Hi
I can't for the life of me work out why this is not
working. When in the campon contect if you hit a DTMF
key 2 you get moved to the exten => 2 defined in the
mainmenu context not the exten => 2 defined in the
campon context. What is wrong? The same happens if you
hit key 1.
[campon]
exten => _*1XXX,1,Answer
exten => _*1XXX,2,SetCallerID(${CALLERIDNUM})
exten => _*1XXX,3,SetVar(CALLEDEXTEN=${EXTEN:2,3})
exten => _*1XXX,4,ResponseTimeout(3)
exten => _*1XXX,5,Background(entagroup/campon)
exten => _*1XXX,6,SetVar(LOOPER=1)
exten => _*1XXX,7,Background(entagroup/silence)
exten => _*1XXX,8,NoOp()
exten => _*1XXX,9,GotoIf($[${LOOPER} < 10]?10:13)
exten => _*1XXX,10,Dial(Local/${CALLEDEXTEN},5,trm)
exten => _*1XXX,11,SetVar(LOOPER=$[${LOOPER} + 1])
exten => _*1XXX,12,Goto(9)
exten => _*1XXX,13,Goto(4)
exten => _*1XXX,14,Hangup
exten => 1,1,VoiceMail(b${CALLEDEXTEN})
exten => 1,2,Hangup
exten => 2,1,SetCallerID("Camped on ${CALLEDEXTEN}")
exten => 2,2,Goto(huntgroups,101,1)
exten => 2,3,Hangup
[mainmenu]
exten => s,1,Set(LOOPER=1)
exten => s,2,ResponseTimeout(6)
exten => s,3,Background(entagroup/mainmenu)
exten => s,4,Background(entagroup/silence)
exten => s,5,Set(LOOPER=$[${LOOPER} + 1])
exten => s,6,GotoIf($[${LOOPER} < 4]?mainmenu,s,2)
exten => s,7,Goto(huntgroups,0,1)
exten => t,1,GotoIf($[${LOOPER} < 4]?mainmenu,s,2)
exten => t,2,Hangup
exten => i,1,Goto(mainmenu,s,1)
exten => 1,1,Goto(sales,s,1)
exten => 2,1,Goto(finance,s,1)
exten => 0,1,Goto(huntgroups,0,1)
exten => #,1,Goto(mainmenu,s,1)
Jon Farmer
Telford, Shropshire, UK
___________________________________________________________
Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail
http://uk.messenger.yahoo.com
Jon Farmer wrote:> Hi > > I can't for the life of me work out why this is not > working. When in the campon contect if you hit a DTMF > key 2 you get moved to the exten => 2 defined in the > mainmenu context not the exten => 2 defined in the >What does your CLI output look like? Doug
Your (abridged) dialplan looks OK to me. Nonetheless, you should: 1) restart * then try again, and if that doesn't work 2) make sure you load the "correct" extensions.conf (the one you think you're loading) Good luck! On 4/5/06, Jon Farmer <viperdude_uk@yahoo.co.uk> wrote:> > Hi > > I can't for the life of me work out why this is not > working. When in the campon contect if you hit a DTMF > key 2 you get moved to the exten => 2 defined in the > mainmenu context not the exten => 2 defined in the > campon context. What is wrong? The same happens if you > hit key 1. >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/f3bbc496/attachment.htm
On 4/5/06, Jon Farmer <viperdude_uk@yahoo.co.uk> wrote:> > I can't for the life of me work out why this is not > working. When in the campon contect if you hit a DTMF > key 2 you get moved to the exten => 2 defined in the > mainmenu context not the exten => 2 defined in the > campon context. What is wrong? The same happens if you > hit key 1. >It will depend on how you get INTO the campon context, by "include", by "Goto", or directly. Regards, Steve
5 apr 2006 kl. 16.40 skrev Jon Weisman:> Anyone know how I can get SIP T working w/ Asterisk?Start with explaining your definition of "SIP T" then we can look into it :-) /Olle --- * Olle E. Johansson - oej@edvina.net * Asterisk Training http://edvina.net/training/
Well what I need is to get the info digits on a sip call (toll free orignation) and send that call out a PRI to my PSTN switch via FeatureGroupD so that I know where the call is originating from. Can I do this with Asterisk? And how??? -Jon ----- Original Message ----- From: "Olle E Johansson" <oej@edvina.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, April 05, 2006 3:37 PM Subject: Re: [Asterisk-Users] SIP T> > 5 apr 2006 kl. 16.40 skrev Jon Weisman: > >> Anyone know how I can get SIP T working w/ Asterisk? > > Start with explaining your definition of "SIP T" then we can look into it > :-) > > /Olle > > --- > * Olle E. Johansson - oej@edvina.net > * Asterisk Training http://edvina.net/training/ > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >