That's the extension incoming calls will ring to -- If you use the example below, and sip calls come into the incoming context, it would go to 99612 instead of "s" extension. This is great if you have multiple DIDs and want to handle them differently.
> -----Original Message----- > From: Nitesh Divecha [mailto:[EMAIL PROTECTED] > Sent: Friday, February 18, 2005 4:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] VONAGE <----> ASTERISK SIP > TERMINATION????? > > > Thanks Jay, > > For the Vonage information on how to make it work! > > Just a quick question, what is the last number (99612) you > specified in the register string and beginning of sip parameter. > > register => 16125551212:[EMAIL PROTECTED]:5061/99612 > > [sip99612] > > Thanks, > > Nitesh > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk > Sent: Friday, February 18, 2005 11:56 AM > To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - > Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] VONAGE <----> ASTERISK SIP > TERMINATION????? > > Yes, it's doable, had this running for several months here. > However, you'll need to get a softphone for $10/month from > them, and they'll provide the sip-credentials on their > website. It's a lousy solution if you really just want one > number, because then you'll have to pay $15/month for their > basic "hardline" service, plus an extra $10 for the softline. > It may make sense if you need several numbers in rate > centers where the usual suspects don't have numbers -- as it > did for me. Once a less expensive provider popped up, I > traded the $25 Vonage mess for a $5 unlimited DID. > http://lists.digium.com/pipermail/asterisk-users/2004-June/052678.html This still works, just add "insecure=very" -----Original Message----- From: Lucas Wrenn [mailto:[EMAIL PROTECTED] Sent: Friday, February 18, 2005 9:25 AM To: [email protected] Subject: [Asterisk-Users] VONAGE <----> ASTERISK SIP TERMINATION????? Has anyone out there successfully set up their * box to terminate their VONAGE calls? I (and I am sure lots of others) would love to hear how you did it. I'd like to be able to get rid of the extra hardware I have hanging around here and use the ASTERISK machine to handle the SIP termination instead of needing to have a Linksys modem (w/phone) and an additional X100P card. Thanks. Wishing for a solution. ([EMAIL PROTECTED]) _______________________________________________ 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 _______________________________________________ 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
