Right - OK - sans comments for brevity: sip.conf: [general] port = 5060
; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context =
default ; Default for incoming calls [248379] username=billdesk
type=friend host=dynamic canreinvite=no mailbox=1234 context=demo
extensions.conf: [general] static=yes writeprotect=no [globals]
CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel
username/password TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits
to strip (usually 1 or 0) [iaxtel700] exten =>
_91700NXXXXXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider] [trunkint] exten =>
_9011.,1,Dial(${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:${TRUNKMSD}}) exten =>
_91800NXXXXXX,2,Congestion exten =>
_91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91888NXXXXXX,2,Congestion exten =>
_91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91877NXXXXXX,2,Congestion exten =>
_91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91866NXXXXXX,2,Congestion [international] ignorepat => 9 include =>
longdistance include => trunkint [longdistance] ignorepat => 9 include
=> local include => trunkld [local] ignorepat => 9 include => default
include => parkedcalls include => trunklocal include => iaxtel700
include => trunktollfree include => iaxprovider [macro-stdexten]; exten
=> s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten
=> s,2,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ u\
navail announce exten => s,3,Goto(default,s,1) ; If they press #, return
to start exten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail
w/ busy \ announce exten => s,103,Goto(default,s,1) ; If they press #,
return to start [demo] exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,5 ; Set
Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,10 ; Set
Response Timeout to 10 seconds exten => s,5,BackGround(demo-congrats) ;
Play a congratulatory message exten => s,6,BackGround(demo-instruct) ;
Play some instructions exten => 2,1,BackGround(demo-moreinfo) ; Give
some more information. exten => 2,2,Goto(s,6) exten =>
3,1,SetLanguage(fr) ; Set language to french exten => 3,2,Goto(s,5) ;
Start with the congratulations exten => 1000,1,Goto(default,s,1) exten
=> 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip
if channel is not up) exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten =>
1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,2,Voicemail(u1234)
; Unless busy exten => #,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 up exten => i,1,Playback(invalid) ; "That's
not valid, try again" exten => 500,1,Playback(demo-abouttotry); Let them
know what's going on exten =>
500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call the Asterisk
de\ mo exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo
site exten => 500,4,Goto(s,6) ; Return to the start over message. exten
=> 600,1,Playback(demo-echotest) ; Let them know what's going on exten
=> 600,2,Echo ; Do the echo test exten => 600,3,Playback(demo-echodone)
; Let them know it's over exten => 600,4,Goto(s,6) ; Start over exten =>
8500,1,VoicemailMain exten => 8500,2,Goto(s,6) [default] include => demo
From: "Glenn Dalgliesh" <[EMAIL PROTECTED]> To:
<[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users]
asterisk-grandstream call Date: Mon, 9 Feb 2004 15:27:55 -0500 Reply-To:
[EMAIL PROTECTED] Please include your sip.conf and
extension.conf files. Hard to say what is wrong without seeing the
configuration ----- Original Message ----- From: "Bill Michaelson"
<[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday,
February 09, 2004 3:15 PM Subject: [Asterisk-Users] asterisk-grandstream
call
I am trying to muddle my way tthrough getting something - actually
anything to work - with Asterisk. I've acquired a Grandstream phone and
I've got * on a Red Hat 9 box. I've gotten to a point where I can see
(via ethereal) that the phone REGISTER's successfully with asterisk, and
then I try to dial into voicemail. This is what I observe in the packet
trace...
GS: INVITE -> *
*: Status 100 (Trying) -> GS
*: Status 200 (OK with session description) -> GS
So far, seems reasonable - but I'm a complete novice with this protocol.
Then I see a huge stream of UDP packets sent by * to the GS on port
5004, but the GS only replies with an ICMP destination unreachable to
each packet. I'm guessing that this is an RTP audio stream, but I don't
know why the GS is not ready or otherwise unwilling to receive the
packets. Examining the GS config, I've confirmed that the "local RTP
port" is set to 5004.
I have many questions about how this should work, but I'll save some
bandwidth and leave it to someone here to suggest what should be checked
next.
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