asterisk users - Dec 2005 - SIP - SIP bridge dropping calls?

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?
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In addition to having this with my SIP phones, I have also experienced
it with SCCP.

 

It started when I updated to the 1.2 release of asterisk. At the time I
updated I also switched VoIP providers and thought it was them.

 

Did you file this as a bug or find a solution to it? Thanks!

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I was able to get a full debug report (packet dump and asterisk debug) for
one of these dropped calls and it does seem to be the provider that is at
fault.  I can see that they stop sending RTP packets to Asterisk when this
happens and after a while they send a BYE.  I will keep investigating
though.

On 12/22/05, David C. Nicosia <david@alchetec.com>
wrote:
>
>  In addition to having this with my SIP phones, I have also experienced it
> with SCCP.
>
>
>
> It started when I updated to the 1.2 release of asterisk. At the time I
> updated I also switched VoIP providers and thought it was them.
>
>
>
> Did you file this as a bug or find a solution to it? Thanks!
>
> _______________________________________________
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