[Asterisk-Users] SIP - SIP bridge dropping calls?

[Adrian A]

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.

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?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051219/4d560cb9/attachment.htm