Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. Any comments would be great! Thanks Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060310/24acd8df/attachment.htm
If you implement multiple Asterisk systems, your challenge is going to be in ensuring that phones registered to one Asterisk know how to reach phones registered to another Asterisk system. Good luck with that! Doug. -----Original Message----- From: Ron McCarthy [mailto:ronmccar@gmail.com] Sent: Friday, March 10, 2006 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Clustering Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. Any comments would be great! Thanks Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060310/82c5b268/attachment.htm
I think there's a difference between sharing extension info and sharing registration info. Just because an extension exists on a given Asterisk box, doesn't mean that the user is available (ie registered and known to Asterisk) at that IP address/port. -----Original Message----- From: Matthew Crocker [mailto:matthew@crocker.com] Sent: Friday, March 10, 2006 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering Would an internal DUNDi configuration help the asterisk servers share their extension info? Or, use e.164 with an internal DNS zone to lookup the routing information. SIP phone logs into Asterisk 'A' and a script runs to update the e.164 DNS info pointing the DID to Asterisk 'A' -Matt On Mar 10, 2006, at 2:42 PM, Douglas Garstang wrote:> If you implement multiple Asterisk systems, your challenge is going > to be in ensuring that phones registered to one Asterisk know how > to reach phones registered to another Asterisk system. Good luck > with that! > > Doug. > -----Original Message----- > From: Ron McCarthy [mailto:ronmccar@gmail.com] > Sent: Friday, March 10, 2006 12:22 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Clustering > > Hello All, > > Ive been doing more and more research on trying to setup a cluster/ > load balancer for Asterisk. All the Asterisk boxes would be using a > config that is the same between them all (via a DB), but we want > one location to point the phones to, and from there that machine/ > device will send it to a Asterisk server so the call can be > processed. I know you cant balance the whole call, ie: once the > call is started the RTP stream has to go to the same server, but a > new call could go to a different server if perhaps the 1st server > was unreachable. > > Has anyone tried this, or got this to work? Ive been looking at > using a Juniper Session Border Controller, but not sure if thats > gonna do the trick, and then we also have SER.. > > Any comments would be great! > > Thanks > Ron > _______________________________________________ > --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-- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com _______________________________________________ --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
Kevin, I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. Doug. -----Original Message----- From: Kevin P. Fleming [mailto:kpfleming@digium.com] Sent: Friday, March 10, 2006 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering Douglas Garstang wrote:> I think there's a difference between sharing extension info and sharing registration info. Just because an extension exists on a given Asterisk box, doesn't mean that the user is available (ie registered and known to Asterisk) at that IP address/port.It does if you use regcontext/regexten to register dialplan extensions dynamically. _______________________________________________ --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
Kevin, From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: "If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us" What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? Doug. -----Original Message----- From: Kevin P. Fleming [mailto:kpfleming@digium.com] Sent: Fri 3/10/2006 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering Douglas Garstang wrote: > I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. It's not broken. If you find some official documentation that says so, then it needs to be fixed. If you read it somewhere else, then that source is not something you should trust. regexten in sip.conf works just fine; it can easily be used to make an extension 'appear' and 'disappear' from the desired context based on the status of the peer's registration. If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. _______________________________________________ --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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 5782 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060310/d40468eb/attachment.bin
------------------------------ Message: 6 Date: Fri, 10 Mar 2006 12:22:12 -0700 From: "Ron McCarthy" <ronmccar@gmail.com> Subject: [Asterisk-Users] Clustering To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <3885f4fe0603101122m6742410ep25276b736072618c@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. Any comments would be great! Thanks Ron -------------- next part -------------- Ron, I'm doing something similar with clustering. I haven't gotten the total design down yet but so far I have 1 * server holding all the routes to several child/registration * servers where my iax and sip clients actually register to. I'm using the 1 * server running dundi to peer with all the registration servers and all the registration servers only peer with the 1 * server, I call this the Registration Presence Server or RPS. When a child/registration server does a lookup request to the RPS, the RPS does a lookup on all the other registration servers and knows who is registered where and relays that info back to the original requesting registration server. Use a dundi ttl=1 in the RPS and ttl=2 in each registration server to avoid routing loops. I'm still in the testing phase but it's going well, but I?m running into some cache timeout issues when a client drops off and re-registers to a different registration server, have to flush dundi to pickup the new location. There is a cache timeout parameter I have yet to play with. I don?t have the load balancing session border controller function down yet, but that is on the list of things to do. Hope this helps. JR JR Richardson Engineering for the Masses
If all the sub-servers register themselves to the frontend load balancer and support reinvite, the load balancer can decide which server to send the call to based on the CPU utilizations of the call processing servers. I'm assuming all calls are voip calls here. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Ron McCarthy Sent: Friday, March 10, 2006 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Clustering Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. Any comments would be great! Thanks Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060310/21a0fbcd/attachment.htm
We're doing this. Our Polycom phones point to a domain name that support SRV records which gives us a roughly even distribution of calls. We have OpenSER systems sitting in front of the phones. Each OpenSER system is configured with different primary/secondary/tertiary Asterisk boxes. When a phone registers with SER, it 'copies' the registration down to all the Asterisk systems. However, now that I find we allegedly could have used regexten on Asterisk to replicate the registrations (yet to see docs on how this works), that $8k we spent on systems for OpenSER suddenly seems like money not quite so well spent. Calls to the PSTN are routed from Asterisk back to the OpenSER proxies where it sends it to the PSTN gateway. Eventhough it all seems to work quite well, and using OSPF we have been able to actually fail an interface on a single OpenSER or Asterisk box and fail over an RTP stream (only a few seconds of dead air), due to the horrible Asterisk documentation, our main challenge has been in replicating phone registrations between the Asterisk systems. It would have been great if the Asterisk product was mature enough to support Realtime SIP for storing registrations from multiple Asterisk boxes. On the surface you'd think it's possible, but every one has a different opinion about whether it's technically shown to work. If your going to try and set up a HA Asterisk solution be prepared for a really tough time. Doug. -----Original Message----- From: Wai Wu [mailto:wwu@Calltrol.com] Sent: Fri 3/10/2006 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Clustering If all the sub-servers register themselves to the frontend load balancer and support reinvite, the load balancer can decide which server to send the call to based on the CPU utilizations of the call processing servers. I'm assuming all calls are voip calls here. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Ron McCarthy Sent: Friday, March 10, 2006 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Clustering Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. Any comments would be great! Thanks Ron
Hi JR. I'm dying to know... where'd you find your DUNDi documentation? Has something new appeared since I looked at it 2-3 months ago? The O'Reilly book's DUNDi section was impossible to follow, and the examples in the Asterisk DUNDi config files are no better. You do a search online and get almost no results (still wondering when Digium is going to realease some docs for what they call their protocol). I spent a few weeks working on it, tearing my hair out, and gave up. So did my boss. Doug. -----Original Message----- From: JR Richardson [mailto:jr.richardson@cox.net] Sent: Fri 3/10/2006 8:55 PM To: ronmccar@gmail.com; asterisk-users@lists.digium.com Cc: jr.richardson@cox.net Subject: re: [Asterisk-Users] Clustering ------------------------------ Message: 6 Date: Fri, 10 Mar 2006 12:22:12 -0700 From: "Ron McCarthy" <ronmccar@gmail.com> Subject: [Asterisk-Users] Clustering To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <3885f4fe0603101122m6742410ep25276b736072618c@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. Any comments would be great! Thanks Ron -------------- next part -------------- Ron, I'm doing something similar with clustering. I haven't gotten the total design down yet but so far I have 1 * server holding all the routes to several child/registration * servers where my iax and sip clients actually register to. I'm using the 1 * server running dundi to peer with all the registration servers and all the registration servers only peer with the 1 * server, I call this the Registration Presence Server or RPS. When a child/registration server does a lookup request to the RPS, the RPS does a lookup on all the other registration servers and knows who is registered where and relays that info back to the original requesting registration server. Use a dundi ttl=1 in the RPS and ttl=2 in each registration server to avoid routing loops. I'm still in the testing phase but it's going well, but I?m running into some cache timeout issues when a client drops off and re-registers to a different registration server, have to flush dundi to pickup the new location. There is a cache timeout parameter I have yet to play with. I don?t have the load balancing session border controller function down yet, but that is on the list of things to do. Hope this helps. JR JR Richardson Engineering for the Masses _______________________________________________ --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
Gabriel. We are using OSPF on our asterisk box. When an interface fails, OSPF switches the default route over to the other interface. :) Fortunately Polycom phones are smart enough to wait for the RTP stream to be re-established. As for OpenSER vs SER... I'm not sure. It really shouldn't make much difference which is used. -----Original Message----- From: Gabriel Afana [mailto:asterisk@gafana.com] Sent: Sat 3/11/2006 6:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering Doug, How did get the RTP stream to fail over in progress? Also, you mentioned your using OpenSER. Why did you choose this over the standard SER? - Gabe ----- Original Message ----- From: "Douglas Garstang" <dgarstang@oneeighty.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>; "Asterisk Users Mailing List -Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Friday, March 10, 2006 10:49 PM Subject: RE: [Asterisk-Users] Clustering > We're doing this. Our Polycom phones point to a domain name that support SRV records which gives us a roughly even distribution of calls. We have OpenSER systems sitting in front of the phones. Each OpenSER system is configured with different primary/secondary/tertiary Asterisk boxes. When a phone registers with SER, it 'copies' the registration down to all the Asterisk systems. > > However, now that I find we allegedly could have used regexten on Asterisk to replicate the registrations (yet to see docs on how this works), that $8k we spent on systems for OpenSER suddenly seems like money not quite so well spent. > > Calls to the PSTN are routed from Asterisk back to the OpenSER proxies where it sends it to the PSTN gateway. > > Eventhough it all seems to work quite well, and using OSPF we have been able to actually fail an interface on a single OpenSER or Asterisk box and fail over an RTP stream (only a few seconds of dead air), due to the horrible Asterisk documentation, our main challenge has been in replicating phone registrations between the Asterisk systems. > > It would have been great if the Asterisk product was mature enough to support Realtime SIP for storing registrations from multiple Asterisk boxes. On the surface you'd think it's possible, but every one has a different opinion about whether it's technically shown to work. > > If your going to try and set up a HA Asterisk solution be prepared for a really tough time. > > Doug. > > > > > > > -----Original Message----- > From: Wai Wu [mailto:wwu@Calltrol.com] > Sent: Fri 3/10/2006 9:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: RE: [Asterisk-Users] Clustering > > > If all the sub-servers register themselves to the frontend load balancer and support reinvite, the load balancer can decide which server to send the call to based on the CPU utilizations of the call processing servers. I'm assuming all calls are voip calls here. > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Ron McCarthy > Sent: Friday, March 10, 2006 2:22 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Clustering > > > Hello All, > > Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. > > Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. > > Any comments would be great! > > Thanks > Ron > > > ---------------------------------------------------------------------------- ---- > _______________________________________________ > --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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 7946 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060311/107de6ca/attachment.bin
So you are actually able to maintain a call in progress even if the server its connected to fails (by routing to another)? - Gabe ----- Original Message ----- From: "David Coulson" <david@davidcoulson.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, March 11, 2006 7:15 PM Subject: Re: [Asterisk-Users] Clustering> > > From what I can find online, OSPF seems to be a technology ormethod,> > not necessarily a program. What are you using to perform OSPF? > > OSPF is a routing protocol. Quagga (quagga.net) is a good open source > implementation of OSPF for Unix. > > David > _______________________________________________ > --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
Think you need to be careful reading his previous post. OSPF can be used to fail over to another "interface" on the same box, but it is _not_ going to fail over to a second box and maintain rtp sessions. Gabriel Afana wrote:> So you are actually able to maintain a call in progress even if the server > its connected to fails (by routing to another)? > > - Gabe > > ----- Original Message -----> >>> From what I can find online, OSPF seems to be a technology or > method, >>> not necessarily a program. What are you using to perform OSPF? >> OSPF is a routing protocol. Quagga (quagga.net) is a good open source >> implementation of OSPF for Unix. >> >> David
That's what we're using. :) -----Original Message----- From: David Coulson [mailto:david@davidcoulson.net] Sent: Sat 3/11/2006 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering > From what I can find online, OSPF seems to be a technology or method, > not necessarily a program. What are you using to perform OSPF? OSPF is a routing protocol. Quagga (quagga.net) is a good open source implementation of OSPF for Unix. David _______________________________________________ --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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4110 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060311/ac39acdb/attachment.bin
No, only if a network interface in the server fails. We have two network interfaces per system (actually we have four, but two are on a private network with a MySQL server). If one of the network interfaces fails, OSPF will switch the default route over to the other interface pretty quick smart. There's probably a little luck involved here too. -----Original Message----- From: Gabriel Afana [mailto:asterisk@gafana.com] Sent: Sat 3/11/2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering So you are actually able to maintain a call in progress even if the server its connected to fails (by routing to another)? - Gabe ----- Original Message ----- From: "David Coulson" <david@davidcoulson.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, March 11, 2006 7:15 PM Subject: Re: [Asterisk-Users] Clustering > > > From what I can find online, OSPF seems to be a technology or method, > > not necessarily a program. What are you using to perform OSPF? > > OSPF is a routing protocol. Quagga (quagga.net) is a good open source > implementation of OSPF for Unix. > > David > _______________________________________________ > --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
How does OSPF tell the remote end (assuming he does not know your setup) start sending RTP packets to the other interface? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Douglas Garstang Sent: Sunday, March 12, 2006 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Clustering No, only if a network interface in the server fails. We have two network interfaces per system (actually we have four, but two are on a private network with a MySQL server). If one of the network interfaces fails, OSPF will switch the default route over to the other interface pretty quick smart. There's probably a little luck involved here too. -----Original Message----- From: Gabriel Afana [mailto:asterisk@gafana.com] Sent: Sat 3/11/2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering So you are actually able to maintain a call in progress even if the server its connected to fails (by routing to another)? - Gabe ----- Original Message ----- From: "David Coulson" <david@davidcoulson.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, March 11, 2006 7:15 PM Subject: Re: [Asterisk-Users] Clustering > > > From what I can find online, OSPF seems to be a technology or method, > > not necessarily a program. What are you using to perform OSPF? > > OSPF is a routing protocol. Quagga (quagga.net) is a good open source > implementation of OSPF for Unix. > > David > _______________________________________________ > --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
It doesn't. It's transparent to the user agent. -----Original Message----- From: Wai Wu [mailto:wwu@Calltrol.com] Sent: Sunday, March 12, 2006 9:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Clustering How does OSPF tell the remote end (assuming he does not know your setup) start sending RTP packets to the other interface? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Douglas Garstang Sent: Sunday, March 12, 2006 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Clustering No, only if a network interface in the server fails. We have two network interfaces per system (actually we have four, but two are on a private network with a MySQL server). If one of the network interfaces fails, OSPF will switch the default route over to the other interface pretty quick smart. There's probably a little luck involved here too. -----Original Message----- From: Gabriel Afana [mailto:asterisk@gafana.com] Sent: Sat 3/11/2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering So you are actually able to maintain a call in progress even if the server its connected to fails (by routing to another)? - Gabe ----- Original Message ----- From: "David Coulson" <david@davidcoulson.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, March 11, 2006 7:15 PM Subject: Re: [Asterisk-Users] Clustering > > > From what I can find online, OSPF seems to be a technology or method, > > not necessarily a program. What are you using to perform OSPF? > > OSPF is a routing protocol. Quagga (quagga.net) is a good open source > implementation of OSPF for Unix. > > David > _______________________________________________ > --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
Hi Gabe, Well I was going to use the SBC to have all phone point to the SBC, and then the SBC takes care of what servers it needs to register with, and then keep a state of what server the RTP stream and the phone need to connect to. Basically like a load balancer would. This is what I understood from Juniper's site. Have you seen anything on this? Thanks! Ron On 3/11/06, Gabriel Afana <asterisk@gafana.com> wrote:> > Hi Ron, > I've been following your thread. I noticed you mentioned about a > Juniper Session Border Controller. I checked online and read about it, but > was unsure exactly how it could intergrate with Asterisk. How would you > have planned to use that device? I am interested because one of my upstream > providers mentioned I should be using an SBC. > > - Gabe > > > ----- Original Message ----- > *From:* Ron McCarthy <ronmccar@gmail.com> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com> > *Sent:* Friday, March 10, 2006 11:22 AM > *Subject:* [Asterisk-Users] Clustering > > Hello All, > > Ive been doing more and more research on trying to setup a cluster/load > balancer for Asterisk. All the Asterisk boxes would be using a config that > is the same between them all (via a DB), but we want one location to point > the phones to, and from there that machine/device will send it to a Asterisk > server so the call can be processed. I know you cant balance the whole call, > ie: once the call is started the RTP stream has to go to the same server, > but a new call could go to a different server if perhaps the 1st server was > unreachable. > > Has anyone tried this, or got this to work? Ive been looking at using a > Juniper Session Border Controller, but not sure if thats gonna do the trick, > and then we also have SER.. > > Any comments would be great! > > Thanks > Ron > > ------------------------------ > > _______________________________________________ > --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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060312/9769c55f/attachment.htm
Uhm, No.
