[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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