[asterisk-dev] Strange Asterisk Behaviour - Stuck channels

[Steve Murphy]

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

-- 
Steve Murphy
Software Developer
Digium
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