Re: [asterisk-users] load-balancing AMI and load-balancing FastAGI?
Anyone? -- _ -- 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
[asterisk-users] load-balancing AMI and load-balancing FastAGI?
Hi, I am starting a new project to develop a predictive dialler system. - Agents can start receiving calls from the queue if agent press Available button on the browser which will unpause the queue on Asterisk. - About 100-150 concurrents calls on a Asterisk box - Call-out initiated. Other end answers. Passes AMD. Lands in Queue and direct to agents that is available and call is recorded. - Update state of the call (Ringing, Talking, etc) on the database. - Listen the events such as Hang Up from customer, check if call is successfully originated or what the failure, etc. - Agent will have ability to transfer customer call to other agent or external number. As described above to develop a predictive dialler system, is it best to use AMI or FastAGI? I am aware that I can setup FastAGI load balancing such as agitator (FastAGI reverse proxy). AMI case: load-balances incoming events/response across multiple processes (multiple AMI connections on the same asterisk machine), should the ami events/response should be pushed into RabbitMQ so the proess can read from RabbitMQ ? Thanks Paul -- _ -- 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
Re: [asterisk-users] Load Balancing with DNS SRV without DUNDI
On 05/24/2015 11:01 PM, Mehdi Shirazi wrote: Hi I want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes. The problem is when a calls come to Asterisk1 but SIP endpoint is registered on Asterisk2.How we can check a SIP endpoint is registered or not and what is Contact information in Dialplan ? Regards babak If you used Opensips with a Mysql backend. The two Opensips servers could query a command db with the contact URI. -- _ -- 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
[asterisk-users] Load Balancing with DNS SRV without DUNDI
HiI want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes.The problem is when a calls come to Asterisk1 but SIPendpoint is registered on Asterisk2.How we can check a SIP endpoint is registered or not and what is Contact information in Dialplan ? Regardsbabak -- _ -- 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
Re: [asterisk-users] Load Balancing
I came accross this article (Asterisk rtp mprovements http://www.voip-forum.com/opensource/2013-04/asterisk-rtp-improvements/) mentioning DNS based load balancing. I will give Opensips loadbalance module further reading to better understand how it works Thanks for the tip. 2013/4/25 achera...@gmail.com You the couple opensips + asterisk will help you. Opensips loadbalance module is your friend. Sent from my iPhone On Apr 25, 2013, at 11:44 AM, Olivier oza_4...@yahoo.fr wrote: Hello, I've been given the task to study what would a good way to load balance SIP trafic. The prospective setup is : - call centers sending outbound SIP trafic (no inbound) from SIP devices (with public fixed IP address), - a couple of outbound SIP trunks to which trafic from call centers is to be forwarded - a load balancing system between call centers and SIP trunks. Load balancing system main task is: - provide some LCR routing, - improve availability. Can (should) it be done with Asterisk alone or should I look for other components ? Regards -- _ -- 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
[asterisk-users] Load Balancing
Hello, I've been given the task to study what would a good way to load balance SIP trafic. The prospective setup is : - call centers sending outbound SIP trafic (no inbound) from SIP devices (with public fixed IP address), - a couple of outbound SIP trunks to which trafic from call centers is to be forwarded - a load balancing system between call centers and SIP trunks. Load balancing system main task is: - provide some LCR routing, - improve availability. Can (should) it be done with Asterisk alone or should I look for other components ? Regards -- _ -- 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
Re: [asterisk-users] Load Balancing
You the couple opensips + asterisk will help you. Opensips loadbalance module is your friend. Sent from my iPhone On Apr 25, 2013, at 11:44 AM, Olivier oza_4...@yahoo.fr wrote: Hello, I've been given the task to study what would a good way to load balance SIP trafic. The prospective setup is : - call centers sending outbound SIP trafic (no inbound) from SIP devices (with public fixed IP address), - a couple of outbound SIP trunks to which trafic from call centers is to be forwarded - a load balancing system between call centers and SIP trunks. Load balancing system main task is: - provide some LCR routing, - improve availability. Can (should) it be done with Asterisk alone or should I look for other components ? Regards -- _ -- 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
Re: [asterisk-users] Load balancing Asterisk.
Foundry serverIron does support SIP and its ASIC not a linux box Load balancer like F5, Refer to Chapter 10 (page 677) of ServerIron manual. It explains everything in detail. Also you may need to play with source nat a little bit to make your specific configuration work, but it should work, at least in theory. On Thu, Nov 20, 2008 at 10:25 AM, Alex Balashov abalas...@evaristesys.comwrote: SIP wrote: As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-based SIP load balancing and only worked with certain SIP proxies. I assume that is because there is no way RFC-supported way to insert a cookie into a SIP session that persists throughout the entire exchange with a client, including all in-dialog requests, subsequent sessions, etc? The only way I know of to make a cookie stick on the UAC side is to put an LR parameter into the route set, but that will only last within a dialog. So, I'm assuming certain SIP proxies had proprietary ways of getting around that in order to work with F5? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load balancing Asterisk.
