Doug,

If you'd be willing to share the patch and AGI, I would be happy to
help test your solution. I know that myself and several others have
been looking for a way to make Asterisk do this for quite some time.

regards,
David

On 6/29/06, Doug G <[EMAIL PROTECTED]> wrote:
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.
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--
Mike
Sales Manager
http://www.theclubvoip.com
Making it happen
1.888.470.7253

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