We have multiple Asterisk boxes. OSPF only fails over between interfaces in a
single Asterisk system.
We're not using regexten (cuz there's no frikkin docs for it!!!).
We're using OpenSER's send() command to forward registrations from a
phone to all Asterisk systems.
-----Original Message-----
From: Ron McCarthy [mailto:ronmccar@gmail.com]
Sent: Sunday, March 12, 2006 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Clustering
Regarding OSPF, so your saying you have multiple * boxes setup with same exact
config and then just have OSPF fail everthing over to the new server if it cant
get to it? That makes sense, just never of even thought of doing it that way.
Heck, if you want to get real complex just run BGP and you could then setup
priorties for each server and all kinds of cool stuff.
Are you then using regexten on all servers so when a * tries to make a call it
can find where to go, or are you using something else?
Thanks!
Ron
On 3/12/06, Douglas Garstang < dgarstang@oneeighty.com> wrote:
It doesn't. It's transparent to the user agent.
-----Original Message-----
From: Wai Wu [mailto: wwu@Calltrol.com]
Sent: Sunday, March 12, 2006 9:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Clustering
How does OSPF tell the remote end (assuming he does not know your setup) start
sending RTP packets to the other interface?
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto: asterisk-users-bounces@lists.digium.com]On Behalf Of Douglas
Garstang
Sent: Sunday, March 12, 2006 1:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Clustering
No, only if a network interface in the server fails. We have two network
interfaces per system (actually we have four, but two are on a private network
with a MySQL server). If one of the network interfaces fails, OSPF will switch
the default route over to the other interface pretty quick smart. There's
probably a little luck involved here too.
-----Original Message-----
From: Gabriel Afana [mailto: asterisk@gafana.com]
Sent: Sat 3/11/2006 10:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Clustering
So you are actually able to maintain a call in progress even if the
server
its connected to fails (by routing to another)?
- Gabe
----- Original Message -----
From: "David Coulson" < david@davidcoulson.net
<mailto:david@davidcoulson.net> >
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
< asterisk-users@lists.digium.com>
Sent: Saturday, March 11, 2006 7:15 PM
Subject: Re: [Asterisk-Users] Clustering
>
> > From what I can find online, OSPF seems to be a technology
or
method,
> > not necessarily a program. What are you using to perform
OSPF?
>
> OSPF is a routing protocol. Quagga ( quagga.net) is a good open
source
> implementation of OSPF for Unix.
>
> David
> _______________________________________________
> --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
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Hi Gabe, Well im guessing your ServerIron wont work because its not near smart enough to know how SIP works, let alone, 99.9% of load balancers I have seen use private IP's on server side, the and Load Balncer then has the public IP's assigned to them. Right there, this creates a problem in itself. But im assiumg the Foundry isnt smart enough to keep track of multiple phones from the same IP, and all the RTP sessions associated with it, since like you said, several hundred port numbers are being used. The Juniper box seems to rewrite the actual SIP header on the outbound transversal to the Internet, this solving the NAT return path problem, and then it keeps track in a state table as to what ports go to what server, etc, etc. But I think there is no way this could "failvoer" in the middle of the car, since it would somehow have to change the RTP stream to another port, but also the phone would have to get to get registered on that server as well, which its not, which is why Douglas is using SER to have it register on several different machines, so when the failover occurs the phone is registered and the RTP stream just needs to pick up. Im trying to see exactly how he is doing this, since this is the exact thing I need, and then Ill just run OSPF on my core router (not sure if that will work yet). I woudl perfer to do this all in hardware vs software since a Cisco/Juniper box is musch less prone to failure then a server with software, but I guess more research will tell what ill be using in the end :) Once I get this going, I want to post a entire howto on the wiki. Thanks! Ron On 3/12/06, Gabriel Afana <asterisk@gafana.com> wrote:> > Hi Ron, > If the SBC would have served mearly as a load balancer...I already > have one and it didn't work too well. I have a Foundry ServerIron XL load > balancer and I've tried using it with Asterisk. It has had positive and > negative results. > > Positive: It *would* load balance between asterisk servers for whatever > port I set (I was using 5060 for SIP). However, I didn't mess with the RTP > because its got so many ports (and you can't add ranges for virtual server > ports, you have to enter exact ports - at least I think) and because I have > no idea how that would work if SIP signaling goes to one server and RTP goes > to another??? (probably not!) I would create a virtual IP on the load > balancers and have all the phones register to this IP. When checking status > of the ports on each server, it showed 5060 for all servers was unused (0 > current connections). When I would make a call, it would show the 5060 port > on one of the * servers in use (1 current connections) and it worked > fine....this is where the problem started. > > Negative: When I would hang up the phone, it would still