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 >
If I call from SIP to PSTN, at the beginning of the call (1 second) after getting phone at the PSTN side I hear voice at the SIP side. After this 1 second I don't hear anything in SIP phone (at the PSTN phone everything is OK). Nobody has had any problems like me? 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
