[Asterisk-Users] Ringdown Circuit Configuration

Eric Wieling eric at fnords.org
Tue Mar 18 20:42:26 MST 2003


For Cisco IOS products you can use "connection plar opx <number to
dial>" and then set up a dial-peer for that number to go
wherever you want it to.  Plar is of course, Private Line Auto
Ringdown.  There are various examples on the Cisco web site.

--Eric
On Tue, Mar 18, 2003 at 08:47:22PM -0600, Stephen Webb wrote:
> You are right with the ATA. Check this doc out
> http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataincfg/186ugcc.htm
> 
> Look at the section about Pre-Dial.
> 
> I wonder if you can do this with a Cisco 7940 or Snom 100/200?
> 
> 
> On Tue, 2003-03-18 at 19:34, John Todd wrote:
> > I believe the Cisco ATA-186 supports it, but you'd have to do more 
> > digging on their site.
> > 
> > This is really not a protocol issue, but a vendor programming issue. 
> > It all depends on if you can get the hardware to do a hotline call 
> > when the phone is taken off the hook.
> > 
> > JT
> > 
> > 
> > >Does anyone know if this can be done by any VoIP Technology (SIP, IAX,
> > >IAX2 or MGCP) I don't know the protocols!
> > >
> > >
> > >On Tue, 2003-03-18 at 09:56, Steven Critchfield wrote:
> > >>  On Tue, 2003-03-18 at 09:04, Don Pobanz wrote:
> > >>  > We have need of a ringdown circuit in an elevator. If someone picks up
> > >>  > the phone, it should dial another extension without any keys being
> > >>  > pressed. (There are no keys on the phone)
> > >>  >
> > >>  > If it was an incoming call to asterisk, the following lines in
> > >>  > extensions.conf would do the trick.
> > >>  > exten => s,1,Answer
> > >>  > exten => s,2,Dial,Zap/10
> > >>  >
> > >>  > However, the 's' state is not valid for just picking up a phone
> > >>  > (extension). With nothing being dialed there are no extension matches
> > >>  > to make. It is like the dial tone needs to time out in a very short
> > >>  > time and instead of getting the busy tone, asterisk should dial an
> > >>  > extension.
> > >>
> > >>  in /etc/asterisk/zapata.conf, I think this is what you are looking for.
> > >>  ;
> > >>  ; Specify whether the channel should be answered immediately or
> > >>  ; if the simple switch should provide dialtone, read digits, etc.
> > >>  ;
> > >>  immediate=no
> > >>
> > >
> > >_______________________________________________
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> > >Asterisk-Users at lists.digium.com
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
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> 
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