[Asterisk-Users] Calls dropping off

[Steve Foy]

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