[Asterisk-Users] asterisk-grandstream call
Glenn Dalgliesh
asterisk at techhat.com
Mon Feb 9 15:50:12 MST 2004
I am assuming the problem you are trying to solve is no audio. Are both the phone and asterisk on public ip address?
----- Original Message -----
From: Bill Michaelson
To: asterisk-users at lists.digium.com
Sent: Monday, February 09, 2004 5:26 PM
Subject: Re: [Asterisk-Users] asterisk-grandstream call
Arg.. my posting was mangled by text-wrapping. Sorry.
Here again...
sip.conf:
[general]port = 5060 ; Port to bind tobindaddr = 0.0.0.0 ; Address to bind tocontext = default ; Default for incoming calls[248379]username=billdesktype=friendhost=dynamiccanreinvite=nomailbox=1234context=demo
extensions.conf:
[general]static=yeswriteprotect=no[globals]CONSOLE=Console/dsp ; Console interface for demoIAXINFO=guest ; IAXtel username/passwordTRUNK=Zap/g2 ; Trunk interfaceTRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)[iaxtel700]exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)[iaxprovider][trunkint]exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})exten => _9011.,2,Congestion[trunkld]exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})exten => _91NXXNXXXXXX,2,Congestion[trunklocal]exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})exten => _9NXXXXXX,2,Congestion[trunktollfree]exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})exten => _91800NXXXXXX,2,Congestionexten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})exten => _91888NXXXXXX,2,Congestionexten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})exten => _91877NXXXXXX,2,Congestionexten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})exten => _91866NXXXXXX,2,Congestion[international]ignorepat => 9include => longdistanceinclude => trunkint[longdistance]ignorepat => 9include => localinclude => trunkld[local]ignorepat => 9include => defaultinclude => parkedcallsinclude => trunklocalinclude => iaxtel700include => trunktollfreeinclude => iaxprovider[macro-stdexten];exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maxi\mumexten => s,2,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w\/ unavail announceexten => s,3,Goto(default,s,1) ; If they press #, return to startexten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ bu\sy announceexten => s,103,Goto(default,s,1) ; If they press #, return to start[demo]exten => s,1,Wait,1 ; Wait a second, just for funexten => s,2,Answer ; Answer the lineexten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 secondsexten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 secondsexten => s,5,BackGround(demo-congrats) ; Play a congratulatory messageexten => s,6,BackGround(demo-instruct) ; Play some instructionsexten => 2,1,BackGround(demo-moreinfo) ; Give some more information.exten => 2,2,Goto(s,6)exten => 3,1,SetLanguage(fr) ; Set language to frenchexten => 3,2,Goto(s,5) ; Start with the congratulationsexten => 1000,1,Goto(default,s,1)exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up)exten => 1234,2,Macro(stdexten,1234,${CONSOLE})exten => 1235,1,Voicemail(u1234) ; Right to voicemailexten => 1236,1,Dial(Console/dsp) ; Ring foreverexten => 1236,2,Voicemail(u1234) ; Unless busyexten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"exten => #,2,Hangup ; Hang them up.exten => t,1,Goto(#,1) ; If they take too long, give upexten => i,1,Playback(invalid) ; "That's not valid, try again"exten => 500,1,Playback(demo-abouttotry); Let them know what's going onexten => 500,2,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the Asterisk\ demoexten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo siteexten => 500,4,Goto(s,6) ; Return to the start over message.exten => 600,1,Playback(demo-echotest) ; Let them know what's going onexten => 600,2,Echo ; Do the echo testexten => 600,3,Playback(demo-echodone) ; Let them know it's overexten => 600,4,Goto(s,6) ; Start overexten => 8500,1,VoicemailMainexten => 8500,2,Goto(s,6)[default]include => demo
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