On Saturday 11 July 2009 22:16:27 you wrote: > 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.
I will not question why are you trying to use Mediaproxy if not for NAT fixing .. X-) > 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. For properly handling the re-invite, did you call force_rtp_proxy INSIDE the in-dialog procces ? -- Raúl Alexis Betancor Santana Dimensión Virtual _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
