[Asterisk-Users] Problem with Vegastream 50 BRI

[Olle E. Johansson]

Michael Devenijn wrote:

> Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ...
> 
> 
> sip.conf extract : 
> 
> [gw001]
> type=friend
> host=dynamic
> defaultip=192.168.0.12
> nat=no
> dtmfmode=rfc2833
> canreinvite=yes
> qualify=no
> context=tlsgw
> 
> 
> 
> extensions.conf extract (from the contact [tlsgw]) :
> 
> exten => 57228047,Dial(SIP/cs001,40,tr) 
> ...

Looking for 57228047 in sip
Transmitting (no NAT):
SIP/2.0 404 Not Found

Asterisk isn't looking in the context tlsgw, for some reason it checks in the "sip" context.
If this is your default context, Asterisk doesn't connect the incoming call with gw001.

You have host=dynamic - is the gateway registred with Asterisk at all?

/O