On 26-07-2005 at 07:23:39PM +0200, [EMAIL PROTECTED] wrote: > Hello, > I have something like this: > SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN > > After calling from SIP to PSTN (and from PSTN to SIP too) > I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. > > I have another network with another h323/sip (in the place of asterisk) > and there everything is OK. > > In AUDIOCODES logs I see that everything goes fine with asterisk, but SIPUSER > can't hear the PSTN user. >
The problem was in oh323.conf: fastStart=no (was yes) Now everything goes fine. Bartek. _______________________________________________ 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
