[Asterisk-Users] Newbie Voicemenu question

[David J Carter]

Brian,

You need to put an include => default in your incoming context.

some samples here http://www.codepipe.com/id25.htm

Regards

Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Brian
Mulligan
Sent: 02 March 2004 17:58
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Newbie Voicemenu question


Hi
I can get the Voicemenu stuff working OK but am unable to switch to a
context where the incoming PSTN caller is able to enter SIP number
(after promting) and have this forwarded to a proxy.What I am trying to
do is give a PSTN caller the choice between voicemail, local extension
or remote SIP user.

Below is an extract from my extensions.conf, clearly this does not work.
Any hint would be most appreciated.

Thanks
Brian

[incoming]
exten => s,1,Answer
exten => s,2,Background(brian-ivr)
exten => 6,1,Voicemail,u5152
exten => 7,1,Goto,pstn-to-sip|s|1
exten => 8,1,Dial,Zap/2
;
[pstn-to-sip]
exten =>s,1,Background(pstn-to-sip); ask user to enter sip number
exten =>9,_9.,1,Dial(SIP/${EXTEN:1}@ser1,30,r)
exten =>8,_8.,1,Dial(SIP/${EXTEN:1}@iptel,30,r)
exten =>7,_7.,1,Dial(SIP/${EXTEN:1}@fwd,30,r)




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