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?
You are not passing the number to the Dial program..
Try this..
ignorepat => 9
exten => _9.,1,Dial(Zap/1/${EXTEN:1})
exten => _9.,2,Congestion
What this does... the 'ignorepat' will maintain a dialtone on the line
when a number is dialed.. The first 'exten' says that any number dialed
beginning with a 9 will be sent to channel 'Zap/1' and the number that
will be passed to it is the one that was dialed without the first digit (being
the 9).. The last line says that if the channel is busy you get a busy tone..
Hope that helps..
> HI, I'm using netmeeting to connect to an asterisk server and dial out.
>
> my extension looks like this
>
> exten => s,1,Dial,Zap/1/
>
> Unfortunatelly the number that I have dialed in Netmeeting is lost ;-(
>
> If I hardcode the number on the line above, like ...
>
> exten => s,1,Dial,Zap/1/6642794
>
> ... everything works fine
>
> What am I missing?
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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using
exten => s,1,Dial,Zap/1/
only connect you to zap channel 1
asterisk must know what you wanna dial, so
you can use (assuming that your netmeeting
client is in context local)
[local]
exten => _XX.,1,Dial(Zap/1,${EXTEN},30,r)
that matches any number with at least 2 digits
and dial it to zap/1
supposing you want to use 9 as special number
to get to the external line (so when
you dial 9+number it will dialled outside,
as normal PBX do)
[local]
exten => _9XX.,1,Dial(Zap/1,${EXTEN:1},30,r)
that matches any number, with at least 3 digits
and starting with 9 and dial to zap/1, after
stripping away the 9 (first digit)
Note that is a very basic example....
Matteo
Il lun, 2003-04-21 alle 11:25, Paulo H. Mannheimer ha
scritto:> HI, I'm using netmeeting to connect to an asterisk server and dial out.
>
> my extension looks like this
>
> exten => s,1,Dial,Zap/1/
>
> Unfortunatelly the number that I have dialed in Netmeeting is lost ;-(
>
> If I hardcode the number on the line above, like ...
>
> exten => s,1,Dial,Zap/1/6642794
>
> ... everything works fine
>
> What am I missing?
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
The 's' extension is the starting point for a call, so for example if you had an FXS channel with a phone connected to it and you wanted all incomming calls to ring through to that phone you would use.. exten => s,1,Dial(Zap/x) Where x is the channel number.. for an H.323 phone I would guess it would look like this.. exten => s,1,Dial(H323/y) Where Y is the name you gave the phone config in the .conf file..> Many thanks. Now I get ... > > *CLI> WARNING[18450]: File pbx.c, Line 1680 (ast_pbx_run): > Channel 'H323:17672' sent into invalid extension 's' in context 'default', but > no invalid handler > > I'vr added your lines to my default context. I understand that I need to have > a line with an 's', but I just don't know which one to use. I've tried using > s,1,Wait(1) but it didn't work > > Best > > > > You are not passing the number to the Dial program.. > > > > Try this.. > > > > ignorepat => 9 > > exten => _9.,1,Dial(Zap/1/${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@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > ______________________________________________ > > http://www.linuxmail.org/ > > Now with e-mail forwarding for only US$5.95/yr > > > > Powered by Outblaze > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
Yes, That was what we achived in the earlier email.. but you asked about the 's' extension which is what I was eplaining in the later email..> I don't wannna do that. > > I want to dial FROM an H323 channel into a Zap channel. > > In other words, I have a call comming from an H323 channel and I want to > reroute it to the Zap channel > > > The 's' extension is the starting point for a call, so for example if you had > > an FXS channel with a phone connected to it and you wanted all incomming > > calls to ring through to that phone you would use.. > > > > exten => s,1,Dial(Zap/x) > > > > Where x is the channel number.. for an H.323 phone I would guess it would > > look like this.. > > > > exten => s,1,Dial(H323/y) > > > > Where Y is the name you gave the phone config in the .conf file.. > > > > > > > > > Many thanks. Now I get ... > > > > > > *CLI> WARNING[18450]: File pbx.c, Line 1680 (ast_pbx_run): > > > Channel 'H323:17672' sent into invalid extension 's' in context 'default', > > but > > > no invalid handler > > > > > > I'vr added your lines to my default context. I understand that I need to > > have > > > a line with an 's', but I just don't know which one to use. I've tried > > using > > > s,1,Wait(1) but it didn't work > > > > > > Best > > > > > > > > > > You are not passing the number to the Dial program.. > > > > > > > > Try this.. > > > > > > > > ignorepat => 9 > > > > exten => _9.,1,Dial(Zap/1/${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@lists.digium.com > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > -- > > > > ______________________________________________ > > > > http://www.linuxmail.org/ > > > > Now with e-mail forwarding for only US$5.95/yr > > > > > > > > Powered by Outblaze > > > > _______________________________________________ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > ______________________________________________ > > http://www.linuxmail.org/ > > Now with e-mail forwarding for only US$5.95/yr > > > > Powered by Outblaze > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze