[Asterisk-Users] Re: asterisk-grandstream call

[Bill Michaelson]

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/${IAXINFO}@iaxtel.com/${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,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/guest at misery.digium.com/s at default) ; 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" <asterisk at techhat.com> To: 
<asterisk-users at lists.digium.com> Subject: Re: [Asterisk-Users] 
asterisk-grandstream call Date: Mon, 9 Feb 2004 15:27:55 -0500 Reply-To: 
asterisk-users at lists.digium.com Please include your sip.conf and 
extension.conf files. Hard to say what is wrong without seeing the 
configuration ----- Original Message ----- From: "Bill Michaelson" 
<bill at cosi.com> To: <asterisk-users at lists.digium.com> 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.
>>
>> 
>