[asterisk-users] Incoming

[Artie Gold]

This is a quite promising idea. Many thanks.
I'll post my results to the list...

Cheers,
--ag

On Fri, Jul 11, 2008 at 11:22 AM, Tilghman Lesher <
tilghman at mail.jeffandtilghman.com> wrote:

> On Friday 11 July 2008 09:17:37 Artie Gold wrote:
> > In updating to 1.4.21 recently, we've encountered a problem, when running
> > over a satellite connection (where the latency is considerable; a
> "regular"
> > internet connection did not exhibit this problem), where incoming calls
> are
> > being dropped as a result of the sip handshake timing out (dropping down
> to
> > 1.4.18.1 solved the problem for us). From reading the change logs and
> other
> > posts, it seems that some work has been done in this area recently to get
> > it "right"; it appears that, at least in the satellite case, things may
> > have gotten a little too "tight"...
> >
> > If this rings a bell for anyone, any insight would be appreciated.
>
> Try setting t1min to something higher than the default, 100 (ms).  This
> value
> is settable globally, as well as per-peer.
>
> --
> Tilghman
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Artie Gold
F4W, Inc.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080711/9d2fc9e6/attachment.htm