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 >
_______________________________________________ -- 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