[asterisk-users] Page() bumps user out of a call

[Cassius Smith]

Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.

The problem comes when a user is on the line, and someone else uses the
intercom function to page all extensions, the call in progress gets
disconnected. I'm wondering if there is a way to either:
1. dynamically figure out the subset of extensions that are not in a call,
or
2. use some other function that will not bump a call?

Has anyone else run into this?

Thanks
Cassius

Here is my intercom context:

[intercom] 
exten => s,1,Answer
exten => s,n,Playback(beep)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,WaitExten(10)

exten => t,1,NoOp(timeout)
exten => t,n,Playback(sorry-youre-having-problems&goodbye)
exten => t,n,Hangup()

exten => *,1,SIPAddHeader(Call-Info: <sip:${SERVER_IP}>\;answer-after=0)
exten => *,n,Page(${ALL-PAGE-EXTS}) ; add all your devices here

exten => _XXXX,1,SIPAddHeader(Call-Info:
<sip:${SERVER_IP}>\;answer-after=0) ; 4 digit extensions
exten => _XXXX,n,Dial(SIP/${EXTEN})