Sounds like there maybe a codec issue. If you are using g729, make sure you have licenses.
bp
On 6/14/06, Mimmus [EMAIL PROTECTED] wrote:
Hi,calling a partner on the other side of a SIP trunk, call gets disconnectedimmediately 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-6ebaJun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channelsSIP/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 counterJun 14 16:25:14 DEBUG[14380] chan_sip.c: Updating call counter for outgoing
callJun 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___
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