[Asterisk-Users] Dial via sip gateway?

[Mike Machado]

Bob, I have a question into mediatrix for this, but maybe you have
figured it out. I am trying to map a SIP user to a specific PSTN line. I
have my extensions.conf file as you show below, but on the 1204, it just
grabs whatever line is available, whereas I want extension 101 to always
be port1 on 1204, and extension 102 to be port 2 and so on. I noticed a
NetToPstnSourceFilter MIB per port, and their docs hint at using this,
but the example in the docs describes their FXS to FXO, so I am not sure
what I would put in that MIB. CallerID info? * calling sip extension
number? Have you been able to make this work?

On Sat, 2004-01-31 at 20:22, Bob Knight wrote:
> Rich Adamson wrote:
> 
> >I'm having a brain fart....
> >
> >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)
> exten => _9NXXXXXX,2,Congestion
> 
> [trunk-toll]
> exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@mediatrix-1204)
> exten => _91NXXNXXXXXX,2,Congestion
--