[Asterisk-Users] Struggling with grandstream sip to asterisk

Mon Nov 17 23:15:57 MST 2003

<quote who="Walker Haddock">
>
> ; SIP Configuration for Asterisk
> ;
> [general]
> port = 5060                     ; Port to bind to
> bindaddr = 0.0.0.0              ; Address to bind to
> context = default               ; Default for incoming calls
> ;
> [205] ; Conference 2, Grandstream Phone
> callerid="Converence 2" <205>
> username=205
> context=intern
> qualify=yes
> incominglimit=1
> type=friend
> insecure=yes
> host=192.168.1.70
> permit=192.168.0.0/255.255.255.0

                 ^
wrong subnet.


> dtmfmode=info
> canreinvite=no
> reinvite=no
> callgroup=1
> pickupgroup=1
> disallow=all
> allow=alaw
> allow=ulaw
>
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> Asterisk-Users at lists.digium.com
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>


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