Thanks Steve, I got the picture :) THANKSSSS!!!
But my doubt is about the cable, what cable should i use? i have a Sangoma A108D in one machine (one machine with one card). What cable should i do? how can i make it? Best Regards! 2010/10/5 Steve Murphy <[email protected]> > > > On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias <[email protected]>wrote: > >> Hello my friend Ingmar, >> >> I would like to know the cable you used? how was the connection? i'm using >> this one: >> >> http://wiki.sangoma.com/Pinouts#A108 Loop Back >> >> Is this ok? what should i do my friend, my problems are "understand" the >> fisicall connection :( >> >> Best Regards!!! >> >> 2010/9/24 Ingmar Steen <[email protected]> >> >>> Hi DD, >>> >>> >>> >>> We usually use loopback cables and use the open source SIP test tool >>> “SIPp” to initiate SIP calls that are sent from one group of 4 ports to >>> another group of 4 ports. >>> >>> >>> >>> Met vriendelijke groet, >>> >>> Ingmar Steen >>> >>> Teleknowledge >>> >>> >>> >>> *Van:* [email protected] [mailto: >>> [email protected]] *Namens *Danny Dias >>> *Verzonden:* vrijdag 24 september 2010 11:05 >>> *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion >>> *Onderwerp:* [asterisk-users] How to test BIG traffic through >>> DAHDI/WANPIPEinterfaces >>> >>> >>> >>> Hello Community, >>> >>> >>> >>> I need to test or simulate many calls through dahdi/wanpipe, i have a >>> Sangoma A108D, and i need to test the stability of the >>> card/drivers/firmwares with a test environment, do you think is possible? >>> >>> >>> >>> What should i do? using some loopback cable maybe? >>> >>> >>> >>> Thanks in advance >>> >>> >>> >>> DD >>> >> > I set up two machines with T1 interfaces, and connected the two with an > appropriate t1 cable. > One was acting as a network (master), the other as a subscriber (slave) > (for timing). wrote two dialplans, one for each machine, > that would answer an incoming call on one dahdi line, and call to the next > numbered line on the other > machine. The other machine was similarly outfit. I'd define the extension > for the first line on the t1, > and call it with any phone you desire. That call will cascade into 23 > separate interlinked calls. If you are > clever, the last call in should dial another real phone you have on-hand. > > You get the picture... right? Phone A dials the exten to call the first > exten on the other machine. The > dialplan should use the first channel on the t1 to place a call to the > first exten on the other machine. > On the other machine, the incoming call on channel 1 is answered, and then > a dial to the second extension > on the first machine, over the 2nd t1 channel. The first machine answers, > and uses the 3rd channel > to call the other machine.... and so on till all channels are being used. > The last exten answers and dials > a phone (dahdi or SIP, no matter) that you pick up. Such a looped call > should probably be awful, but > it's going thru 23 t1 channels! > > If you have two t1 interaces in a single card (or two cards), then you do > this on one machine. > > Another approach: set up equal numbers of FZO and FXS lines, and similarly > loop s single call thru all the > channels.This would require just one machine. > > Other approaches would involve running multiple threads to call an > extension and then hang up and > repeating this over and over again on all channels to ascertain the load > placed just by call setup and tear-down. > This kind of load is different than when all lines are just shoveling data > back and forth. > > Watch your load averages, your %cpu, your swap, etc, as the tests are in > full swing. > > murf > > > > > >> >>> -- >>> _____________________________________________________________________ >>> -- 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 >> > > > > -- > Steve Murphy > ParseTree Corp > > > -- > _____________________________________________________________________ > -- 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 >
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