[Asterisk-Users] Dial via sip gateway?
Rich Adamson
radamson at routers.com
Sun Feb 1 06:00:26 MST 2004
Mike,
I'm hoping one can specify a particular mediatrix "port" in the Dial Sip
command, but haven't found any Dial syntax that would allow passing a
userid/password to the gateway. Since the 1204 provides a AuthUsrPwd on
a per port basis, my guess would be that we either have to pass the Alias
defined for that port or the AuthUsrPwd in the Dial command.
When I attempt
exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:1}@205.22.93.1)
I get an immediate "407 Proxy Authentication Required" back. However, with
a packet sniffer running, * isn't even sending a packet to the mediatrix.
I'd have to guess and assume * is doing this because the mediatrix isn't
'registered' with *, but the mediatrix was not designed to register anyway.
I'm stuck in the Dial syntax, and can't seem to find any google reference
as to how to pass the needed parameters.
Rich
------------------------
> Bob, I have a question into mediatrix for this, but maybe you have
> figured it out. I am trying to map a SIP user to a specific PSTN line. I
> have my extensions.conf file as you show below, but on the 1204, it just
> grabs whatever line is available, whereas I want extension 101 to always
> be port1 on 1204, and extension 102 to be port 2 and so on. I noticed a
> NetToPstnSourceFilter MIB per port, and their docs hint at using this,
> but the example in the docs describes their FXS to FXO, so I am not sure
> what I would put in that MIB. CallerID info? * calling sip extension
> number? Have you been able to make this work?
>
> On Sat, 2004-01-31 at 20:22, Bob Knight wrote:
> > Rich Adamson wrote:
> >
> > >I'm having a brain fart....
> > >
> > >What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
> > >
> > >Been trying stuff similar to:
> > > exten => _6X.,1,Dial(SIP/3091 at 205.22.93.1/${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
> --
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
---------------End of Original Message-----------------
More information about the asterisk-users
mailing list