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/[EMAIL PROTECTED]/${EXTEN-1}) > >where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did > >even try the IP. > > > >Rich > > > from my extensions.conf: > > ;****************************************************** > [trunk-local] > ;****************************************************** > exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) > exten => _9NXXXXXX,2,Congestion > > [trunk-toll] > exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) > exten => _91NXXNXXXXXX,2,Congestion -- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users