I added those lines to my configuration, and i just see with ethereal that my client dial and the 1204 led turn on and they started to interchange packets, im newbie with asterisk i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up? could u send me all the configuration i need step by step?
----- Original Message ----- From: "Wojciech Tryc" <[EMAIL PROTECTED]> Date: Mon, 7 Jun 2004 21:59:43 -0400 To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Mediatrix 1204 > The Mediatrix box will not registered with * as the user name and password > for sip are not yet implemented in their firmware. > All what you have to do is to protect the box from the internet (firewall) > and access is like: > exten => _1905XXXXXXX,1,Dial(SIP/[EMAIL PROTECTED]) > exten => _1905XXXXXXX,1,Congestion > > This way you basically have a pool of 4 outgoing lines. You can however > route properly incoming calls. > I hope this will help you, > Regards, > Wojtek > > ----- Original Message ----- > From: "Gonzalo Gasca" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, June 07, 2004 9:45 PM > Subject: [Asterisk-Users] Mediatrix 1204 > > > > Actually im trying to set up a Mediatrix 1204 to place outgoing calls,i > just cand do internal ones, i would like to know if someone could help me > with this issue, i declared in sip.conf line1 to line4 for each 1204 port > > > > SIP.conf > > > > [100] ; My SIP agent > > type=friend ; This device takes and makes calls > > username=100 ; Username on device > > secret=100 ; Password for device > > host=dynamic ; This host is not on the same IP addr > every time > > context=sip ; Inbound calls from this host go here > > mailbox=100 ; Activate the message waiting light if > this voicemailbox has messages in it > > callerid="Gonzalo Gasca" <100> ; Caller ID > > > > [line1] > > type=friend ; This device takes and makes calls > > username=line1 ; Username on device > > host=110.10.200.10 ; This host is not on the same IP addr > every time > > context=sip > > callerid="Line 1" <line1> ; Caller ID > > > > > **************************************************************************** > ************ > > extensions.conf > > > **************************************************************************** > ************ > > > > [sip] > > ignorepat => 9 > > exten => _9NNNNNNNN,1,Dial(SIP/line1) > > exten => :9NNNNNNNN,2,Congestion > > > > But it just put the box in busy and interchange rtp G711 packets with my > client SJphone form sjlabs > > I would like a helping hand! > > -- > > _______________________________________________ > > Get your free email from http://www.hackermail.com > > > > Powered by Outblaze > > _______________________________________________ > > 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 > > > > _______________________________________________ > 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 -- _______________________________________________ Get your free email from http://www.hackermail.com Powered by Outblaze _______________________________________________ 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