[Asterisk-Users] Calls dropping off
Eric Wieling
eric at fnords.org
Mon Feb 2 10:16:16 MST 2004
Do you have busydetect=yes and/or callprogress= in zapata.conf? If so
set them to no.
On Mon, 2004-02-02 at 11:10, Steve Foy wrote:
> Hi,
>
> On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote:
> > Steve,
> >
> > this really is a FAQ. You need add to EACH (!) sip user something like
> >
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=gsm
>
> I do have that in my sip.conf. I am using ulaw.
>
> Calls from the SIP phones through Asterisk and out one of my X100P cards are
> working 95% of the time and also, incoming calls through the X100P cards to
> the SIP phones are the same.
>
> The only problem is that every once in a while, without any odd circustances
> that I can see, the call just drops and the remote user is gone.
>
> The box running asterisk isn't under heavy load, so I can't see why this is
> happening.
>
> I am not using g.729 or 723, just plain old ulaw, which I have got enabled in
> sip.conf
>
> Cheers,
> Steve
--
Go to http://www.digium.com/index.php?menu=documentation and look at
the "Unofficial Links" section. This section has links to a wide
variety of 3rd party Asterisk related pages. My page is the
"Asterisk Resource Pages".
BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643
More information about the asterisk-users
mailing list