<div>Sounds like there maybe a codec issue. If you are using g729, make sure you have licenses.</div>
<div> </div>
<div>bp<br><br> </div>
<div><span class="gmail_quote">On 6/14/06, <b class="gmail_sendername">Mimmus</b> <<a href="mailto:dviggiani@tiscali.it">dviggiani@tiscali.it</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi,<br>calling a partner on the other side of a SIP trunk, call gets disconnected<br>immediately after answer. This is the content of log file:
<br><br>Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel:<br>SIP/cerved-out-6eba<br>Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channels<br>SIP/232-2e41 and SIP/cerved-out-6eba<br>
Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up channel<br>'SIP/cerved-out-6eba'<br>Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Hangup call SIP/cerved-out-6eba,<br>SIP callid <a href="mailto:362258b02bbafa8117eecbb6755837a0@10.97.1.254">
362258b02bbafa8117eecbb6755837a0@10.97.1.254</a>)<br>Jun 14 16:25:14 DEBUG[14380] chan_sip.c: update_call_counter(9704) -<br>decrement call limit counter<br>Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Updating call counter for outgoing
<br>call<br>Jun 14 16:25:14 DEBUG[14380] app_dial.c: Exiting with DIALSTATUS=ANSWER.<br><br>I have Asterisk 1.2.8 but remote server has 1.2.4.<br><br>Any help?<br>--<br>Domenico Viggiani<br><br>_______________________________________________
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