Hello! We're looking for a solution to reliably load balance our Asterisk boxes. So far we've been using a hodge-podge of directing different services to different boxes/IPs, but eventually I'd like to consolidate things so we can present a single IP address to the outside world. My question is - how do we go about doing that? I've read a lot of things like load-balancing via DUNDi or OpenSER, but it seems to me like these approaches just add to the list of possible failures. In other words I'd like to avoid software solutions. Is it possible to just put Asterisk behind a load balancer? I imagine most of them are optimized for web traffic rather than UDP voice packets. Does that matter? Would any load balancer do - or only specific models will work? my guess is any model will work, but some of them may not be able to handle the load. Any recommended models? I know there are some fancy LBs out there that can actually load balance based on the SIP session rather than something like IP, but I'm afraid to even look at the price tag. I'm more than fine with balancing by user IP address instead - if that works. :) Would appreciate any comments or ideas. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
2008/11/20 Nitzan Kon [EMAIL PROTECTED] Hello! We're looking for a solution to reliably load balance our Asterisk boxes. So far we've been using a hodge-podge of directing different services to different boxes/IPs, but eventually I'd like to consolidate things so we can present a single IP address to the outside world. My question is - how do we go about doing that? I've read a lot of things like load-balancing via DUNDi or OpenSER, but it seems to me like these approaches just add to the list of possible failures. In other words I'd like to avoid software solutions. Is it possible to just put Asterisk behind a load balancer? I imagine most of them are optimized for web traffic rather than UDP voice packets. Does that matter? Would any load balancer do - or only specific models will work? my guess is any model will work, but some of them may not be able to handle the load. Any recommended models? I know there are some fancy LBs out there that can actually load balance based on the SIP session rather than something like IP, but I'm afraid to even look at the price tag. I'm more than fine with balancing by user IP address instead - if that works. :) Would appreciate any comments or ideas. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
What do you mean by hardware options? There are no ASIC-assisted SIP load balancers out there. :-) The embedded hardware-based options are load balancers built just like PCs - often on top of a UNIX kernel - that run a software application-aware load balancing suite. Your best bet is a proxy for the round-robin part, and Linux-HA for the high availability of the proxy, as Grygoriy suggested. Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Hardware solutions are of course simply packaged software solutions. Personally I would go with something that has this wonderful support base and quick solutions versus dealing with a vendor. You did mention that price was a consideration, right? j On Thu, 20 Nov 2008, Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Alex, I realize and agree that hardware load balancers are actually software based. I'm less concerned about that and more about the general specs: Foundry ServerIron XL: rated for 1,000,000 concurrent connections Linux box where OpenSIPS is sitting: rated for ...??? Not to mention a simple rule on a load balancer would be much, much easier to implement. All I need is IP-based load balancing so installing and maintaining OpenSIPS is an overkill. Again, I appreciate the feedback but I am not asking nor looking for a software solution. My question is simple: Will a HARDWARE load balancer work? any reason why it WON'T work? Thanks! --- On Thu, 11/20/08, Alex Balashov [EMAIL PROTECTED] wrote: What do you mean by hardware options? There are no ASIC-assisted SIP load balancers out there. :-) The embedded hardware-based options are load balancers built just like PCs - often on top of a UNIX kernel - that run a software application-aware load balancing suite. Your best bet is a proxy for the round-robin part, and Linux-HA for the high availability of the proxy, as Grygoriy suggested. Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Nitzan Kon wrote: Foundry ServerIron XL: rated for 1,000,000 concurrent connections Linux box where OpenSIPS is sitting: rated for ...??? Because OpenSER's load balancer is hash-based and not stateful, it is rated for far, far more than that. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Unless the LB is SIP-aware, and can maintain a SIP session, I don't see how it would work. As the SIP command stream sends discrete commands, without some sort of basic level of session awareness, there's no guarantee over a reasonable-length call that the INVITE and BYE would even get sent to the same Asterisk box. If there are on-hold messages or transfers occurring, you add even more possibility of error into the mix. Now... you could do some sort of VERY long session timeout, but overall, that's a hack that's going to drop your concurrent connection count faster than using a smaller box would. I don't know of any functioning, SIP-aware load balancers at the moment. Doesn't mean they don't exist. I just can't think of any off the top of my head. N. Nitzan Kon wrote: Alex, I realize and agree that hardware load balancers are actually software based. I'm less concerned about that and more about the general specs: Foundry ServerIron XL: rated for 1,000,000 concurrent connections Linux box where OpenSIPS is sitting: rated for ...??? Not to mention a simple rule on a load balancer would be much, much easier to implement. All I need is IP-based load balancing so installing and maintaining OpenSIPS is an overkill. Again, I appreciate the feedback but I am not asking nor looking for a software solution. My question is simple: Will a HARDWARE load balancer work? any reason why it WON'T work? Thanks! --- On Thu, 11/20/08, Alex Balashov [EMAIL PROTECTED] wrote: What do you mean by hardware options? There are no ASIC-assisted SIP load balancers out there. :-) The embedded hardware-based options are load balancers built just like PCs - often on top of a UNIX kernel - that run a software application-aware load balancing suite. Your best bet is a proxy for the round-robin part, and Linux-HA for the high availability of the proxy, as Grygoriy suggested. Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
The solution to make this work and still work statelessly is to hash various unique identifying bits of the SIP headers without maintaining transactional, session or dialog information as such. SIP wrote: Unless the LB is SIP-aware, and can maintain a SIP session, I don't see how it would work. As the SIP command stream sends discrete commands, without some sort of basic level of session awareness, there's no guarantee over a reasonable-length call that the INVITE and BYE would even get sent to the same Asterisk box. If there are on-hold messages or transfers occurring, you add even more possibility of error into the mix. Now... you could do some sort of VERY long session timeout, but overall, that's a hack that's going to drop your concurrent connection count faster than using a smaller box would. I don't know of any functioning, SIP-aware load balancers at the moment. Doesn't mean they don't exist. I just can't think of any off the top of my head. N. Nitzan Kon wrote: Alex, I realize and agree that hardware load balancers are actually software based. I'm less concerned about that and more about the general specs: Foundry ServerIron XL: rated for 1,000,000 concurrent connections Linux box where OpenSIPS is sitting: rated for ...??? Not to mention a simple rule on a load balancer would be much, much easier to implement. All I need is IP-based load balancing so installing and maintaining OpenSIPS is an overkill. Again, I appreciate the feedback but I am not asking nor looking for a software solution. My question is simple: Will a HARDWARE load balancer work? any reason why it WON'T work? Thanks! --- On Thu, 11/20/08, Alex Balashov [EMAIL PROTECTED] wrote: What do you mean by hardware options? There are no ASIC-assisted SIP load balancers out there. :-) The embedded hardware-based options are load balancers built just like PCs - often on top of a UNIX kernel - that run a software application-aware load balancing suite. Your best bet is a proxy for the round-robin part, and Linux-HA for the high availability of the proxy, as Grygoriy suggested. Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
This baby talks about being able to do hardware SIP load balancing. http://www.f5.com/news-press-events/press/2007/20070212.html I've never used an f5 product so I can't provide any comments from experience. I did look at an f5 load balancer product once and the price was over 6 figures that was a few years ago though. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
