[Asterisk-Users] netmeeting dial

[WipeOut .]

Yes, That was what we achived in the earlier email.. but you asked about the 's' extension which is what I was eplaining in the later email.. 

> I don't wannna do that.
> 
> I want to dial FROM an H323 channel into a Zap channel.
> 
> In other words, I have a call comming from an H323 channel and I want to 
> reroute it to the Zap channel
> 
> > The 's' extension is the starting point for a call, so for example if you had
> > an FXS channel with a phone connected to it and you wanted all incomming
> > calls to ring through to that phone you would use..
> > 
> > exten => s,1,Dial(Zap/x)
> > 
> > Where x is the channel number.. for an H.323 phone I would guess it would
> > look like this..
> > 
> > exten => s,1,Dial(H323/y)
> > 
> > Where Y is the name you gave the phone config in the .conf file..
> > 
> > 
> > 
> > > Many thanks. Now I get ...
> > > 
> > > *CLI> WARNING[18450]: File pbx.c, Line 1680 (ast_pbx_run): 
> > > Channel 'H323:17672' sent into invalid extension 's' in context 'default',
> > but 
> > > no invalid handler
> > > 
> > > I'vr added your lines to my default context. I understand that I need to
> > have 
> > > a line with an 's', but I just don't know which one to use. I've tried
> > using 
> > > s,1,Wait(1) but it didn't work
> > > 
> > > Best
> > > 
> > > 
> > > > You are not passing the number to the Dial program..
> > > > 
> > > > Try this..
> > > > 
> > > > ignorepat => 9
> > > > exten => _9.,1,Dial(Zap/1/${EXTEN:1})
> > > > exten => _9.,2,Congestion
> > > > 
> > > > What this does... the 'ignorepat' will maintain a dialtone on the line
> > when a
> > > > number is dialed.. The first 'exten' says that any number dialed
> > beginning
> > > > with a 9 will be sent to channel 'Zap/1' and the number that will be
> > passed
> > > > to it is the one that was dialed without the first digit (being the 9)..
> > The
> > > > last line says that if the channel is busy you get a busy tone..
> > > > 
> > > > Hope that helps..
> > > > 
> > > > > HI, I'm using netmeeting to connect to an asterisk server and dial
> > out.
> > > > > 
> > > > > my extension looks like this
> > > > > 
> > > > > exten => s,1,Dial,Zap/1/
> > > > > 
> > > > > Unfortunatelly the number that I have dialed in Netmeeting is lost ;-(
> > 
> > > > > 
> > > > > If I hardcode the number on the line above, like ...
> > > > > 
> > > > > exten => s,1,Dial,Zap/1/6642794
> > > > > 
> > > > > ... everything works fine
> > > > > 
> > > > > What am I missing?
> > > > > 
> > > > > 
> > > > > _______________________________________________
> > > > > Asterisk-Users mailing list
> > > > > Asterisk-Users at lists.digium.com
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > 
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> > > 
> > > 
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