[Asterisk-Users] Dial via sip gateway?

[Olle E. Johansson]

Rich Adamson wrote:

>>>What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
>>>
>>>Been trying stuff similar to:
>>>exten => _6X.,1,Dial(SIP/3091 at 205.22.93.1/${EXTEN-1})
>>>where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
>>>even try the IP.
>>>
>>>Rich
>>>
>>
>>from my extensions.conf:
>>
>>;******************************************************
>>[trunk-local]
>>;******************************************************
>>exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:1}@mediatrix-1204)
> 
> 
> The above does not seem to work either. Since the mediatrix has four pstn
> ports, there must be a way to construct a Dial command that would embed
> a userid:password, port alias name, or something like that. Just can't find
> any reference to what that syntax would look like. (The gateway is properly
> handling incoming pstn calls, just not the outgoing pstn attempts.)
> 
> Really need to the sip dial command to include...
>   - the string of digits to be called
>   - either a userid:password, or, port alias name (or both)
>   - ip address of the gateway
> 
> Anybody have a clue what that dial sip syntax would look like????
Yes, it's
    SIP/extension at host
There's no 'sub-extension'.

So SIP/extension at mediatrix is the proper way to go, where extension is
the string of digits to be called. If the box acts as a SIP proxy, you
might need to register with a register=> in sip.conf beforehand.

This is like calling any FWD extension. First, register, then place
a call with
   DIAL(SIP/fwd-no at fwd.pulver.com)

Any pointer to the manual?

/O