show 1 current > connection to that server's 5060 port. Every call I would make from then on > would *still* go to that same server. It seems the ports are "sticky" or > set with a keepalive. Of course I can define these options on the > ServerIron, but even with "sticky" disabled and keepalive disabled, the port > would appear active (like keepalive was enabled) and every call would go to > the same server (like "sticky" was enabled). Even if I would shutdown > asterisk on that server, it would still show an active user on that port and > when I would make the call, the call would not go through. The LB was not > failing the port. I think maybe if I keep playing with it...? Any > suggestions? > > If I can get my ServerIron working, I will do a complete write up on > it...but it works only partially. > > This is why I was so interested in the Juniver SBC....if it would be able > to act a proxy, do all the load balancing and instantly failover if a server > fails; basically a VoIP Load Balancer. But I guess thats not what it does. > Does a VoIP load balancer hardware exist or is the only solution right now > software proxies like SER? > > - Gabe > > > > ----- Original Message ----- > *From:* Ron McCarthy <ronmccar@gmail.com> > *To:* Gabriel Afana <asterisk@gafana.com> ; Asterisk Users Mailing List > -Non-Commercial Discussion <asterisk-users@lists.digium.com> > *Sent:* Sunday, March 12, 2006 1:16 PM > *Subject:* Re: [Asterisk-Users] Clustering > > Hi Gabe, > Well I was going to use the SBC to have all phone point to the SBC, and > then the SBC takes care of what servers it needs to register with, and then > keep a state of what server the RTP stream and the phone need to connect to. > Basically like a load balancer would. This is what I understood from > Juniper's site. Have you seen anything on this? > > Thanks! > Ron > > On 3/11/06, Gabriel Afana <asterisk@gafana.com> wrote: > > > > Hi Ron, > > I've been following your thread. I noticed you mentioned about a > > Juniper Session Border Controller. I checked online and read about it, but > > was unsure exactly how it could intergrate with Asterisk. How would you > > have planned to use that device? I am interested because one of my upstream > > providers mentioned I should be using an SBC. > > > > - Gabe > > > > > > ----- Original Message ----- > > *From:* Ron McCarthy <ronmccar@gmail.com> > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com> > > *Sent:* Friday, March 10, 2006 11:22 AM > > *Subject:* [Asterisk-Users] Clustering > > > > Hello All, > > > > Ive been doing more and more research on trying to setup a cluster/load > > balancer for Asterisk. All the Asterisk boxes would be using a config that > > is the same between them all (via a DB), but we want one location to point > > the phones to, and from there that machine/device will send it to a Asterisk > > server so the call can be processed. I know you cant balance the whole call, > > ie: once the call is started the RTP stream has to go to the same server, > > but a new call could go to a different server if perhaps the 1st server was > > unreachable. > > > > Has anyone tried this, or got this to work? Ive been looking at using a > > Juniper Session Border Controller, but not sure if thats gonna do the trick, > > and then we also have SER.. > > > > Any comments would be great! > > > > Thanks > > Ron > > > > ------------------------------ > > > > _______________________________________________ > > --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 > > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060312/f21bf9ef/attachment.htm
Ron,
Think the discussion has drifted a bit. Looking back at your original post. What
you wanted was a simple load lalancer to distribute the calls from registered
sip phones across multiple servers. I think you can accomblish this with a
script in the entry extension (on the master server) that pulls for CPU
utilization of the other servers and send the call to the one that's least
utilized. As for RTP packets. I thanks the 'canrevite' scheme in * can
handle it automatically, i.e. RTP packets will bypass the master server and
directly to the call processor server.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Ron McCarthy
Sent: Sunday, March 12, 2006 5:07 PM
To: Gabriel Afana; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Clustering
Hi Gabe,
Well im guessing your ServerIron wont work because its not near smart enough to
know how SIP works, let alone, 99.9% of load balancers I have seen use private
IP's on server side, the and Load Balncer then has the public IP's
assigned to them. Right there, this creates a problem in itself. But im assiumg
the Foundry isnt smart enough to keep track of multiple phones from the same IP,
and all the RTP sessions associated with it, since like you said, several
hundred port numbers are being used. The Juniper box seems to rewrite the actual
SIP header on the outbound transversal to the Internet, this solving the NAT
return path problem, and then it keeps track in a state table as to what ports
go to what server, etc, etc. But I think there is no way this could
"failvoer" in the middle of the car, since it would somehow have to
change the RTP stream to another port, but also the phone would have to get to
get registered on that server as well, which its not, which is why Douglas is
using SER to have it register on several different machines, so when the
failover occurs the phone is registered and the RTP stream just needs to pick
up. Im trying to see exactly how he is doing this, since this is the exact thing
I need, and then Ill just run OSPF on my core router (not sure if that will work
yet).
