However I happened to be running rtp debug at one time this happened and here's the log:
Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27353, ts 8114560, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54972, ts 11318620, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27354, ts 8114720, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54973, ts 11318780, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27355, ts 8114880, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54974, ts 11318940, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27356, ts 8115040, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <<User IP>>:36420 (type 0, seq 54975, ts 11319100, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <<VoIP Provider>>:12006 (type 0, seq 27357, ts 8115200, len 160)
Dec 19 19:38:29 DEBUG[7114] channel.c: Didn't get a frame from channel: SIP/298-6427
Dec 19 19:38:29 DEBUG[7114] channel.c: Bridge stops bridging channels SIP/298-6427 and SIP/provider-81d5
Dec 19 19:38:29 DEBUG[7114] chan_sip.c: update_call_counter(1xxxxxxxxx) - decrement call limit counter
Dec 19 19:38:29 DEBUG[7114] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Dec 19 19:38:29 VERBOSE[7114] logger.c: == Spawn extension (default, 91xxxxxxxx, 1) exited non-zero on 'SIP/298-6427'
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.
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).
Again, this is completely random and it's hard to get a packet dump of the whole thing..
Does anyone have any idea why this is happening before I file this as a bug?
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