[asterisk-users] Stress Test new system

[Danny Nicholas]

Here's one way - set up calls to the sip provider using "local" channels
instead of actual phones.  

In extensions.conf
[monkeys]
Exten => s,1,playback(tt-monkeys)
Exten => s,n,hangup

Create Call file (monkey1.call)
Channel: sip/5551212
CallerID: Local/8
MaxRetries: 1
WaitTime: 60
retryTime: 5
extension: local/8
Context: monkeys

Cp monkey1.call /var/spool/asterisk/outgoing

Copy monkey1.call to monkeyx.call for each line you want to test 

I use this methodology to test my asterisk for 25 simultaneous calls.

If you change local/8 to sip/123 where 123 is one of your physical
extensions, this places a real call.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eddie Mikell
Sent: Wednesday, May 12, 2010 9:00 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Stress Test new system

All:

Getting ready to put the system in production.

Any suggestions on "stress testing" the system?  I'd like to initiate 
say 10 sip phone calls to make sure the provider has the bandwidth.  Can 
you do that in CLI?  I've called 4 numbers simultaneously with the hard 
phones I currently have and am thinking of adding 6 or so soft-phones to 
various pc's to make a total of ten outgoing calls at the same time.

Any thing else that can be tested before we go live (total of 60 users)?

Thanks,

Eddie Mikell

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