I woudl perfer to do this all in hardware vs software since a Cisco/Juniper box
is musch less prone to failure then a server with software, but I guess more
research will tell what ill be using in the end :)
Once I get this going, I want to post a entire howto on the wiki.
Thanks!
Ron
On 3/12/06, Gabriel Afana < asterisk@gafana.com> wrote:
Hi Ron,
If the SBC would have served mearly as a load balancer...I already have one
and it didn't work too well. I have a Foundry ServerIron XL load balancer
and I've tried using it with Asterisk. It has had positive and negative
results.
Positive: It *would* load balance between asterisk servers for whatever port I
set (I was using 5060 for SIP). However, I didn't mess with the RTP because
its got so many ports (and you can't add ranges for virtual server ports,
you have to enter exact ports - at least I think) and because I have no idea how
that would work if SIP signaling goes to one server and RTP goes to another???
(probably not!) I would create a virtual IP on the load balancers and have all
the phones register to this IP. When checking status of the ports on each
server, it showed 5060 for all servers was unused (0 current connections). When
I would make a call, it would show the 5060 port on one of the * servers in use
(1 current connections) and it worked fine....this is where the problem started.
Negative: When I would hang up the phone, it would still show 1 current
connection to that server's 5060 port. Every call I would make from then on
would *still* go to that same server. It seems the ports are "sticky"
or set with a keepalive. Of course I can define these options on the
ServerIron, but even with "sticky" disabled and keepalive disabled,
the port would appear active (like keepalive was enabled) and every call would
go to the same server (like "sticky" was enabled). Even if I would
shutdown asterisk on that server, it would still show an active user on that
port and when I would make the call, the call would not go through. The LB was
not failing the port. I think maybe if I keep playing with it...? Any
suggestions?
If I can get my ServerIron working, I will do a complete write up on it...but it
works only partially.
This is why I was so interested in the Juniver SBC....if it would be able to act
a proxy, do all the load balancing and instantly failover if a server fails;
basically a VoIP Load Balancer. But I guess thats not what it does. Does a
VoIP load balancer hardware exist or is the only solution right now software
proxies like SER?
- Gabe
----- Original Message -----
From: Ron McCarthy <mailto:ronmccar@gmail.com>
To: Gabriel Afana <mailto:asterisk@gafana.com> ; Asterisk Users
<mailto:asterisk-users@lists.digium.com> Mailing List -Non-Commercial
Discussion
Sent: Sunday, March 12, 2006 1:16 PM
Subject: Re: [Asterisk-Users] Clustering
Hi Gabe,
Well I was going to use the SBC to have all phone point to the SBC, and then the
SBC takes care of what servers it needs to register with, and then keep a state
of what server the RTP stream and the phone need to connect to. Basically like a
load balancer would. This is what I understood from Juniper's site. Have you
seen anything on this?
Thanks!
Ron
On 3/11/06, Gabriel Afana < asterisk@gafana.com> wrote:
Hi Ron,
I've been following your thread. I noticed you mentioned about a
Juniper Session Border Controller. I checked online and read about it, but was
unsure exactly how it could intergrate with Asterisk. How would you have
planned to use that device? I am interested because one of my upstream
providers mentioned I should be using an SBC.
- Gabe
----- Original Message -----
From: Ron McCarthy <mailto:ronmccar@gmail.com>
To: Asterisk Users <mailto:asterisk-users@lists.digium.com> Mailing List
- Non-Commercial Discussion
Sent: Friday, March 10, 2006 11:22 AM
Subject: [Asterisk-Users] Clustering
Hello All,
Ive been doing more and more research on trying to setup a cluster/load balancer
for Asterisk. All the Asterisk boxes would be using a config that is the same
between them all (via a DB), but we want one location to point the phones to,
and from there that machine/device will send it to a Asterisk server so the call
can be processed. I know you cant balance the whole call, ie: once the call is
started the RTP stream has to go to the same server, but a new call could go to
a different server if perhaps the 1st server was unreachable.
Has anyone tried this, or got this to work? Ive been looking at using a Juniper
Session Border Controller, but not sure if thats gonna do the trick, and then we
also have SER..
Any comments would be great!
