Yeah, I suppose so... :) There is no NAT here, however. All public IPs. The root of the issue isn't NAT fixing, it's that Mediaproxy doesn't seem to use properly the new information from a reinvite.
Failing call flow is: PSTN Gateway -> Asterisk w/ reinvites -> Opensips -> SIP Phone* * Note: "SIP Phone" is really an Asterisk box with a Polycom behind it, but it's not doing anything screwy. No reinvites from this one. I can reproduce the same behavior with a Sipura or Polycom registered directly to Opensips. It's just much harder to test because I don't have any extra public IPs available in my "home" lab. - Jeff On 7/11/09 4:52 PM, "Raúl Alexis Betancor Santana" <r...@dimension-virtual.com> wrote: > If you want to improve readability .. just don't use IP's from a same range in > a capture that is supposed to be about NAT fixing ... ;-) > > After a first read ... your call flow is Asterisk -> OpenSIPS -> Asterisk -> > SIP Phone ? ... or Asterisk -> OpenSIPS -> SIP Phone (beging the same NAT > router as Asterisk) ? > > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users