Le mardi 10 Mai 2005 08:01, Laurent Foulonneau a �crit�: > Hi, Hello,
> > We are testing a SIP solution * + ser solution for a large implementation. > All the clients are nated. > When a client is dialing outside the domain (to a FWD sip account for > example) all is perfect ! ;-) > But ,when a call is done to a sip account, the client is ringing, then the > caller can hear the nated client very well, but the nated client does'nt > hear anything. RTP issue no ? > I've follow the SER, Asterisk and Lucent TNT by Michael > Shuler (http://www.voip-info.org/wiki-Asterisk+at+large ), but I think > I've missed something. > The * and ser are using public ip, no nat for them. > I've tried different config, with and without rtpproxy, with forward > instead of t_relay, but same or more problems. > > If someone could help me please. > > Here are my conf files : > Try to add canreinvite=yes in general section of sip.conf, and switch it to yes for the ser server. ++ -- Guy Decarpentrie - ipnotic - switch to ip Responsable syst�me Tel / Fax : 01.72.29.05.08 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
