Douglas Garstang
2005-Dec-29 12:41 UTC
[Asterisk-Users] Realtime Multiple Asterisk boxes andrtcachefriends MWI
The word from Kevin Fleming and Digium is that the use of realtime to support multiple Asterisk boxes sharing sip is not supported or even known to work at this point. -----Original Message----- From: Asterisk [mailto:Asterisk@isgcom.com] Sent: Thursday, December 29, 2005 12:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Realtime Multiple Asterisk boxes andrtcachefriends MWI I am working an a multiple box asterisk solution. I need phones to be able to login to multiple asterisk servers. I need Phone A to be able to register to switch A and call Phone B that is registered to switch B. With rtcachfriends=no this can be done, However I then loss MWI and "sip show peers" plus if a Phone becomes unreachable the phone I get dead air until the dial timeout reached. With rtcachfriends=yes I get MWI & "Sip show peers", However I cannot call phones that register to a different switch. My current working solution is to have rtcachfriends=yes. Place the call via sip if dialstatus= chanunavaliable I then route the call to the other switch via an IAX trunk. Everything works but I don't have a true load balance soltuion. Plus it really only works for 2 boxes. It get out of hand when I add more.. I have tried using AGI and dialing the "full contact" found in the SIP realtime table. It works if the phone is active, but if the phone is no active I get dead air until the dial timeout is reached. This will not work as I cannot have 12 sec of dead air. So is there a way know the status of a SIP UA? It is it in the SIP realtime data? I looked at regseconds but it does not seem to be it because I can have a UA that is unreachable and the regseconds are not expired. Could realtime be altered to add a status filed to the SIP realtime table? Or is there a asterisk configuration option that I missed? This is my first post so please forgive me if I posted this in the wrong list. Many thanks! Doug Gillespie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051229/7de486eb/attachment.htm
Simone Cittadini
2005-Dec-30 03:20 UTC
[Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Douglas Garstang ha scritto:> The word from Kevin Fleming and Digium is that the use of realtime to > support multiple Asterisk boxes sharing sip is not supported or even > known to work at this point.What about IAX ? If I connect two asterisk servers to a common mysql backend (only iaxusers, no sip or extensions) will it : a) work smoothly, don't waste time optimizing your agi b) definitively will not work, you're doomed c) we don't know, try it and let us know
Kevin P. Fleming
2006-Jan-03 07:16 UTC
[Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Simone Cittadini wrote:> What about IAX ? If I connect two asterisk servers to a common mysql > backend (only iaxusers, no sip or extensions) will it :There is no support for sharing dynamic peer registrations between Asterisk servers via Realtime for SIP or IAX2. Sharing the Realtime database for users and non-dynamic peers works fine, since there is no updating of the database required.
Rich Adamson
2006-Jan-03 07:24 UTC
[Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
> > What about IAX ? If I connect two asterisk servers to a common mysql > > backend (only iaxusers, no sip or extensions) will it : > > There is no support for sharing dynamic peer registrations between > Asterisk servers via Realtime for SIP or IAX2. Sharing the Realtime > database for users and non-dynamic peers works fine, since there is no > updating of the database required.If you take the word "dynamic" out of that, then can he effectively have primary/secondary/backup systems that allows the user to re-register and/or redial his call on a different * server?
Mike Fedyk
2006-Jan-03 10:53 UTC
Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Kevin P. Fleming wrote:> Mike Fedyk wrote: > >> Think of this scenario: You have two * RT servers running heartbeat >> and one goes down. If the SIP registration information was kept in >> the DB tables, the backup server could take over the ethernet and IP >> addresses and continue without forcing the phones to re-register. > > > Yes, that could work just as you described.With the current *RT release?
Andrew Kohlsmith
2006-Jan-12 16:42 UTC
[Asterisk-Users] Why can Asterisk Auto Attendant pick up on first ring?
On Thursday 12 January 2006 17:24, Dakota wrote:> I have an Asterisk installation, however whever someone calls into it, they > hear it ring two times, before the Auto Attendant comes on. > > Is there a way to get the system to answer on the first ring?Turn off caller ID detection. With it on, Asterisk waits for the second ring because in its default configuration, it expects North American caller ID, which appears between the first and second rings. -A.
Tomislav Parcina
2006-Jan-16 07:22 UTC
[Asterisk-Users] Re: Mediatrix windows-based setup?
Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina name.surname@email.t-com.hr
What are you talking about? That is the address I used. -Kerry> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Tomislav Parcina > Sent: Monday, January 16, 2006 6:22 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Re: Mediatrix windows-based setup? > > Please stop replaying to mesage. If you plan to open thread > do so by writing mail to this address asterisk-users@lists.digium.com > > > > > -- > > Tomislav Parcina > name.surname@email.t-com.hr > > _______________________________________________ > --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 >