[Asterisk-Users] Newbie Voicemenu question

[Brian Mulligan]

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)