[asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

[Danny Dias]

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 <murf at parsetree.com>

>
>
> On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias <ing.diasdanny at gmail.com>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 <i.steen at teleknowledge.nl>
>>
>>>  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:* asterisk-users-bounces at lists.digium.com [mailto:
>>> asterisk-users-bounces at lists.digium.com] *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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/4f407cc9/attachment-0001.htm