N, SIP-aware LBs do exist - but way way out of my price range. Alex, Remember we are an Asterisk-based provider. I'm not going to drop enough money on a load balancer to go bankrupt. ;) That's exactly why I'm wondering if it's possible to do this with a DUMB load balancer. i.e. one that would cost about the same as building another Linux box for OpenSIPS. I don't need a million concurrent connections. I'd be perfectly happy with a fraction of that. Not looking to replace ATT here, just looking for something simple that will work reliably. :) My concerns with OpenSIPS: 1. It's a software based solution, which means higher chance of software-related failure, and higher chance of failure due to problems with the Linux box hosting it. 2. Overkill to install and maintain (if we can get a simpler solution) 3. Incoming calls - I admit complete ignorance. I don't know how OpenSIPS handles incoming calls, but for those to arrive at the user reliably they must arrive from the same IP address the user is registered to. Otherwise their broadband router's NAT firewall will just block the connection. How does OpenSIPS handle this? (does it handle this??) Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com --- On Thu, 11/20/08, SIP [EMAIL PROTECTED] wrote: Unless the LB is SIP-aware, and can maintain a SIP session, I don't see how it would work. As the SIP command stream sends discrete commands, without some sort of basic level of session awareness, there's no guarantee over a reasonable-length call that the INVITE and BYE would even get sent to the same Asterisk box. If there are on-hold messages or transfers occurring, you add even more possibility of error into the mix. Now... you could do some sort of VERY long session timeout, but overall, that's a hack that's going to drop your concurrent connection count faster than using a smaller box would. I don't know of any functioning, SIP-aware load balancers at the moment. Doesn't mean they don't exist. I just can't think of any off the top of my head. N. Nitzan Kon wrote: Alex, I realize and agree that hardware load balancers are actually software based. I'm less concerned about that and more about the general specs: Foundry ServerIron XL: rated for 1,000,000 concurrent connections Linux box where OpenSIPS is sitting: rated for ...??? Not to mention a simple rule on a load balancer would be much, much easier to implement. All I need is IP-based load balancing so installing and maintaining OpenSIPS is an overkill. Again, I appreciate the feedback but I am not asking nor looking for a software solution. My question is simple: Will a HARDWARE load balancer work? any reason why it WON'T work? Thanks! --- On Thu, 11/20/08, Alex Balashov [EMAIL PROTECTED] wrote: What do you mean by hardware options? There are no ASIC-assisted SIP load balancers out there. :-) The embedded hardware-based options are load balancers built just like PCs - often on top of a UNIX kernel - that run a software application-aware load balancing suite. Your best bet is a proxy for the round-robin part, and Linux-HA for the high availability of the proxy, as Grygoriy suggested. Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
I was about to say, I'm sure F5 can do it... but... price was over 6 figures Why??! It's spending money on these types of things when they are unnecessary that is the undoing of every struggling VoIP provider I watch, in the misguided belief that only will half a million dollars get you enterprise strength. That was the conventional wisdom about Linux ten years ago too. Who's saying that now? Ditto. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Nitzan Kon wrote: My concerns with OpenSIPS: 1. It's a software based solution, which means higher chance of software-related failure, and higher chance of failure due to problems with the Linux box hosting it. A little bit of proper engineering will overcome that reasonably. 2. Overkill to install and maintain (if we can get a simpler solution) Really? It is, admittedly, a somewhat recondite product, but you don't have to build everything you run into your core competency; you can divest yourself of some parts of your infrastructure and streamline and all that and get someone else to do it, like a real Enterprise. :-) Secondly, as difficult as it may be, I can't imagine anything simpler to accomplish what you're looking for. The logic required is quite granular. 3. Incoming calls - I admit complete ignorance. I don't know how OpenSIPS handles incoming calls, but for those to arrive at the user reliably they must arrive from the same IP address the user is registered to. Otherwise their broadband router's NAT firewall will just block the connection. How does OpenSIPS handle this? (does it handle this??) What role are you envisioning the proxy to be in here? If it's a registrar, it will have their IP information in the stored contact URI. If not, the calls can be sent somewhere else for resolution. Something, somewhere must know how to contact the user, yes. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Alex Balashov wrote: I was about to say, I'm sure F5 can do it... but... price was over 6 figures Why??! It's spending money on these types of things when they are unnecessary that is the undoing of every struggling VoIP provider I watch, in the misguided belief that only will half a million dollars get you enterprise strength. That was the conventional wisdom about Linux ten years ago too. Who's saying that now? Ditto. F5 has ALWAYS been overpriced. Incidentally, anyone who wants to know, F5 is a unix-based box, just like the others. Last we used the F5s, they were all running a slightly modified BSDI. And only slightly modified in packaging. As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-based SIP load balancing and only worked with certain SIP proxies. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
2. Overkill to install and maintain (if we can get a simpler solution) I am not agreed on point 2: If I understood how to install opensips + heartbeat WITHOUT knowing any program (opensips ? heartbear ?) or programming language(hell yes!) in a week ( just knew what's invite and bye ;) a more aware IT professional could do it in 2 days ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: I am not agreed on point 2: If I understood how to install opensips + heartbeat WITHOUT knowing any program (opensips ? heartbear ?) or programming language(hell yes!) in a week ( just knew what's invite and bye ;) a more aware IT professional could do it in 2 days I'm actually referring mostly to the need to build, install, and maintain another set (2?) of Linux boxes. The software is the easy part. Granted, if that's what we need to do - that's what we'll do. -- Nitzan Kon, CEO Future Nine Corporation http://www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: I am not agreed on point 2: If I understood how to install opensips + heartbeat WITHOUT knowing any program (opensips ? heartbear ?) or programming language(hell yes!) in a week ( just knew what's invite and bye ;) a more aware IT professional could do it in 2 days I'm actually referring mostly to the need to build, install, and maintain another set (2?) of Linux boxes. The software is the easy part. As someone who hates dealing with hardware, I can relate and appreciate why this is a pain. But it's a lot easier than setting up the alternatives! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
3. Incoming calls - I admit complete ignorance. I don't know how OpenSIPS handles incoming calls, but for those to arrive at the user reliably they must arrive from the same IP address the user is registered to. Otherwise their broadband router's NAT firewall will just block the connection. How does OpenSIPS handle this? (does it handle this??) That's the big question! My company uses a custom SIP Proxy and SIP Registrar so I can't speak for the details of SER derivatives but the theory is most likely the same. Our SIP Registrar records the proxy the REGISTER request arrived on and updates the Asterisk realtime database outboundproxy field with that value. When Asterisk needs to send an incoming call to the user it looks up the SIP username in the realtime database and sends the call thorugh the correct Proxy which solves the NAT issue you mention. One trick for young players here is that the outboundproxyport setting is broken in Asterisk so your Proxy will have to run on port 5060. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
SIP wrote: As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-based SIP load balancing and only worked with certain SIP proxies. I assume that is because there is no way RFC-supported way to insert a cookie into a SIP session that persists throughout the entire exchange with a client, including all in-dialog requests, subsequent sessions, etc? The only way I know of to make a cookie stick on the UAC side is to put an LR parameter into the route set, but that will only last within a dialog. So, I'm assuming certain SIP proxies had proprietary ways of getting around that in order to work with F5? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
