[asterisk-users] Split incoming call volume across queues on several asterisk servers
Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, -- Dr. Haider Raza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Split incoming call volume across queues on several asterisk servers
But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED]wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, -- Dr. Haider Raza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Split incoming call volume across queues on several asterisk servers
I guess what I want to ask is...how do I setup a proxy? In a nutshell...how are calls transfered or handed off to other asterisk servers leaving the originating server free from all call handling once the transfer is done. What dialplan command would do that? Do I setup a trunk and then Dial the call to the trunk? Maybe write an agi script to connect to manager interfaces on the different asterisk servers to see who has a spot free on their queue and then transfer on a trunk. I guess what I am not clear on is, are IAX trunks between asterisk servers what I need to accomplish this (Using a proxy or daisy chained asterisk servers)? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 11:36 PM, Alex Balashov [EMAIL PROTECTED]wrote: Proxies do not handle media, so, one can definitely handle 300 simultaneous calls. Haider Raza wrote: But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, --Dr. Haider Raza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Split incoming call volume across queues on several asterisk servers
I will now look into reinvites and openser. Thank you so much for your time and all the excellent advice. -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 11:59 PM, Alex Balashov [EMAIL PROTECTED]wrote: Asterisk is not a SIP proxy. You would have to use another piece of software, such as Kamailio/OpenSIPS (formerly OpenSER). Haider Raza wrote: I guess what I want to ask is...how do I setup a proxy? In a nutshell...how are calls transfered or handed off to other asterisk servers leaving the originating server free from all call handling once the transfer is done. What dialplan command would do that? Do I setup a trunk and then Dial the call to the trunk? Maybe write an agi script to connect to manager interfaces on the different asterisk servers to see who has a spot free on their queue and then transfer on a trunk. I guess what I am not clear on is, are IAX trunks between asterisk servers what I need to accomplish this (Using a proxy or daisy chained asterisk servers)? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 11:36 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Proxies do not handle media, so, one can definitely handle 300 simultaneous calls. Haider Raza wrote: But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? --Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, --Dr. Haider Raza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ http://www.astricon.net/ 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 --Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
