[asterisk-dev] Strange Asterisk Behaviour - Stuck channels

[John Lange]

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,
-- 
John Lange
www.johnlange.ca