[asterisk-dev] Strange Asterisk Behaviour - Stuck channels
creslin at digium.com
Wed Jul 30 13:57:24 CDT 2008
> 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.
Can you give any indication of what type of work load you are running?
I have customers running trunk that do (I suspect) far more traffic
(over 100,000 TDM SS7->IAX) calls per day and I have not seen a leak of
this magnitude yet.
> -----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?
> Which version of Asterisk are you seeing this in? 1.4? trunk?
> Steve Murphy
> Software Developer
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