[asterisk-dev] Possible bug in either RTP handling or AGI handling - depends on how you look at it

[Olle E. Johansson]

15 aug 2011 kl. 09:45 skrev Nir Simionovich:

> Hi All,
> 
> I've been trying to investigate the scenario I'm going to describe for the past 2 week, however,
> I am unable to pinpoint the exact issue - or to which component it relates directly. I will describe
> what happens, to as much details as I can.
> 
> I have a customer that operates several A2Billing servers (yes, I don't like a2b, but people do).
> The scenario that happens looks like this, as a call is being made on the system and the two ends
> are talking, for no apparent reason, media will suddenly go away to be replaced by a screeching
> high pitch sound, forcing you to disconnect the call.
> 
> Now here is the funny part, once the sound is heard and you disconnect the call, two things
> happen:
> 
> 1. The actual inbound channel remains active, although the channel had been disconnected.
> 2. The channel will remain active for a period of exactly (and I do mean exactly) 16 minutes!
> 3. The AGI script dies once the channel dies, however, will not terminate properly causing a miss
>  of billing.
> 4. And this is the odd part, if the call had lasted 45 minutes till the sound was heard, it's counters
>  in Asterisk, when looking at "core show channels" would show 00:00:00 - in other words, something
>  had caused a complete channel reset or dead-lock somewhere.
> 5. When doing a "sip show channels", the problematic channel will show up as a "BYE" state, and
>  will remain in that state for the 16 minutes - how odd is that?
> 
> The above had been observed on SIP channels - and I've regressed both 1.4.X, 1.6.2.X and 1.8.X,
> all exhibit the same behavior. This can be seen on both development versions and stable versions.
> I hadn't yet tested this with IAX2 channels, however, as the media handling in IAX2 is totally different
> than with SIP, and it would appear that this has some relation to SIP RTP handling - I can assume I
> won't see the issue there. Non the less, the problem is still something that needs to be investigated.
> 
> I've got a system setup that can be used in order to debug the situation.
> 
> Any thoughts anybody? I hadn't posted this on the users list, simply due to the fact that I've ascertained
> this is not a config or a2billing issue.

Try to find out the SIP history (use dumphistory in sip.conf) for the call. It seems like there are SIP packets getting lost. Can't explain the sound or the billing, but maybe the hanging channels. Check that you really get an answer to the BYE.

/O