Hi, I'm trying to setup Asterisk as an outgoing SIP dial tester. There will be no phones connected to this installation, and I don't need to process incoming calls. I just need to dial a number, have the person acknowledge the call, and log that fact. (Basically an automated soft phone). I found some info on how to do this here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out. My company already has a functioning SIP/VOIP gateway. I can configure soft-phones to talk it and make calls, so I know my credentials are good. I've configured Asterisk to registers to this SIP server. That appears to be working ok, a "sip show registry" shows the registry. I've tried dropping call files into the outgoing directory, but asterisk appears to ignore them. I'm guessing my problem is that I haven't properly configured "extensions.conf" to dial thru the SIP provider. (Or haven't properly defined the provider in "sip.conf". I'm using Asterisk version 1.4.21, on Red Hat Enterprise Linux 5.1. I'm trying to use the minimal configuration possible, so I'm not using the sample configuration files. Configuration files follow. Any ideas? Joe ========================sip.conf=========================== [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allowguest=no allowsubscribe=no allowtransfers=no allowsauthreject=yes defaultexpiry=1200 dumphistory=yes registerattempts=3 registertimeout=30 sipdebug=yes register => user1:[EMAIL PROTECTED] [user1] type=friend host=sip.example.com fromuser=user1 fromdomain=example.com secret=password dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw insecure=invite =========================================================== ========================extensions.conf==================== [globals] [general] ;autofallthrough=yes [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [incoming_calls] [internal] exten => 500,1,Verbose(1|Echo test application) exten => 500,n,Echo() exten => 500,n,Hangup() [phones] include => internal include => outgoing_calls [outgoing_calls] ; Dial(technology/user[:[EMAIL PROTECTED]:port][/remote_extension]) exten => _X.,1,NoOp() exten => _X.,n,Dial(SIP/user1/${EXTEN}) [outboundmsg1] exten => s,1,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds exten => s,2,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds exten => s,3,Answer exten => s,4,Wait(1) exten => s,5,Background(hello-world) ; "play outbound msg" ; "Press 1 to replay or 2 to acknowledge receiving this message" exten => s,6,Background(tt-monkeys) exten => s,n,Noop(Waiting for input) exten => s,n(end),WaitExten(60,) exten => 1,1,Goto(s,5) ; replay message exten => 2,1,Goto(msgack,s,1) ; acknowledge message exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup [msgack] exten => s,1,Playback(auth-thankyou) exten => s,2,Playback(vm-goodbye) exten => s,3,Hangup ; at this point we might want to log the message acknowledgement somewhere ; and perhaps trigger some additional processing =========================================================== _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
