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

[Steve Murphy]

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





>
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-- 
Steve Murphy
ParseTree Corp
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