[asterisk-users] Variable that gives the SIP channel

2006-09-18 Thread Andre Courchesne - Consultant

Hi,

 I have a dialplan code to flash hook from a SIP phone. Everything 
works great except that it requires the SIP phone to have 2 lines since 
when the call comes back after the dialplan flash hook, the 1st line 
instance on the SIP (softphone) is still active.


 What I would like to do is in my flash hook dialplan code to ass 
something like Hangup(SIP/100-fe65), but where can I get that 
SIP/100-fe65 ? Is there a variable set with this information available 
in the dialplan ?


Andre Courchesne
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Variable that gives the SIP channel

2006-09-18 Thread Time Bandit

  What I would like to do is in my flash hook dialplan code to ass
something like Hangup(SIP/100-fe65), but where can I get that
SIP/100-fe65 ? Is there a variable set with this information available
in the dialplan ?


${CHANNEL}

have a look here : http://www.voip-info.org/wiki-Asterisk+variables

hth
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users