For outbound trunking we go directly from Asterisk to the terminating gateway no SIP Proxy involved. For inbound trunking we do go through the SIP Proxy for the same reason you get users to. Incoming calls are going to be more reliable if they are not tied to a single Asterisk server (I guess you could use SRV records for your Asterisk servers for inbound trunking as well but then you're kind of duplicating the role of the SIP proxy). How do you decide which Asterisk server to send the inbound call to? If the Asterisk server that the user is registered on goes down what happens to the inbound call? Have you considered having the SIP clients register with the SIP proxy rather than Asterisk or is that too difficult to get working? Regards Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
There are a few gotchas with a SIP Proxy the main one being transfers. But if you can get away with not allowing transfers then you are best to do so as the CDR's Asterisk produces are wrong anyway. What is the transfer problem? Is it the Asterisk native type using features.conf or the SIP type using REFER that causes problems? Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] load balancing and high availability
I am evaluating the best way to make a high avail and load balanced system. I have two identical asterisk servers. Most clients are SIP phones. The only special hardware I have on both systems (they are identical) is: 1 E1 PRI card and 1 4-port BRI card. I have 8 ISDN lines so 4 go to each pbx server. I have 2 PRI lines that connect to an Alcatel PBX so each asterisk pbx has 1 PRI connection (routed the same way of course). I need to implement an active-active cluster of 2 servers. I'm new to Heartbeat and I've read this: http://www.ultramonkey.org/3/topologies/ha-lb-eg.html Could I setup Asterisk with this topology? Would I just need to have 2 identical servers? Would call routing/SIP registrations/internal astdb be handled correctly (ie. as if it were a single server)? Thanks for your input. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] load balancing
Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
On Fri, Feb 29, 2008 at 2:01 AM, Ron [EMAIL PROTECTED] wrote: Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I can think of would be to forward an incoming call for a user to all 4 of your Asterisk servers that way the call will get through but if they are not behind NAT they'll get 4 incoming calls. Bascially it's messy using the set up you've got. What you really need is a SIP Proxy (assuming you're using SIP, if not it's even trickier). The SIP Proxy could load balance requests across your Asterisk servers. For calls destined for your users you can use the outboundproxy field in the sippeers table, by setting it to the IP address of your SIP Proxy server you can get Asterisk to forward all requests for a SIP account through the proxy (there is also an outboundproxyport setting but avoid it as it's been broken forever). There are a few gotchas with a SIP Proxy the main one being transfers. But if you can get away with not allowing transfers then you are best to do so as the CDR's Asterisk produces are wrong anyway. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
Hi Greyman, Should it look like this now? Can i use 2 SIP Proxies just to make sure i have a backup? will that cause any problem again with regards to calling extension to extension? Extensions will register on the asterisk still? How about outbound calls to other SIP provider or a telcobridge, SIP proxy will handle that also? Basically asterisk will ask SIP proxy of everything? Will RTP stream still go thru asterisk? Also, i plan on setting these up as a Virtual PBX for multiple offices, which means company A can only use Trunks for A, B is for Trunk B etc etc. Does outbound to trunks have any issues? or problem is just basically calling extension to extension? [other voip provider][telcobridge] -- [pstn] || [ SIP Proxy ] | | | | [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | | mysql cluster| Thanks Regards, Ron Grey Man wrote: On Fri, Feb 29, 2008 at 2:01 AM, Ron [EMAIL PROTECTED] wrote: Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I can think of would be to forward an incoming call for a user to all 4 of your Asterisk servers that way the call will get through but if they are not behind NAT they'll get 4 incoming calls. Bascially it's messy using the set up you've got. What you really need is a SIP Proxy (assuming you're using SIP, if not it's even trickier). The SIP Proxy could load balance requests across your Asterisk servers. For calls destined for your users you can use the outboundproxy field in the sippeers table, by setting it to the IP address of your SIP Proxy server you can get Asterisk to forward all requests for a SIP account through the proxy (there is also an outboundproxyport setting but avoid it as it's been broken forever). There are a few gotchas with a SIP Proxy the main one being transfers. But if you can get away with not allowing transfers then you are best to do so as the CDR's Asterisk produces are wrong anyway. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I See the discussion a few days ago. The Asterisk server saves the value of SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL. Regards, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
On Fri, Feb 29, 2008 at 4:03 AM, Ron [EMAIL PROTECTED] wrote: Hi Greyman, Should it look like this now? Can i use 2 SIP Proxies just to make sure i have a backup? will that cause any problem again with regards to calling extension to extension? Extensions will register on the asterisk still? How about outbound calls to other SIP provider or a telcobridge, SIP proxy will handle that also? Basically asterisk will ask SIP proxy of everything? Will RTP stream still go thru asterisk? Also, i plan on setting these up as a Virtual PBX for multiple offices, which means company A can only use Trunks for A, B is for Trunk B etc etc. Does outbound to trunks have any issues? or problem is just basically calling extension to extension? [other voip provider][telcobridge] -- [pstn] || [ SIP Proxy ] | | | | [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | | mysql cluster| Thanks Regards, Ron Hi Ron, Yep it starts to get tricky :). There will be slight difference depending exactly on what you need to accomplish. I work for a VoIP Proivder that provides services to users in internet land so our set up is designed for that. If you've got VPNs or are on a LAN things will be different. Two SIP Proxy's are definitely a good idea, you can load balance your users across them using DNS SRV records, DNS Round Robin, IP Load Balancer (although then you prob should have two load balancers). If you're just starting your build network build or only have 1000 users the extra SIP Proxy should go to the bottom of the list. SIP Proxy's such as OpenSER are pretty stable and don't do anywhere near as much work as the media server. It's the fault tolerance on your Asterisk servers that is the most critical thing. They do a lot more work and in my experience with them (4+ years) they are a lot more likely to crash than your SIP Proxy. With two SIP Proxy's you have an additional problem in that now you need to set the outboundproxy field in the Asterisk realtime database to the value of the proxy the user agent came through. Asterisk can't do that for you (as far as I know) so you could possibly use the SIP Proxy to do registrations or use a custom SIP Registrar. Both are a good idea as they take registration load away from Asterisk and this can be VERY significant as your user base grows. We use a custom SIP Registrar. For outbound trunking we go directly from Asterisk to the terminating gateway no SIP Proxy involved. For inbound trunking we do go through the SIP Proxy for the same reason you get users to. Incoming calls are going to be more reliable if they are not tied to a single Asterisk server (I guess you could use SRV records for your Asterisk servers for inbound trunking as well but then you're kind of duplicating the role of the SIP proxy). The RTP stream will always be between the users and Asterisk the SIP Proxy is never invovled. If you send an RTP packet to a SIP Proxy and it will just shake its head in an irritated manner and ignore you. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
On Fri, 29 Feb 2008 6:21 +0200, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: See the discussion a few days ago. The Asterisk server saves the value of SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL. Regards, __Yehavi: Ahh that's handy. That would allow a half way solution between multiple Asterisk servers and a SIP Proxy by utilising an AGI script or database lookup in each Asterisk server's dialplan. When the incoming calls arrive you'll be able to know which Asterisk server to forward them to. You still have the problems about failing over the Asterisk servers and putting two Asterisk servers in the media path is always best avoided if possible although probably not a huge deal. Actually from memory there is something in sip.conf regarding autoregexten or something where when a SIP account registers with Asterisk it automatically adds an entry to the dialplan. If this were employed you could forward a call to all 4 Asterisk servers and only the one that had the registered user would forward the call. There are lots of ways to skin the cat but the SIP Proxy is the best way to utilise mutliple Asterisk servers when being used for SIP calls. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing SIP extensions.
