On 6/14/11 9:26 AM, Cassius Smith wrote:
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})
Hey Cassius!
Nice to hear from you, what crazy country are you deploying Asterisk
in now? You might want to checkout the DEVICE_STATE() function. Should
be able to build your ALL-PAGE-EXTS while leaving out the busy
extensions. Probably not the best solution, but the first one I thought of.
--
Russ Meyerriecks
Digium | Linux Kernel Developer
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users