww.xx.yy.83] > DBG:core:grep_sock_info: checking if port 5060 matches port 5066 > DBG:core:check_self: host != me > DBG:core:grep_sock_info: checking if host==us: 11==11 && [ww.xx.yy.83] > == [ww.xx.yy.83] > DBG:core:grep_sock_info: checking if port 5060 matches port 5060 > DBG:rr:after_loose: Topmost route URI: > 'sip:[email protected]:5060;lr=on;ftag=as27ab13f7;did=37e.68fb59e7' > is me > > ww.xx.yy.83 is Opensips. > ww.xx.yy.82 is Asterisk 1.2.26 (IP is right next to Opensips) > ff.gg.hh.94 is Asterisk 1.4.23.1 (IP is completely different than > Opensips) > > I see that the failing reINVITE causes a “DBG:rr:after_strict” while > the one that succeeds causes a “DBG:rr:after_loose”, but I’m not sure > what that means here. I suppose the next stop is the RFC. > > If anyone has any thoughts on this one, I’d much appreciate hearing them. > > > Thanks, > Jeff > > > On 3/21/09 11:12 AM, "Jeff Pyle" <[email protected]> wrote: > > > Hello, > > > > I seem to be having a problem with loose_route() not properly > detecting when > > a Route set is its own. Opensips 1.5 build 5491, same PSTN carrier in all > > cases. > > > > Flow is Asterisk --> Opensips --> PSTN (Sonus NBSe) > > > > The call sets up properly. 90 or 120 seconds into the call, the PSTN > > carrier sends a reINVITE to refresh the session. If Asterisk 1.4.23.1 is > > the UAC, all is well. If Asterisk 1.2.26 is the UAC, Opensips > misidentifies > > the Route header in the carrier’s reINVITE as foreign. The t_relay then > > routes the packet to itself and bad things happen. > > > > I’ve done stare-’n-compares on the packets in all cases. The reINVITE > from > > the carrier is almost exactly the same. The differences are as follows: > > > > - The Asterisk 1.4.23.1 UAC is using port 5066, where Asterisk 1.2.26 > uses > > 5060. This difference is reflected in the RURI of the reINVITE. > > > > - The To field of the 1.4.23.1 UAC has the :5066 at the end of the > URI; the > > 1.2.26 host does not have a port. > > > > That's it. The only other difference I can see in the messaging is > that the > > 1.2.26 host puts "received=ww.xx.yy.zz" in its Via header of the initial > > transaction, where the 1.4.23.1 host does not. But the reINVITE is a new > > transaction so I don't know how this could effect affect the reINVITE. > > > > I am at a loss. Any thoughts? > > > > > > Thanks, > > Jeff > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
