On Tue, 2009-03-24 at 20:19 -0500, Jim Van Meggelen wrote: > Sorry for piping in here late, but I think that YATE would be able to do that > for you fairly simply. > > It compiles easily on Windows or Linux, and has both a call and tone > generator built in.
Thanks Jim. Ultimately I solved it by installing Asterisk on my laptop and dropping testcall files into /var/lib/asterisk/outgoing. Probably should have just done that from the beginning but I kept thinking there must be a "better" way. For the sake of completeness on this thread, 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/204xxxx...@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 --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
