[*] Generating test audio

[John Lange]

Just to be clear, generating the source audio is not the problem.

Piping it through a SIP user-agent to Asterisk is...

In any case, thanks for all the suggestions. Ultimately it just turns
out to be much easier to run Asterisk on my laptop and initiate the call
that way.

Here is the complete solution:

;sip.conf
[voip1]
type=friend
host=testhost.com
nat=yes
context=local_pstn
dtmfmode=rfc2833
canreinvite=no  
qualify=no
disallow=all   
allow=ulaw

; extensions.conf
[default]
exten => 500,1,Playback(followme/pls-hold-while-try)
exten => 500,2,Milliwatt()

; testcall file:
Channel: SIP/204xxxxxxx at voip1
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: default
Extension: 500

----

Then copy the test file into the outgoing directory:

cp testcall /var/spool/asterisk/outgoing/

-  
John Lange
http://www.johnlange.ca