can you elaborate on modify sip to update the "status" on the sip friends in realtime<br>thanks<br><br><div><span class="gmail_quote">On 6/29/06, <b class="gmail_sendername">Doug G</b> <<a href="mailto:Asterisk@isgcom.com">
Asterisk@isgcom.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">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/1112223333@10.10.10.1:5060) 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.
<br><br><br><br>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...<br><br><br><br>Doug<br><br><br><br><br><br>________________________________
<br><br>From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> on behalf of David Thomas<br>Sent: Thu 6/29/2006 1:05 PM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion
<br>Subject: Re: [Asterisk-Users] Realtime SIP Registrations<br><br><br><br>I think lots of us know about it... We're just not sure how to go<br>about fixing it. :-(<br>I know it's been a thorn in my side since I started using Asterisk.
<br><br>I would suspect that many of those saying "works for me" have never<br>actually tested their system in failure scenarios, or they are working<br>in a controlled environment without NAT and such...<br><br>
regards,<br>David<br><br>On 6/29/06, Douglas Garstang <<a href="mailto:dgarstang@oneeighty.com">dgarstang@oneeighty.com</a>> wrote:<br>> > -----Original Message-----<br>> > From: Aaron Daniel [mailto:<a href="mailto:amdtech@shsu.edu">
amdtech@shsu.edu</a>]<br>> > Sent: Thursday, June 29, 2006 9:27 AM<br>> > To: Asterisk Users Mailing List - Non-Commercial Discussion<br>> > Subject: RE: [Asterisk-Users] Realtime SIP Registrations<br>> >
<br>> ><br>> > On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote:<br>> > > How about fixing realtime SIP so that multiple Asterisk<br>> > boxes can reference the same database?<br>> > >
<br>> > > Doug.<br>> ><br>> > That's kinda what I'm hoping to work towards :)<br>><br>> 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.....)
<br>><br>> Doug.<br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>><br>> Asterisk-Users mailing list<br>
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