[asterisk-dev] SIP ghost calls

[Jonathan k. Creasy]

I've seen this behavior on our systems. A single SIP channel can be
there for what could be indefinitely. I've seen them stay for days until
I do a hang up on that channel. 

In my case the SIP phone is bridged to an IAX channel and the SIP phone
will go away for some reason. The IAX channel will not be there but the
SIP channel will be on the PBX for days. I'll usually see the SIP
channel on my MRTG graph during the night hours and know that I need to
go in and hang it up. 

Not really offering any suggestions here but wanted to contribute the
symptoms I have observed. 

-Jonathan

> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-
> bounces at lists.digium.com] On Behalf Of Johansson Olle E
> Sent: Thursday, November 30, 2006 2:36 PM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] SIP ghost calls
> 
> 
> 30 nov 2006 kl. 20.15 skrev Klaus Darilion:
> 
> > On Thu, November 30, 2006 19:57, Hans Petter Selasky said:
> >> On Thursday 30 November 2006 18:57, Johansson Olle E wrote:
> >>> The proper way would be to develop the SIP timer extension.
> >>>
> >>> That would also solve the issue when you have external RTP
bridges.
> >>>
> >>
> >> http://www.jdrosen.net/sip_timer.html
> >>
> >> http://www.jdrosen.net/papers/draft-ietf-sip-session-timer-14.txt
> >>
> >
> > its already an RFC: 4028
> >
> Thanks for the update!
> 
> >> When I turn on SIP debug, I see that my phone supports the "timer"
> >> extension.
> >>
> >> Are there any plans to get this into Asterisk?
> >>
> >> It should not be that difficult?
> 
> All depends on community efforts and funding...
> 
> /O
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