[Asterisk-Users] Asterisk PBX Functions via SIP phone

[James Freire]

I am suprised that one would have to create a dialplan since its an already built in function that works with regular POTS phones. Or is it because of the way DTMF is sent via SIP?

> Someone correct me if I'm wrong but I believe you'll need the dialplan for
> this one...
> 
> What I envision is doing something like this...
> 
> [verticalservice]
> 
> exten => *78,1,DbGet(${dnd}=features/dnd)
> exten => *78,2,DbPut(features/dnd=1)
> exten => *78,3,Playback(pbx-dndenabled)
> exten => *78,4,Hangup()
> exten => *78,102,GotoIf($[${dnd} = '0')]?103:104)
> exteh => *78,103,DbPut(features/dnd=1)
> exten => *78,104,Playback(pbx-dndenabled)
> exten => *78,105,Hangup()
> 
> exten => *79 ... etc...

Wouldn't you need to track each extension? something like:
exten => *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten => *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten => *78,3,Playback(pbx-dndenabled)
exten => *78,4,Hangup()
etc.?

The wiki has an exmple for call forwarding:
http://www.voip-info.org/wiki-Asterisk+call+forwarding

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