I'm not quite sure what do you mean with MSC. Anyway, I assume your environment is like
[PSTN (Public Switched Telephone Network)]<------------------>[DTM Switch]<-----------SS7 (PRI line)---------------->[Asterisk Box]<----------------VoIP (SIP/IAX etc...)---------> IP net If you mean MSC Mobile Switching Center it could look like [GSM Network]<------------------------->[MSC]<----------------- SS7 -------------------->[Asterisk Box]<---------------------VoIP---------------------->IP net Normally, the DTM Switch or MSC should be configurable for load-balancing and failover. Point code is for SS7 networking like IP address for IP networking. huu giang schrieb: > Do you mean that SS7 switch is a MSC and do all MSC support load > balancing without any hardware between it and my Server. > > Sorry for my English, what do you mean two point codes for my servers > ?. I have at least two servers. > > > --- On *Wed, 3/31/10, Tobias Wolf /<tobias.w...@evision.de>/* wrote: > > > From: Tobias Wolf <tobias.w...@evision.de> > Subject: Re: [asterisk-users] Asterisk load balancing and failover > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Date: Wednesday, March 31, 2010, 4:27 AM > > huu giang schrieb: > > Hi Zeeshan > > > > I know a solution using DRBD, Heartbeat and RedFone hardware to > > provide failover ability to Asterisk. > > > > If I have two Asterisk Servers, and each server has a TDM card > and a > > PRI line connect to each card, how your solution can provide > failover > > ability to Asterisk ? Do you need any other hardware? > > > > The calles to my IVR System don't just come from IP network > (SIP) but > > can come from SS7 network. > > > Well, if that case the SS7 Switch to which you are connected > should be > able to load balance the call to both of your servers. I guess you > have > two point codes for you servers? If one server goes down, the ss7 > switch > received the red alarms and > stops to route calls to it. Once the server is up again it will > get new > calls. > > So, we only thing you have to worry about is to keep state > information > between the two servers consistent if people record messages or > access > databases. > > Regards, > > Tobias > > > > Thanks. > > > > > > > > > > --- On *Fri, 3/26/10, Zeeshan Zakaria /<zisha...@gmail.com > </mc/compose?to=zisha...@gmail.com>>/* wrote: > > > > > > From: Zeeshan Zakaria <zisha...@gmail.com > </mc/compose?to=zisha...@gmail.com>> > > Subject: Re: [asterisk-users] Asterisk load balancing and > failover > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <asterisk-users@lists.digium.com > </mc/compose?to=asterisk-us...@lists.digium.com>> > > Date: Friday, March 26, 2010, 1:51 AM > > > > About two years ago I setup two high availability solutions > using > > DRBD and Heartbeat. The worked great and shutting down or > > unplugging one server stayed transparent for the callers, as > IVRs > > stayed available. Having said this, it was not very straight > > forward to set it up, but not very difficut either. So Heartbeat > > and DRBD can be a good starting point for you. > > > > -- > > Zeeshan A Zakaria > > > >> On 2010-03-26 4:40 AM, "huu giang" <huugiang...@yahoo.com > </mc/compose?to=huugiang...@yahoo.com> > >> </mc/compose?to=huugiang...@yahoo.com > </mc/compose?to=huugiang...@yahoo.com>>> wrote: > >> > >> Hi List, > >> > >> I'm finding a solution to provide failover and load balancing > >> features to my IVR system. > >> > >> Anyone suggest me what is the best solution please?. what the > >> hardware I should use ?. > >> > >> I heard about RedFone, but someone on the mail list said > that it > >> is not good because *TDMoE* module in asterisk is not so > *stable* > >> and TDMoE is stale. And It seems that RedFone doesn't not > support > >> load balancing ability (I can't find any document about this > >> feature). > >> > >> Best Regards, > >> Giang Huu. > >> > >> > >> > >> > >> -- > >> > _____________________________________________________________________ > >> -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > >> New to Asterisk? Join us for a live introductory webinar > every Thurs: > >> http://www.asterisk.org/hello > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -----Inline Attachment Follows----- > > > > -- > > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar > every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users