[Asterisk-Users] tieline digit timeout
Tony
tony at mail.applog.com
Wed May 26 13:44:59 MST 2004
On Wed, 2004-05-26 at 14:34, Steven Critchfield wrote:
> On Wed, 2004-05-26 at 13:11, Tony wrote:
> > I'm connecting to an NEC t1 card via t100p (working great so far!)
> > however I'm having problems dialing from the NEC system to an asterisk
> > extension (sip-grandstream). If I hit the trunck line and dial REAL
> > quick 103 I get the sip extension ringing; if I don't I get an invalad
> > selection message from asterisk - and I can see on the console only one
> > or two digits arrived.
> >
> > How can I globally make asterisk wait for several seconds on that t100p?
>
> You don't want it to wait several seconds, you want it to only wait the
> amount of time to know what it is you keyed.
>
> My guess is you have some form of wildcard matching going on and you are
> getting caught early. Provide some of your configs for real help.
Here is my extensions.conf: (based on Matt's start from scratch)
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g4 ; Trunk interface
TRUNKX=Zap/g4 ; 2nd trunk interface
[default]
; Extension 8600 + 8601 conference rooms
exten => 8600,1,Meetme,8600
exten => 8601,1,Meetme,8601
; Extension 102 - Grandstream hardphone
exten => 102,1,Playback,transfer|skip ; "Please hold while..."
exten => 102,2,Dial,sip/gs102|20|t ; Ring, 20 secs max
exten => 102,3,Voicemail,u102 ; Send to voicemail...
; Extension 100
exten => 100,1,Playback(alsimove),transfer|skip ; "Please hold
while..."
exten => 100,2,Dial,sip/gs102|10|t ; Ring, 10 secs max
exten => 100,3,Voicemail,u102 ; Send to voicemail...
; Extension s
;exten => s,1,Playback(alsimove),transfer|skip ; "Please hold
while..."
;exten => s,2,Dial,sip/gs102|10|t ; Ring, 10 secs max
;exten => s,3,Voicemail,u102 ; Send to voicemail...
; Extension 103 - Grandstream hardphone
exten => 103,1,Playback,transfer|skip ; "Please hold while..."
exten => 103,2,Dial,sip/gs103|20|t ; Ring, 20 secs max
exten => 103,3,Voicemail,u103 ; Send to voicemail...
; Extension to make a new recording
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,Wait(2)
exten => 205,6,Hangup
; from nec
; nec bridge
exten => _1XX,1,Dial(zap/g4/${EXTEN})
exten => _2XX,1,Dial(zap/g4/${EXTEN})
; Extension 2000 Sipura line 1
exten => 2000,1,Dial,sip/spa2000|30|t ; Ring, 30 secs max
exten => 2000,2,Voicemail,u2000 ; Send to voicemail...
; Extension 2001 Sipura line 2
exten => 2001,1,Dial,sip/spa2001|30|t ; Ring, 30 secs max
exten => 2001,2,Voicemail,u2001 ; Send to voicemail...
; Extension 2020 rings both sipura lines
exten => 2020,1,Dial,sip/spa2000&sip/spa2001|30|t ; Ring, 30 secs
max
exten => 2020,2,Voicemail,u2000 ; Send to voicemail...
; Extension 3429 - Inbound 800 number (1-800-555-3429)
exten => _**3429,1,Ringing
exten => _**3429,2,Answer
exten => _**3429,3,Dial,sip/spa2000&sip/spa2001|30|t ; Ring, 30 secs
max
exten => _**3429,4,Voicemail,u2000 ; Send to voicemail...
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,Answer
exten => _*NXXNXXXXXX*3429,3,Dial,sip/spa2000&sip/spa2001|30|t ;
Ring, 30 secs max
exten => _*NXXNXXXXXX*3429,4,Voicemail,u2000 ; Send to
voicemail...
; dial an 800 outbound number - added 9 for nec link
exten => _91800NXXXXXX,1,Dial(${TRUNK}/9 ${EXTEN:1},,Tt)
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/9 ${EXTEN:1},,Tt)
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/9 ${EXTEN:1},,Tt)
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/9 ${EXTEN:1},,Tt)
exten => _91866NXXXXXX,2,Congestion
; dial a long distance outbound number - added 9 for nec link
exten => _91NXXNXXXXXX,1,Dial(${TRUNKX}/9 ${EXTEN:1},,Tt)
exten => _91NXXNXXXXXX,2,Congestion
; dial a local outbound number - added 9 to dial via nec
exten => _9NXXXXXX,1,Dial(${TRUNK}/9 ${EXTEN:1},,Tt)
exten => _9NXXXXXX,2,Congestion
; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup
; # timeout invalid rules
; exten => #,1,Playback(invalid) ; "Thanks for trying the demo"
; exten => #,2,Hangup ; Hang them up.
; exten => t,1,Goto(#,1) ; If they take too long, give
; exten => i,1,Playback(invalid) ; "That's not valid,"
; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
Here is what I see on the console: (I didn't dial fast the first time -
but did the second time)
*CLI> -- Starting simple switch on 'Zap/25-1'
== Unknown extension '1' in context 'default' requested
-- Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/25-1'
*CLI>
*CLI>
-- Starting simple switch on 'Zap/25-1'
-- Executing Playback("Zap/25-1", "transfer|skip") in new stack
-- Executing Dial("Zap/25-1", "sip/gs103|20|t") in new stack
-- Called gs103
-- SIP/gs103-1ac3 is ringing
== Spawn extension (default, 103, 2) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
zaptel.conf
# from fibernet to asterisk t100p-1
span=1,1,0,esf,b8zs
e&m=1-24
# from asterisk t100p-2 to nec
span=2,0,0,esf,b8zs
e&m=25-48
loadzone = us
defaultzone=us
Thanks for taking the time to look!
t o n y
More information about the asterisk-users
mailing list