Hello,
Here is how I do this. The prerequisits are:
- MySQL to hold the extensions realtime database. MySQL is synchronized
among all servers using the Master/slave replication model.
- The phones are spread by some external algorithm over the Asterisk servers
(statefull load balancer, statically defined in the config file of the
phone, etc.).
The idea is to locate on which server the destination phone is registered and
redirect the call to it. For this:
/etc/asterisk/asterisk.conf has the parameter sysname set to its IP address
(you can use also a DNS name, but I want to be independent of name
resolution). This causes the server to set the field regserver to be
saved in the MySQL database to the IP address of the server.
/etc/asterisk/extensions: The logic to check whether the value of regsever
is different from sysname and if so - redirect the call. The code
fragments are (I am using AEL):
To get the regserver from the database:
MYSQL(Query resID ${connid} SELECT regserver from sip_users where
name='${EXTEN}');
MYSQL(Fetch FetchId ${resID} RegServer);
MYSQL(Clear ${resID});
so now RegServer contains the server where the phone is registered. Next:
if((${DEVSTATE(SIP/${EXTEN})} == UNAVAILABLE) ||
(${DEVSTATE(SIP/${EXTEN})} == INVALID)) {
if(${SYSTEMNAME} != ${RegServer}) {
Transfer(SIP/[EMAIL PROTECTED]);
return;
};
};
I check for the device state so in case the phone has double registration
(primary and backup server) it will be processed localy.
Regards, __Yehavi:
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
Vieri wrote: What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there I tried to use DUNDi on my local servers but I can't seem to make it work. Most howtos out there explain the use of DUNDi when the extension ranges do not overlap. So in my case where both *1 and *2 have the same local extension range 4XXX, can I go the DUNDi route or should I stop bashing my head on that and explore another solution? If someone has configured a similar system then I'd greatly appreciate some tips. I read a few dundi docs like http://www.voip-info.org/wiki-DUNDi. Thanks Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried placing the sip registrations in a db using realtime? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
On Fri, 22 Feb 2008 19:44 +0200, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: When a call arrives I check whether the REGSERVER coloumn is the same as the local server or not. If not, then there are two options: - Pass the call via IAX to the other servers; this makes both server process the call and the audio. - Send a refer message to the caller to contact the other server. You may actually want to use a redirect message for this (e.g SIP 302 response). In any case, traversing only one server in the signaling/media path as opposed to two would generally seem more efficient. -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
--- Anthony Francis [EMAIL PROTECTED] wrote: Have you tried placing the sip registrations in a db using realtime? I'm not that sure I want to use realtime because I would then depend on the sql service never failing (I could use clustered active-active MySQL but that sounds overkill, or maybe not). I'll take a look at the pdf link of the previous post. Thanks Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] load balancing SIP extensions
What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there I tried to use DUNDi on my local servers but I can't seem to make it work. Most howtos out there explain the use of DUNDi when the extension ranges do not overlap. So in my case where both *1 and *2 have the same local extension range 4XXX, can I go the DUNDi route or should I stop bashing my head on that and explore another solution? If someone has configured a similar system then I'd greatly appreciate some tips. I read a few dundi docs like http://www.voip-info.org/wiki-DUNDi. Thanks Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there ... I've tried doing something similar and came with two options. The common to them is that I use MySQL for realtime extensions, and set systemname parameter to the IP address of the server where the phone registers. When a call arrives I check whether the REGSERVER coloumn is the same as the local server or not. If not, then there are two options: - Pass the call via IAX to the other servers; this makes both server process the call and the audio. - Send a refer message to the caller to contact the other server. I had this working in the lab but not in production yet. If you want the dialplan code for this then email me. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
--- Andres Jimenez [EMAIL PROTECTED] wrote: On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote: --- Andres Jimenez [EMAIL PROTECTED] wrote: On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote: Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. You are right. That could happen if the phone is not registered anywhere You can put some security in the dialplan. if calls comes from IAX it means that PHONE is not registered in the other server. Just create special extensions to take the IAX calls (instead of GoTo): PHONE is 101 SERVER 1 exten = 101,1, Dial SIP/101 exten = 101,1, Dial IAX-SERVER2/55101 exten = 55101,1, Dial SIP/101 exten = 55101,1, Hangup SERVER 2 exten = 101,1, Dial SIP/101 exten = 101,1, Dial IAX-SERVER1/55101 exten = 55101,1, Dial SIP/101 exten = 55101,1, Hangup I hope it helps, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
I tried to use DUNDi on my local servers but I can't seem to make it work. Most howtos out there explain the use of DUNDi when the extension ranges do not overlap. The following doc describes using the same extensions across multiple * servers. It requires using realtime, but seems to do what you describe. http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load Balancing over 2 E1 Lines
You can use the asterisk db for this. Simply set a variable to 1 or 0 if 1 set to 0 and use g2 if 0 set to 1 and use g1. - Original Message - From: Andres Jimenez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 12, 2007 11:28 AM Subject: Re: [asterisk-users] Load Balancing over 2 E1 Lines On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote: I read something about DIAL(Zap/r1/…) for using round robin, and it seems to work. That will give you the same number of calls routed to each line Is there any other possible way to make sure that all lines are used in the same amount of minutes? You are going to need an AGI app or something storing how many minutes have been routed through each line and, on every call, choosing the less used one as the line to go out. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load Balancing over 2 E1 Lines
Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/.) for using round robin, and it seems to work. Is there any other possible way to make sure that all lines are used in the same amount of minutes? Thanks in regard, Eric ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load Balancing over 2 E1 Lines
On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote: I read something about DIAL(Zap/r1/…) for using round robin, and it seems to work. That will give you the same number of calls routed to each line Is there any other possible way to make sure that all lines are used in the same amount of minutes? You are going to need an AGI app or something storing how many minutes have been routed through each line and, on every call, choosing the less used one as the line to go out. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load Balancing over 2 E1 Lines
Why not Random application available in Asterisk ? quite simple I believe. asterisk1*CLI show application Random -= Info about application 'Random' =- [Synopsis] Conditionally branches, based upon a probability [Description] Random([probability]:[[context|]extension|]priority) probability := INTEGER in the range 1 to 100 best regards, Marco Mouta On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote: Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/…) for using round robin, and it seems to work. Is there any other possible way to make sure that all lines are used in the same amount of minutes? Thanks in regard, Eric ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing SIP trunks?
On Wed, 15 Aug 2007, Nicholas Blasgen wrote:
I have 10 SIP trunks that I'd really like to round-robin load balance.
