[asterisk-users] Incoming

[Artie Gold]

sip

Thanks,
--ag

On Tue, Jul 15, 2008 at 8:39 PM, Matt Riddell <matt at venturevoip.com> wrote:

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> Artie Gold wrote:
> > Folks:
> >
> > This is my first post, so please let me know if I transgress in any
> way...
> >
> > 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.
>
> These calls sip or iax?
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
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-- 
Artie Gold
F4W, Inc.
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