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