Currently I have a macro that switches between available lines, but there
really must be a function in Asterisk to do this on its own. So my question
is just that, are there any easy ways for Asterisk to either balance between
SIP trunks or even just a built in function to find the next available SIP
trunk. I think using a macro to test the state of each trunk is silly, but
it's the only method I've found.
I wonder (and sometimes question!) the wisdom of putting everything into
asterisk when it can be implemented in the dial-plan (or as I posted
recently putting stuff out of the dialplan into AGI when it can be done in
the dialplan) I'm sure there are cases where both are valid, but I'm a
great believer in the KISS principle, and if we keep the core of
asterisk clean and simple, then we can develop add-ons in the dialplan, or
elsewhere...
(And after saying that, I have to say that the dial-plan programming
language is one of the more esoteric programming languages I've used in
the 26 years or so I've been programming!)
So, if we have sip-outX as out 10 sip trunks (0-9), then: (untested ;-)
[globals]
sipTrunk=9
...
[macro-dialSipTrunk]
exten = s,1,Noop(Dialling out via round-robin SIP trunk)
exten = s,n,Set(sipTrunk=$[${sipTrunk}+1])
exten = s,n,GotoIf($[${sipTrunk}=10],skip)
exten = s,n,Set(sipTrunk=0)
exten = s,n(skip)Noop(SIP dialling on trunk: ${sipTrunk})
exten = s,n,Dial(SIP/sip-out${sipTrunk},${ARG1},${ARG2})
... maybe stuff here to deal with result of the DIAL.
... eg. on congestion you might to jump back to step 2
... but if you did that then you might want to start a 2nd counter
... which when it reached 10, you're SOL and can return congestion
... to the caller.
Maybe I've just been writing too much dialplan stuff lately!!!
Gordon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing SIP trunks?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nicholas Blasgen wrote: I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using a macro to test the state of each trunk is silly, but it's the only method I've found. If the SIP trunks are other Asterisk machines you have access to you could use DUNDi: http://www.asterisk.org/node/48321 - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGxGMtDQNt8rg0Kp4RApvUAKCW6M9hbvALuRVp4m6acOW3D+ifQgCgjNiM T1HuCDx4NrlMHTY5S/3ZFUc= =W27O -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load balancing SIP trunks?
I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using a macro to test the state of each trunk is silly, but it's the only method I've found. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load Balancing
Hello Users, How can I perform the load Balancing in My SIP server of Both OpenSER and Asterisk , Currently I have One OpenSER server and Asterisk Server, For OpenSER is to need use these modules, and is any 1) LCR and Dispatcher modules, 2) OSP Modules ( also need ) Please can anyone help me .. -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.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
[asterisk-users] Load balancing of IAX2
hI any idea how to loadbalance IAX2 trafic to multiple asteirsk thanks kAMRAN __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] Load balancing of IAX2
Hi,
Kamran Ahmad wrote:
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
Use app_random:
exten = _X.,2,Random(50:6)
exten = _X.,3,Dial(IAX2/server01/${EXTEN})
exten = _X.,4,Dial(IAX2/server02/${EXTEN})
exten = _X.,5,Goto(8)
exten = _X.,6,Dial(IAX2/server02/${EXTEN})
exten = _X.,7,Dial(IAX2/server01/${EXTEN})
exten = _X.,8,Congestion
exten = i,1,Congestion
Florian
___
--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
[Asterisk-Users] load balancing
Hi list, I was wondering i anybody ever tried to use asterisk on an openmosix loadbalancing cluster. Obviously, hw-related processes can not migrate from one system to another, but any other pricess else should be able. Or not? Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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
[Asterisk-Users] Load-balancing / offload question
I have 3 trunks coming into my pbx, I want one to be the main number. I would like to take calls coming into that trunk (trunk 1) and send them to another trunk that is not busy (trunk 2 or 3). Outbound is easily handled by me assigning max channels and and outbound routing order to each. Incoming I want trunk 1 to always be available unless the other two are in use. Any way to accomplish this, has anyone done this with asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Load Balancing with SER
hello Can we use SER in front of 10 Asterisk for load balancing. any idea Thanks in advance Kamran Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Load balancing for each protocol
Hello, I'm trying to find a good solution for load-balancing of several Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have a front VoIP router (or several) who dispatches the calls of the different boxes. This routing can be done with SER for SIP (redirect server) GnuGK for H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk box to act as a Redirect server and not as a proxy ? Any help would be appreciated, even you're telling me I'm going in the wrong direction. cheers, Yves. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load balancing for each protocol
Yves wrote: Hello, I'm trying to find a good solution for load-balancing of several Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have a front VoIP router (or several) who dispatches the calls of the different boxes. This routing can be done with SER for SIP (redirect server) GnuGK for H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk box to act as a Redirect server and not as a proxy ? Any help would be appreciated, even you're telling me I'm going in the wrong direction. You're looking for notransfer=no (the default behaviour for IAX2) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load balancing for each protocol
On 00:03, Mon 13 Jun 05, Matt Riddell wrote: Yves wrote: Hello, I'm trying to find a good solution for load-balancing of several Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have a front VoIP router (or several) who dispatches the calls of the different boxes. This routing can be done with SER for SIP (redirect server) GnuGK for H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk box to act as a Redirect server and not as a proxy ? Any help would be appreciated, even you're telling me I'm going in the wrong direction. Hi, Are the asterisk boxes on the same subnet ? If so, you could setup an OpenBSD CARP combi. It will also provide failover for when one of the boxes go down. All you have to do then is to setup the load balancing stuff in PF and you're set. What's next is the voicemail and dialplan propagation on all the boxes that take part in the loadbalance setup. Guess you already have something in mind for that cause your post is talking about connections. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load balancing for each protocol
I was more thinking about a linux-ha heartbeat system between the redundant devices. The voicemail propagation is not a problem as I don't use it, and the dialplan is stored on a remote database (that's another problem to make redundant :) ). The point I have to check now is how to configure GnuGK,SER Asterisk to do round-robin routing. I'm looking forward to see it work ... that's a really interesting project. Yves. Michiel van Baak wrote: On 00:03, Mon 13 Jun 05, Matt Riddell wrote: Yves wrote: Hello, I'm trying to find a good solution for load-balancing of several Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have a front VoIP router (or several) who dispatches the calls of the different boxes. This routing can be done with SER for SIP (redirect server) GnuGK for H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk box to act as a Redirect server and not as a proxy ? Any help would be appreciated, even you're telling me I'm going in the wrong direction. Hi, Are the asterisk boxes on the same subnet ? If so, you could setup an OpenBSD CARP combi. It will also provide failover for when one of the boxes go down. All you have to do then is to setup the load balancing stuff in PF and you're set. What's next is the voicemail and dialplan propagation on all the boxes that take part in the loadbalance setup. Guess you already have something in mind for that cause your post is talking about connections. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIPload balancer
