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

[Kristian Kielhofner]

On 10/1/07, Kevin P. Fleming <kpfleming at digium.com> wrote:
> John Todd wrote:
>
> > 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?
>
> The testing is most welcome, but I believe we will hold off on merging
> until we have come to agreement on the new release policy that is being
> discussed (that Russell outlined on Thursday at Astricon). If we that
> resolved soon, you could see this in a real, supported release much
> sooner than you think!
>
> --
> Kevin P. Fleming
> Director of Software Technologies
> Digium, Inc. - "The Genuine Asterisk Experience" (TM)
>

Kevin,

  Any updates here?  A lot of us would like to see this merged soon.

-- 
Kristian Kielhofner