[Asterisk-Users] Problem with Vegastream 50 BRI

[Michael Devenijn]

sorry, the sip extract is from a previous test now i get the same problem but with 
looking for 57228047 in tlsgw and it's the same error, it searching in this direction : 

why are the 2 ast values 0 ??

Non-codec capabilities: us - 1, them - 0, combined - 0 

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Olle E.
Johansson
Sent: Saturday, March 20, 2004 11:05 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Problem with Vegastream 50 BRI


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
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