I need to do load balancing only for the following functionalities: 1) Registration of SIP clients to * servers. 2) Load balancing of the INVITEs from SIP clients to different * servers. I'm not interested in supporting the features, which you have mentioned below. I'm not aware how the below mentioned features would be suppported in load balancing. -Jagan On Fri, 11 Mar 2005 08:54:39 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: How do you plan on supporting call queues, parking and agents with 2 * servers? This is something that has blocked us from being able to do our own SER-based load balancing. -Matthew Jagan Mohan wrote: Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 No proxies are up - can not send message to anyone Xlite is not able to register to the asterisk server. Is there anything which needs to be tweaked on Asterisk side to get this working? Please help. Thanks, Jagan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load Balancing b/w 2 asterisk servers usingSIPload balancer
Jagan Mohan wrote: I need to do load balancing only for the following functionalities: 1) Registration of SIP clients to * servers. 2) Load balancing of the INVITEs from SIP clients to different * servers. I'm not interested in supporting the features, which you have mentioned below. I'm not aware how the below mentioned features would be suppported in load balancing. -Jagan On Fri, 11 Mar 2005 08:54:39 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: How do you plan on supporting call queues, parking and agents with 2 * servers? This is something that has blocked us from being able to do our own SER-based load balancing. -Matthew Jagan Mohan wrote: Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 No proxies are up - can not send message to anyone Xlite is not able to register to the asterisk server. Is there anything which needs to be tweaked on Asterisk side to get this working? Please help. Thanks, Jagan SER can do what you want. Google for some example SER configs. I found a nice one, but don't have the link here at work. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIP load balancer
Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 No proxies are up - can not send message to anyone Xlite is not able to register to the asterisk server. Is there anything which needs to be tweaked on Asterisk side to get this working? Please help. Thanks, Jagan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIPload balancer
How do you plan on supporting call queues, parking and agents with 2 * servers? This is something that has blocked us from being able to do our own SER-based load balancing. -Matthew Jagan Mohan wrote: Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 No proxies are up - can not send message to anyone Xlite is not able to register to the asterisk server. Is there anything which needs to be tweaked on Asterisk side to get this working? Please help. Thanks, Jagan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
I beleive what you're looking for is a scalable SIP proxy, like SER :) That way, all clients registers to SER and SER redirects the caller to one of the asterisk boxes. Search the wiki at voip-info.org for asterisk at large :) Yes, that is one of the many pages I've read. But we still have a problem. Take a look at this image to get a better idea of my end goal. http://drmac.homeunix.net/images/load_balancer.jpg You won't need the second balancer. SER can do that. For growth, all you do is add more SER and more Asterisk boxes. Are you sure one SER box won't be sufficient? But if Asterisk won't work correctly with the load balancing due to packet movement, then I need to approach this differently. perhaps setting up a second SER box for failover will do? just failover with heartbeat or something... roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
Hi! http://drmac.homeunix.net/images/load_balancer.jpg You won't need the second balancer. SER can do that. Seconded. For growth, all you do is add more SER and more Asterisk boxes. Are you sure one SER box won't be sufficient? Makes sense to me to have these TWO - you can take one of those off-line without interrupting service, and that's the entire idea of this discussion, isn't it? ;- Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
For growth, all you do is add more SER and more Asterisk boxes. Are you sure one SER box won't be sufficient? Makes sense to me to have these TWO - you can take one of those off-line without interrupting service, and that's the entire idea of this discussion, isn't it? ;- Yeah Get two cisco load balancers. One of them _will_ fail. Put them in front of two SER boxes, crossover connected. Get a gigabit switch with a good backplane Put your asterisk servers behind the SER Play :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
Inline... I've read several other emails and pages on the wiki but none give any deffinate answers. if you have 20 asterisk servers each with 4 pri's, all running RealTime Extensions and RealTime SIPBuddies from the same MySQL server, what prevents you from putting all 20 servers behind a single load balancer? That way all of your UA's can use the same IP to register to; vs maintaining which customer is assigned to which machine. Load balancers vary rather dramatically in exactly how they fucntion. Some work at layer-2, others at layer-3, and some at layers above. Some include a small app that executes on each server to monitor processor utilitization, etc, communicating key parameters to the balancer. I've not tried any of these with *, but I'd have to guess that selecting a specific model that properly handles udp sessions (including variable length registrations) might require some resarch that is a little more extensive then what the causual observer might guess. perhaps its just that i am not that familiar with load balancers. i was under the impression that a load balancer could/would send each recieved packet to a different server. That assumption is basically correct, however most balancers will maintain some sort of session-oriented function that will try to keep the flow directed to the server for which it first assigned the traffic, keeping in mind that it balances 'inbound' data flows not outbound traffic. this doesn't matter in the case of register requests since all asterisk boxes share same SIP registry database. I'd have to guess that registrations would be the tricky part of an implementation simply because there are so many variations of that. (Eg, some devices/systems register every minute while others every hour, and about everything in between. From a load balancer perspective, does the first registration look any different from the second and follow-on registrations, and would the balancer consider those as the same or different end points? Might that cause a flurry of other system activities that have not been considered?) but what about invite requests and the rtp stream? you would have a majorly broken conversation if each packet in the rtp stream went to a different asterisk box. No, there are parameters available to cause all packets associated with a session to stay with the initial system and not try to load balance on a per-packet basis. Some balancers refer to the parameter as a sticky bit. However, careful thought has to be given to how the balancer functions when an rtp session is _first_ initiated from an internal * system verses a remote * system as an example. or are load balancers SIP aware? or is there some sort of session control that the balancer is aware of and will send all packets in a sip session to the same asterisk box? I have not heard of any balancer being sip-aware, and would suspect that some of the nat-like issues probably apply to load balancers. It certainly would _not_ be cool for the balancer to treat the rtp session setup as a new session when its tied directly to the sip negotitation process. and then what about meet me conferences? if 10 UA's all dial a conference DID number and all 10 get balanced to 10 different servers then they are all sitting in seperate rooms right? Meetme (as well as other * functions) would certainly need to be well thought out before considering a balancer. (Eg, where does the customer's voicemail actually reside? How much inter-system traffic would generated because various resources are scattered across multiple systems such as meetme sessions, etc?) There are obviously a lot of folks on the list that shot from the hip with little or no practical experience. Load balancing will require a little more well thought out engineering then that. I'm not sure the actual realtime * implementation is at a point where these issues can be addressed today. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
I'd have to guess that registrations would be the tricky part of an implementation simply because there are so many variations of that. Actually, this is the easiest part. It doesn't matter how often a UA registers nor does it matter to which of the 20 servers handles the registration since all servers share the same database tables. Meetme (as well as other * functions) would certainly need to be well thought out before considering a balancer. (Eg, where does the customer's voicemail actually reside? Voicemail is not a problem. Again, all voicemails are stored in database including the audio portions. The problem with MeetMe conferences still bugs me. I was un-aware that UDP had sessions. Keep in mind that all 20 servers are basically clones of eachother. They all share the same extensions.conf, sip.conf, voicemail.conf (all via RealTime). -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
