[asterisk-dev] Strange Asterisk Behaviour - Stuck channels

[Venefax]

I want to point out that Asterisk Trunk has big memory or handle leak. I
leave it running for three days and the memory climbs to 1.5 GB until I have
to restart it. I have a changing amount of calls but I can get to 360 open
calls at peak. The amount of calls can go down, but the memory never gets
released. Also the number of files open climbs until it kills the OS.
My machine is open for inspection. I don't open a bug because the bug
marshals require traces and stuff that I cannot obtain from a running
system, where I have 100% of my business.


-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Steve Murphy
Sent: Wednesday, July 30, 2008 1:55 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Strange Asterisk Behaviour - Stuck channels

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