Hi, calling a partner on the other side of a SIP trunk, call gets disconnected immediately after answer. This is the content of log file:
Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel: SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channels SIP/232-2e41 and SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up channel 'SIP/cerved-out-6eba' Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Hangup call SIP/cerved-out-6eba, SIP callid [EMAIL PROTECTED]) Jun 14 16:25:14 DEBUG[14380] chan_sip.c: update_call_counter(9704) - decrement call limit counter Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Updating call counter for outgoing call Jun 14 16:25:14 DEBUG[14380] app_dial.c: Exiting with DIALSTATUS=ANSWER. I have Asterisk 1.2.8 but remote server has 1.2.4. Any help? -- Domenico Viggiani _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