Thanks
Ron
_____
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Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... -----Original Message----- From: Kristian Larsson [mailto:kristian@netatonce.se] Sent: Monday, March 13, 2006 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote:> Kevin, > > From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: > > "If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us"Pretend we have peer 123456, then put exten => 123456,2,Dial(SIP/123456) in your extensions.conf When phone 123456 becomes available and registers to the Asterisk, the dialplan will look like: exten => 123456,1,NoOp exten => 123456,2,Dial(SIP/123456) and as you know the dialplan always begin on priority 1 so if the phone is not registered you don't automatically move to priority 2. What I'm curious to know is whether there is a way to use this with SIP RealTime... there doesn't seem to exist a setting for both regexten and regcontext. Any pointers? Kristian.> What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. > > Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. > > Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? > > Doug. > > > -----Original Message----- > From: Kevin P. Fleming [mailto:kpfleming@digium.com] > Sent: Fri 3/10/2006 8:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: [Asterisk-Users] Clustering > > > > Douglas Garstang wrote: > > > I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. > > It's not broken. If you find some official documentation that says so, > then it needs to be fixed. If you read it somewhere else, then that > source is not something you should trust. > > regexten in sip.conf works just fine; it can easily be used to make an > extension 'appear' and 'disappear' from the desired context based on the > status of the peer's registration. If that context is then shared among > the Asterisk servers (via DUNDi, IAX2 switches or some other technique), > then calls to that extension will be handled by the server it registered > to automatically. > _______________________________________________ > --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-- Kristian Larsson, Net At Once AB Email: kristian@netatonce.se Phone: +46 470 592717 Cell: +46 704 910401 _______________________________________________ --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
Now that I've read that paragraph of Kevin's a few times, it strikes me that this is not a redundant configuration. If the call is handled by the Asterisk system where the phone registered, what happens if that system becomes available? Can another system (one that did not handle the registration) process the call? -----Original Message----- From: Peter Bowyer [mailto:peeebeee@gmail.com] Sent: Monday, March 13, 2006 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On 13/03/06, Douglas Garstang <dgarstang@oneeighty.com> wrote:> Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though...Like Kevin already said:> If that context is then shared among > the Asterisk servers (via DUNDi, IAX2 switches or some other technique), > then calls to that extension will be handled by the server it registered > to automatically.Use an IAX2 switch for a small, known number of servers. Consider DUNDi to extend into a larger, more dynamic 'cloud'. Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org VoIP: *5048707000@sipbroker.com FWD: **275*5048707000 VoipTalk: **473*5048707000 _______________________________________________ --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
Yes, people mention DUNDi ocassionaly. It's a shame it's completely useless as their is no documentation for it. -----Original Message----- From: Kristian Larsson [mailto:kristian@netatonce.se] Sent: Monday, March 13, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On Mon, Mar 13, 2006 at 08:12:40AM -0700, Douglas Garstang wrote:> Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though...I'm not quite certain as I build my call routing on scripts instead of Asterisk built in commands, but I beleive Dundi should be able to help you out in situations like this. Kristian> > -----Original Message----- > From: Kristian Larsson [mailto:kristian@netatonce.se] > Sent: Monday, March 13, 2006 12:22 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Clustering > > > On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote: > > Kevin, > > > > From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: > > > > "If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us" > Pretend we have peer 123456, then put > > exten => 123456,2,Dial(SIP/123456) > > in your extensions.conf > When phone 123456 becomes available and registers > to the Asterisk, the dialplan will look like: > > exten => 123456,1,NoOp > exten => 123456,2,Dial(SIP/123456) > > and as you know the dialplan always begin on > priority 1 so if the phone is not registered you > don't automatically move to priority 2. > > What I'm curious to know is whether there is a way > to use this with SIP RealTime... there doesn't > seem to exist a setting for both regexten and > regcontext. Any pointers? > > Kristian. > > > What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. > > > > Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. > > > > Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? > > > > Doug. > > > > > > -----Original Message----- > > From: Kevin P. Fleming [mailto:kpfleming@digium.com] > > Sent: Fri 3/10/2006 8:05 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Cc: > > Subject: Re: [Asterisk-Users] Clustering > > > > > > > > Douglas Garstang wrote: > > > > > I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. > > > > It's not broken. If you find some official documentation that says so, > > then it needs to be fixed. If you read it somewhere else, then that > > source is not something you should trust. > > > > regexten in sip.conf works just fine; it can easily be used to make an > > extension 'appear' and 'disappear' from the desired context based on the > > status of the peer's registration. If that context is then shared among > > the Asterisk servers (via DUNDi, IAX2 switches or some other technique), > > then calls to that extension will be handled by the server it registered > > to automatically. > > _______________________________________________ > > --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 > > > -- > Kristian Larsson, Net At Once AB > Email: kristian@netatonce.se > Phone: +46 470 592717 > Cell: +46 704 910401 > _______________________________________________ > --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-- Kristian Larsson, Net At Once AB Email: kristian@netatonce.se Phone: +46 470 592717 Cell: +46 704 910401 _______________________________________________ --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
The phone won't be able to receive any calls nor will it be able to make any calls. However, if you somehow can get the phone to register with multiple servers, the phone can still receive calls if the primary * is unavailable. How about this. I have a few Cisco 7960s which let me specify a back up proxy address so can still make out going calls if the primary is unavailable. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Douglas Garstang Sent: Monday, March 13, 2006 11:06 AM To: peter@bowyer.org; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Clustering Now that I've read that paragraph of Kevin's a few times, it strikes me that this is not a redundant configuration. If the call is handled by the Asterisk system where the phone registered, what happens if that system becomes available? Can another system (one that did not handle the registration) process the call? -----Original Message----- From: Peter Bowyer [mailto:peeebeee@gmail.com] Sent: Monday, March 13, 2006 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On 13/03/06, Douglas Garstang <dgarstang@oneeighty.com> wrote:> Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though...Like Kevin already said:> If that context is then shared among > the Asterisk servers (via DUNDi, IAX2 switches or some other technique), > then calls to that extension will be handled by the server it registered > to automatically.Use an IAX2 switch for a small, known number of servers. Consider DUNDi to extend into a larger, more dynamic 'cloud'. Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org VoIP: *5048707000@sipbroker.com FWD: **275*5048707000 VoipTalk: **473*5048707000 _______________________________________________ --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
