[asterisk-dev] Session-Timers patch for SIP, anyone?

[asterisk]

Thank you John!   I was sitting in front of you at the dev summit at
astricon and brought this subject up.   This patch is MOST welcome.  I
will need to get it merged into my asterisk. Then I will test and keep
you posted..

Doug


-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of John Todd
Sent: Monday, October 01, 2007 5:02 PM
To: asterisk-dev at lists.digium.com
Subject: [asterisk-dev] Session-Timers patch for SIP, anyone?


So at Astricon, there were quite a few sidebar conversations on this 
patch for Session-Timers in chan_sip.  Everyone wholeheartedly agreed 
that the patch was a good idea, and that the documentation brought a 
tear to everyone's eye.  It's been tested on three systems (two under 
my control, one under the developer's control.)  So far, so good.

I had hoped to pester enough people at Astricon this year to get it 
submitted.  The problem, as I see it, was that the CodeZone was a bit 
more <ahem> rowdy than in years past, and I suspect not a lot of 
concentrated work was possible due to distance from the main 
conference center and proximity of the pool and bar (a dangerous 
geography, to be sure!) as well as other distractions which kept 
attention focused on areas where there were more (or less) bikinis 
and no keyboards.

Therefore, it is incumbent upon some other people here on the list to 
try this patch out, and/or hopefully submit it to TRUNK before it 
becomes hopelessly un-merge-able.  This feature is a slam-dunk for 
any service provider, since it kills zombie calls from both 
directions in the case of an RTP-less session which otherwise would 
go forever.   Saves money, tastes great, low fat - who could want 
more?

http://bugs.digium.com/view.php?id=10665

JT


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