Asterisk 1.2.1 installation. It seems that calls are being dropped for no valid reason, completely random, in the middle of the call. I first thought that it was either the network or the VoIP provider dropping packets and confusing Asterisk into hanging up the call.
<br><br>However I happened to be running rtp debug at one time this happened and here's the log:<br><br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27353, ts 8114560, len 160)
<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54972, ts 11318620, len 160)<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27354, ts 8114720, len 160)
<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54973, ts 11318780, len 160)<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27355, ts 8114880, len 160)
<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54974, ts 11318940, len 160)<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27356, ts 8115040, len 160)
<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54975, ts 11319100, len 160)<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27357, ts 8115200, len 160)
<br>Dec 19 19:38:29 DEBUG[7114] channel.c: Didn't get a frame from channel: SIP/298-6427<br>Dec 19 19:38:29 DEBUG[7114] channel.c: Bridge stops bridging channels SIP/298-6427 and SIP/provider-81d5<br>Dec 19 19:38:29 DEBUG[7114] chan_sip.c: update_call_counter(1xxxxxxxxx) - decrement call limit counter
<br>Dec 19 19:38:29 DEBUG[7114] app_dial.c: Exiting with DIALSTATUS=ANSWER.<br>Dec 19 19:38:29 VERBOSE[7114] logger.c: == Spawn extension (default, 91xxxxxxxx, 1) exited non-zero on 'SIP/298-6427'<br><br>Asterisk complains that it did not get a frame from the channel...but everything seems to be OK...unless there is another type of frame that it's expecting. The channel that Asterisk thinks it "did not receive a frame" from is then left with dead air.
<br><br>All messages that I've seen on the list regarding the "Didn't get a frame from channel" error seem to refer to failure in call setup (ie. call dropped as soon as it is answered), however in this case it happens *during* the call, while the two parties are speaking (so not a silence timeout issue).
<br><br>Again, this is completely random and it's hard to get a packet dump of the whole thing..<br><br>Does anyone have any idea why this is happening before I file this as a bug?<br><br> <br><br><br>