Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate tables with config information, and use a view for the actual compiled configuration, instead of how it is now, where there may be repeating info all over the database. Does any of that make sense? -- Aaron Daniel -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 546 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/63dd365a/signature.pgp
On 29 Jun 2006, at 02:08, Aaron Daniel wrote:> Has anyone considered the idea of splitting the sip registration > information in a realtime database from the actual configuration of > the > peers? > > I mean, instead of having a table full of the configuration > information > (i.e. name, regexten, secret, etc) and registration information (i.e. > ipaddr, fullcontact, etc), you have separate tables with their own > information. This way, you can have separate tables with config > information, and use a view for the actual compiled configuration, > instead of how it is now, where there may be repeating info all > over the > database. > > Does any of that make sense?Yes, except, if I understand you correctly, you would also need to write insert and update triggers on the view, so that when asterisk writes to the compiled config, the correct changes are applied to your separate tables. That might limit your choice of databases a bit. The other thing to watch is that you have to ensure that the resulting view behaves exactly the way that asterisk expects it to, unless you get the join right, you can get duplicate (apparently identical) records back which would confuse asterisk. Overall I like the idea, we do this sort of thing lots in the web world, I'll probably try something similar in cdr odbc . By the way, has anyone used cdr_odbc to oracle XE (the free one) yet ? Tim.> > -- > Aaron Daniel > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersTim Panton tim@mexuar.com
> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Wednesday, June 28, 2006 7:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Realtime SIP Registrations > > > Has anyone considered the idea of splitting the sip registration > information in a realtime database from the actual > configuration of the > peers? > > I mean, instead of having a table full of the configuration > information > (i.e. name, regexten, secret, etc) and registration information (i.e. > ipaddr, fullcontact, etc), you have separate tables with their own > information. This way, you can have separate tables with config > information, and use a view for the actual compiled configuration, > instead of how it is now, where there may be repeating info > all over the > database. > > Does any of that make sense?How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug.
> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > 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.
I'm intensely curious why it doesn't currently work. You have multiple Asterisk systems, all referring to a common table for SIP peer information. The fact that there is multiple Asterisk systems accessing the same MySQL data should be completely transparent to each of them, and I don't understand why this doesn't work. Anyone? Doug.> -----Original Message----- > From: David Thomas [mailto:punknow@gmail.com] > Sent: Friday, June 30, 2006 9:40 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Realtime SIP Registrations > > > 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 <Asterisk@isgcom.com> 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: asterisk-users-bounces@lists.digium.com 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 < Asterisk@isgcom.com > <mailto:Asterisk@isgcom.com> > 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/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. > > > > > > > > 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: asterisk-users-bounces@lists.digium.com 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 <dgarstang@oneeighty.com> wrote: > > > > -----Original Message----- > > > > From: Aaron Daniel [mailto: amdtech@shsu.edu > <mailto:amdtech@shsu.edu> ] > > > > 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 > > > > _______________________________________________ > > --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 >