[Asterisk-Users] Calls dropping off
Steve Foy
steve at unite.net
Fri Jan 30 10:14:27 MST 2004
Thanks, I'll try that and see how it goes.
Cheers,
Steve
On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote:
> Try adding it to the phones involved so it looks like this:
>
> ; Shirley
> [100]
> type=friend
> username=xxx
> secret=xxx
> host=dynamic
> dtmfmode=rfc2833
> callerid="Shirley O'Neill" <100>
> context=internal
> mailbox=100 at default
> qualify=yes
> canreinvite=no
>
> -b
>
>
> Quoting Steve Foy <steve at unite.net>:
>
> > Bill,
> >
> > On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
> > > Shot in the dark here ...
> > >
> > > Do you have:
> > >
> > > canreinvite=no
> > >
> > > Set in sip.conf for the SIP phones in question ?
> >
> > No, I don't.
> >
> > All I have in sip.conf is the general stuff like:
> >
> > [general]
> > port = 5060 ; Port to bind to (SIP is 5060)
> > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
> >
> > allow=all
> > allow=GSM
> > allow=G729
> > allow=iLBC
> > allow=SpeeX ; Allow all codecs
> > allow=ulaw
> >
> > and then about 10 friends like this:
> >
> > ; Shirley
> > [100]
> > type=friend
> > username=xxx
> > secret=xxx
> > host=dynamic
> > dtmfmode=rfc2833
> > callerid="Shirley O'Neill" <100>
> > context=internal
> > mailbox=100 at default
> > qualify=yes
> >
> > --
> > Steve Foy | http://www.unite.net
> > UNITE Solutions | Tel: 028 9077 7338
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--
Steve Foy | http://www.unite.net
UNITE Solutions | Tel: 028 9077 7338
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