[asterisk-dev] Strange Asterisk Behaviour - Stuck channels

[John Lange]

"current release versions" which more specifically means 1.4.21.1 &
1.4.21.2 but it has been occurring for a while so at least 1.4.19 is
also effected but it could go even further back.

Unfortunately when it first started happening I didn't recognize it as a
bug with Asterisk.

For the first many times this happened I wrongly assumed that the
garbled audio was a problem with the phone, not with Asterisk.

In hindsight I realize that normally one of the channels is just dead
air which causes a choppy "roboto" sound on the other audio channel
which sounds like a malfunctioning phone handset.

It was only by fluke that yesterday the one of the two bridged audio
channels happened to be music on hold which I could distinctively hear.
So for the first time I realized that the problem was much different
that I originally thought.

I then did a packet capture (tcpdump) and was able to see RTP packets
streaming to the phone even the phone was on-hook.

Keeping the call open I then issued a "soft hangup" to the other channel
and the audio cleared up and everything was back to normal.

Previous to this event I had never connected the garbled audio problem
with the stuck channels.

-- 
John Lange
www.johnlange.ca


On Wed, 2008-07-30 at 11:54 -0600, Steve Murphy wrote:
> On Wed, 2008-07-30 at 11:49 -0500, John Lange wrote:
> > I have discovered a bug in current release versions of Asterisk which
> > causes channels to be "stuck" open forever.
> > 
> > The problem is this behaviour seems to be very hard to replicate and so
> > it is very difficult to file a meaningful bug report.
> > 
> > I'll briefly describe the issue and I hope to get some advice on the
> > best way to capture additional information that could be put into a bug
> > report.
> > 
> > Occasionally we see Sip/Zap bridged channels "stuck" in an open state.
> > Despite the actual call having terminated the channels never go away and
> > we are forced to do a "soft hangup".
> > 
> > But here is where it gets strange. Asterisk continues to send RTP audio
> > packets to the SIP endpoint even though the device is on-hook and not
> > part of any call.
> > 
> > If the user attempts to make another call using the same device, the RTP
> > audio is interpreted by the device as being part of the new audio stream
> > and the device attempts to "mix" the two RTP streams together causing
> > garbled audio.
> > 
> > What info would I need to capture in order to file a bug report and help
> > with debugging this issue?
> > 
> > Regards,
> 
> Which version of Asterisk are you seeing this in? 1.4? trunk?
> 
> murf
> 
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