I'd have to guess that registrations would be the tricky part of an implementation simply because there are so many variations of that. Actually, this is the easiest part. It doesn't matter how often a UA registers nor does it matter to which of the 20 servers handles the registration since all servers share the same database tables. The actual registration interaction (those few packets) I wouldn't expect to be an issue either. My comment was more oriented towards the more real time interactions of call handling shortly after the registration process, and what _might_ be impacted in terms of those calls. By that I mean, a call (in either direction) starts out using sip to negotiate an rtp session. If a sticky bit is applied, then all traffic from a specific IP address is essentially assigned to a single server. If the sticky bit is not used, then the load balancer _may_ send the initial rtp data flow to a different server, thus breaking the sip negotiation process (the call won't get set up). Meetme (as well as other * functions) would certainly need to be well thought out before considering a balancer. (Eg, where does the customer's voicemail actually reside? Voicemail is not a problem. Again, all voicemails are stored in database including the audio portions. The problem with MeetMe conferences still bugs me. I was un-aware that UDP had sessions. I was using the term more generically. The application assumes udp sessions exist; layer-three doesn't contain session data. In other words, from a load balancer perspective, there is noting in an individual packet for it to recognize a session. Therefore, the load balancer has to keep track of these so called sessions at layer-3 only (eg, ip address). The balancer (again) in watching/balancing incoming connections and doesn't really know about outbound data. So, if server1 was _completing_ 90% of all outgoing calls, how would the balancer know that it should not allocate another _incoming_ rtp session to that server? (Maybe a poor example, but think that process through and I'm not sure a load balancer can truly deal with the problem.) If the sticky bit kind of thing is applied, then a business customer with an * box will send all calls to the same itsp server. Analyzing the call setups from both an incoming and outgoing perspective becomes very important. Separating the two is certainly doable, but more thought has to be given to the different sip setup states to ensure the process flows correctly and still load balances. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
I've read several other emails and pages on the wiki but none give any deffinate answers. if you have 20 asterisk servers each with 4 pri's, all running RealTime Extensions and RealTime SIPBuddies from the same MySQL server, what prevents you from putting all 20 servers behind a single load balancer? That way all of your UA's can use the same IP to register to; vs maintaining which customer is assigned to which machine. snip I beleive what you're looking for is a scalable SIP proxy, like SER :) That way, all clients registers to SER and SER redirects the caller to one of the asterisk boxes. Search the wiki at voip-info.org for asterisk at large :) roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
I beleive what you're looking for is a scalable SIP proxy, like SER :) That way, all clients registers to SER and SER redirects the caller to one of the asterisk boxes. Search the wiki at voip-info.org for asterisk at large :) Yes, that is one of the many pages I've read. But we still have a problem. Take a look at this image to get a better idea of my end goal. http://drmac.homeunix.net/images/load_balancer.jpg For growth, all you do is add more SER and more Asterisk boxes. But if Asterisk won't work correctly with the load balancing due to packet movement, then I need to approach this differently. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
Matthew, i think it would be convenient that you use dns round-robin for load balancing, registering the clients against Ser or Asterisk boxes. Greetings. Ariel. - Original Message - From: "Matthew Boehm" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Wednesday, February 02, 2005 5:43 PM Subject: Re: [Asterisk-Users] load balancing 20 asterisk servers I beleive what you're looking for is a scalable SIP proxy, like SER :) That way, all clients registers to SER and SER redirects the caller to one of the asterisk boxes. Search the wiki at voip-info.org for "asterisk at large" :) Yes, that is one of the many pages I've read. But we still have a problem. Take a look at this image to get a better idea of my "end goal". http://drmac.homeunix.net/images/load_balancer.jpg For growth, all you do is add more SER and more Asterisk boxes. But if Asterisk won't work correctly with the load balancing due to packet movement, then I need to approach this differently. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.3 - Release Date: 31/01/2005 No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.3 - Release Date: 31/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] load balancing 20 asterisk servers
I've read several other emails and pages on the wiki but none give any deffinate answers. if you have 20 asterisk servers each with 4 pri's, all running RealTime Extensions and RealTime SIPBuddies from the same MySQL server, what prevents you from putting all 20 servers behind a single load balancer? That way all of your UA's can use the same IP to register to; vs maintaining which customer is assigned to which machine. perhaps its just that i am not that familiar with load balancers. i was under the impression that a load balancer could/would send each recieved packet to a different server. this doesn't matter in the case of register requests since all asterisk boxes share same SIP registry database. but what about invite requests and the rtp stream? you would have a majorly broken conversation if each packet in the rtp stream went to a different asterisk box. or are load balancers SIP aware? or is there some sort of session control that the balancer is aware of and will send all packets in a sip session to the same asterisk box? and then what about meet me conferences? if 10 UA's all dial a conference DID number and all 10 get balanced to 10 different servers then they are all sitting in seperate rooms right? hints, opinions, facts...all welcome and appreciated. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] load balancing 20 asterisk servers
Hi, You may want to look into LVS (Linux Virtual Server). It allows load ballancing in a highly configurable way. http://www.linuxvirtualserver.org/ We use it on our web and mail server to load ballance across multiple hosts. The way we have it configured it will maintain a session for 15 minutes between a client and a specific server. So long as you have qualify=yes in your configuration files, each client will continue to talk to the one server until they are turned off/ deactivated for at least 15 minutes (or whatever time period you configure into it). I've not tested LVS with Asterisk, but it may be the right direction for you to take. Cheers, -Shaun Matthew Boehm wrote: I've read several other emails and pages on the wiki but none give any deffinate answers. if you have 20 asterisk servers each with 4 pri's, all running RealTime Extensions and RealTime SIPBuddies from the same MySQL server, what prevents you from putting all 20 servers behind a single load balancer? That way all of your UA's can use the same IP to register to; vs maintaining which customer is assigned to which machine. perhaps its just that i am not that familiar with load balancers. i was under the impression that a load balancer could/would send each recieved packet to a different server. this doesn't matter in the case of register requests since all asterisk boxes share same SIP registry database. but what about invite requests and the rtp stream? you would have a majorly broken conversation if each packet in the rtp stream went to a different asterisk box. or are load balancers SIP aware? or is there some sort of session control that the balancer is aware of and will send all packets in a sip session to the same asterisk box? and then what about meet me conferences? if 10 UA's all dial a conference DID number and all 10 get balanced to 10 different servers then they are all sitting in seperate rooms right? hints, opinions, facts...all welcome and appreciated. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