> There's a book on my desk right now that disagrees with you... > > ISBN: 0-596-00962-3I believe Doug's experience with the TFOT book's DUNDi section was less than stellar. If memory serves, it is possible that some of the examples from the book were out of date. A few months back there was a thread full of passionate emails on this subject but as of that time there weren't a lot of people using DUNDi, or they weren't being very vocal about their successes.> > Besides, this is Linux. Sometimes you'll simply have to use the > internet, right? > > I think you might find more willing ears if you trimmed back your > negativity just a tad bit.Probably true! However, Doug's extreme difficulties with implementing some of *'s more advanced features has been an educational experience for many of us on the list. And some of us find his rants entertaining! :) -MC
Hey Ron, What you are referring to is totally dependant on the phone registration time (read: the phone itself [manufacturer]). I have Cisco phones that re-register in 30 seconds regardless of original registration server presence and I have Polycom phones that register every 5 min, only if the original registration server is NOT there. I have been fighting this issue but have not come up with a viable solution just yet. If I reduce the registration time, to say, 10 seconds, some phones will not be able to do this (Polycom). Others, like the Cisco will initiate a lot of un-wanted network traffic every 10 seconds. It's not so much an issue of when a server fails and the phone can not make outbound calls, the phone will immediately send traffic to a backup proxy if a user wants to make a call during the registration cycle and server failure as long as the phone has a backup proxy specified. The real issue is when a registration server fails, inbound calls to any phone registered on the server will not be processed (maybe voicemail) until the phones re-register to another registration server and their route becomes available within the cluster again. I have not tried SER, only because a lot of my clients use IAX and this network will be complicated enough without having to manage 2 different registration systems, SER for SIP and Asterisk for IAX. I will pound this issue until I find the answer or realize it just can not be done with Asterisk. I am struggling with this but would like suggestions. JR _____ From: Ron McCarthy [mailto:ronmccar@gmail.com] Sent: Thursday, March 16, 2006 5:19 PM To: jr.richardson@cox.net Subject: Clustering JR, This is Ron, the one sho originally started this thread on the mailing list. Ive been keeping up with all the replies, and it seems like you have it working, but I hope you can answer a few questions of mine. The phone gets registered to one server, my question is, if you say "kill" the server that phone is registered to, would it then automagically attempt to re register to on one of the other * boxes, or do you know what would happen. I know this can be done, im setting up 3 test boxes this weekend, and am going to experment, also with SER involved to handle registratiosn, unless I get * to do it all for me in a efficcinet way. Any info on this would be great! Thanks Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/c2990d37/attachment.htm
Quick question, you said a lot of your clients are using IAX....what kind of clients are these? Are they running some kind of softphone or are there hardphones that use IAX? My main concern with SER is that if I have customers where I need to put * boxes on site, it will cause issues to try to do an IAX trunk between my main facility and the on-site * box. Is it possible to do a SIP trunk?? I totally imagine yes, but I tried doing this (spent 2 hours) and had one problem after the other (mainly authentication issues). If I can trunk my main servers to on-site customer servers via SIP, I will have no problem running SER! - Gabe>Hey Ron, > > > >What you are referring to is totally dependant on the phone registration >time (read: the phone itself [manufacturer]). I have Cisco phones that >re-register in 30 seconds regardless of original registration server >presence and I have Polycom phones that register every 5 min, only if the >original registration server is NOT there. > > > >I have been fighting this issue but have not come up with a viable solution >just yet. If I reduce the registration time, to say, 10 seconds, some >phones will not be able to do this (Polycom). Others, like the Cisco will >initiate a lot of un-wanted network traffic every 10 seconds. > > > >It's not so much an issue of when a server fails and the phone can not make >outbound calls, the phone will immediately send traffic to a backup proxy if >a user wants to make a call during the registration cycle and server failure >as long as the phone has a backup proxy specified. The real issue is when a >registration server fails, inbound calls to any phone registered on the >server will not be processed (maybe voicemail) until the phones re-register >to another registration server and their route becomes available within the >cluster again. > > > >I have not tried SER, only because a lot of my clients use IAX and this >network will be complicated enough without having to manage 2 different >registration systems, SER for SIP and Asterisk for IAX. I will pound this >issue until I find the answer or realize it just can not be done with >Asterisk. > > > >I am struggling with this but would like suggestions. > > > >JR > > > > _____ > >From: Ron McCarthy [mailto:ronmccar@gmail.com] >Sent: Thursday, March 16, 2006 5:19 PM >To: jr.richardson@cox.net >Subject: Clustering > > > >JR, > >This is Ron, the one sho originally started this thread on the mailing list. > >Ive been keeping up with all the replies, and it seems like you have it >working, but I hope you can answer a few questions of mine. > >The phone gets registered to one server, my question is, if you say "kill" >the server that phone is registered to, would it then automagically attempt >to re register to on one of the other * boxes, or do you know what would >happen. > >I know this can be done, im setting up 3 test boxes this weekend, and am >going to experment, also with SER involved to handle registratiosn, unless I >get * to do it all for me in a efficcinet way. > >Any info on this would be great! > >Thanks >Ron > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/a217add3/attachment.htm
> Date: Thu, 16 Mar 2006 21:35:44 -0800 (PST) > From: asterisk@gafana.com > Subject: Re: [Asterisk-Users] RE: Clustering > To: asterisk-users@lists.digium.com > Message-ID: <20060317053544.AD57065014A@ga0> > Content-Type: text/plain; charset="iso-8859-1" > > Quick question, you said a lot of your clients are using IAX....what kind of clients are these? Are they running some kind of softphone or are there hardphones that use IAX? >I use, Iaxy's, no hard phones just yet and IAX trunks from * Servers sitting at cusomers prem. I was using AT-320EE from iaxtalk. JR JR Richardson Engineering for the Masses