Well, to dial a peer direclty the only thing that is missing in realtime is the status of the sip peer. (registered, Unregistered, unknown, reachable). If you dial a peer via ip and it is unavaliable you get dead air. So you need to know the status of the peer before dialing it. The change basicly updates realtime with the peers status. I did the same thing for IAX as well.. Doug
________________________________ From: [EMAIL PROTECTED] on behalf of Mike Lynchfield Sent: Thu 6/29/2006 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations can you elaborate on modify sip to update the "status" on the sip friends in realtime thanks On 6/29/06, Doug G < [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > wrote: What I did was modify sip to update the "status" on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial ( SIP/[EMAIL PROTECTED]:5060) Of course you need to check the "status" in realtime data before you dial. This allows MANY Asterisk servers to share the same SIP data. I then load balance with DNS SRV.. Yes I have tested in failover it works. I too have been told that by many that this will not work. So I keep expecting to hit some problem with it, but to date I have not... Doug ________________________________ From: [EMAIL PROTECTED] on behalf of David Thomas Sent: Thu 6/29/2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations I think lots of us know about it... We're just not sure how to go about fixing it. :-( I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying "works for me" have never actually tested their system in failure scenarios, or they are working in a controlled environment without NAT and such... regards, David On 6/29/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > -----Original Message----- > > From: Aaron Daniel [mailto: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ] > > Sent: Thursday, June 29, 2006 9:27 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] Realtime SIP Registrations > > > > > > On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: > > > How about fixing realtime SIP so that multiple Asterisk > > boxes can reference the same database? > > > > > > Doug. > > > > That's kinda what I'm hoping to work towards :) > > I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.....) > > Doug. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.theclubvoip.com Making it happen 1.888.470.7253
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