You've got a mixture of stuff here all in the same [rom] context. I would
try and simplify things by separating your incoming context from your
outbound context.... dialing from SIP/101 should go nowhere near your
inbound context.
Also, I'd be very wary of include statements. Try removing them and just
imbedding the Dial() commands directly into the [rom] context. The reason
for this is that asterisk sorts the extensions internally to assist with its
extension matching. However, extensions inside an include are only sorted
within that include, not with all the other extensions inside the context.
Its all very confusing (see
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting )
which is good enough reason to try and avoid includes.
As for detecting a key sequence during an incoming call, there is no need
for AGI scripts. I use the following...
; Caller interrupted prompts by pressing 0 (zero). This is an escape to let
caller
; authenticate themselves with a PIN...
exten = 0,1,Authenticate(/whitelist,da,4)
exten = 0,n,Set(CDR(userfield)=${CDR(userfield)}-"PIN OK")
exten = 0,n,Set(CALLERID(ani)=${CDR(accountcode)})
exten = 0,n,Set(CALLERID(num)=${CDR(accountcode)})
exten = 0,n,Set(DB_RESULT=${DB(whitelist/${CDR(accountcode)})})
exten = 0,n,Goto(whitelist,1)
This lets you place a 4-digit PIN into the astdb under "whitelist" in my
example. If caller presses '0' they will hear the prompt "please enter your
passcode followed by the pound key." If they passcode is correct, it falls
through to the next instruction. If incorrect after three attempts it says
"goodbye" and hangs up.
David
On Mon, Jul 12, 2010 at 6:34 PM, Ionel Chila <ionelch...@yahoo.com> wrote:
> Please forgive me for posting a more gereneric question around my
> extensions.conf configuration in the astlinux forum but the level of help
> and
> support in this forum is outstanding so I will give it a try.
>
> Bellow please see my entire configuration. I have my astlinux box
> configured for
> two providers, one in USA and one in Europe and my internal PAP2-NA box
> being
> client 101. My analog phone is conected to the PAP2-NA (101) box.
>
> Now, I added a custom configuration name "TRANSPARENT MENU" to allow me
> when
> dialing my own number from oustside hit a key sequence which will send me
> to an
> asociated agi script. All of that works fine except, when I am trying to
> dial
> out from my PAP2-NA (101) via the european provider from my I get this
> error.
>
> [Jul 12 17:27:24] NOTICE[2106]: chan_sip.c:15133 handle_request_invite:
> Call
> from '101' to extension '0238711050' rejected because extension not found.
>
> Is something have to do with the include statement. As soon as I remove the
> "include => outteliax" it works fine but the custom "TRANSPARENT MENU" no
> longer
> works when I dial in and try to trigger one of the agi scripts.
>
> Does it make sense :-) Can one of you guys give me some ideas how to fix
> this? Thanks bunch.....
>
> ;
> [globals]
> iROMs = SIP/101
> ;
> [rom]
> exten => _339910611,1,DIAL(SIP/101, 16)
> exten => _339910611,2,Followme(${EXTEN})
> exten => _339910611,3,Voicemail(u101)
> exten => _339910611,101,Hangup()
> ;
> ; TRANSPARENT MENU
> :
> exten =>
> s,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1)
> exten => s,2,Playback(wait-moment) ; 4 seconds of ringing music
> exten => s,3,Waitexten(1)
> exten => s,4,Goto(bypass,s,1)
> exten => 99,1,Goto(script1,s,1)
> exten => 88,1,Goto(script2,s,1)
> exten => i,1,Goto(bypass,s,1)
> ;
> ;
> include => outteliax
> include => outeurope
> include => vmail
> ;
> ; INCOMING IF NO CODE ENTERED
> ;
> [bypass]
> exten =>
> s,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1)
> exten => s,2,DIAL(SIP/101)
> exten => s,3,Voicemail(u101)
> exten => s,101,Hangup()
> ;
> ; SCRIPT-1 MENU (99)
> ;
> [script1]
> exten => s,1,Answer
> exten => s,2,AGI(script1.agi)
> ;
> ;
> ; SCRIPT-2 MENU (88)
> ;
> [script1]
> exten => s,1,Answer
> exten => s,2,AGI(script2.agi)
> ;
> ; BLACKLIST
> ;
> [blacklisted]
> exten => s,1,Answer
> exten => s,n,Wait(2)
> exten => s,n,Playback(vm-nobodyavail)
> exten => s,n,Wait(1)
> exten => s,n,Hangup
> ;
> ;
> [outteliax]
> exten => _1XXXXXXXXXX,1,DIAL(SIP/teliax/${EXTEN},90,tr)
> ;
> ;
> [outeurope]
> exten => _XXXXXXXXXX,1,DIAL(SIP/europe/4${EXTEN}, 60)
> ;
> ;
> [vmail]
> exten => 111,1,VoicemailMain(s${CALLERID(num)})
> exten => 111,2,Hangup
> ;
>
>
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