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rschall at callone.net Guest
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Posted: Wed Jan 09, 2008 1:28 pm Post subject: [asterisk-users] Intercom & Paging with Polycoms |
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I've been able to page to a specific phone (intercom type of thing), but
I'd like to have a macro or agi that pages all phones but first checks
if their on the phone. It looked like there used to be a pageall.agi
type of script on the wiki, but that link isn't valid anymore. Does
anyone have that script, or something else that would work? I would just
do SIP/1000&SIP/1001, but there are about 60 phones involved.
Rob |
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shmaltz at gmail.com Guest
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Posted: Wed Jan 09, 2008 6:42 pm Post subject: [asterisk-users] Intercom & Paging with Polycoms |
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You can use app_page.
If you call a local channel that uses app_chanisavail first then you
should be able to call as many as you need to. You can actualy break
it down in groups that way.
On Jan 9, 2008 1:28 PM, Rob Schall <rschall at callone.net> wrote:
Quote: | I've been able to page to a specific phone (intercom type of thing), but
I'd like to have a macro or agi that pages all phones but first checks
if their on the phone. It looked like there used to be a pageall.agi
type of script on the wiki, but that link isn't valid anymore. Does
anyone have that script, or something else that would work? I would just
do SIP/1000&SIP/1001, but there are about 60 phones involved.
Rob
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rschall at callone.net Guest
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Posted: Thu Jan 10, 2008 12:40 pm Post subject: [asterisk-users] Intercom & Paging with Polycoms |
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How would I go about doing this?
The way I have paging set up (just to test it), is:
[intercom] ; Paging context
exten => s,1,Answer
exten => s,2,Playback(beep)
exten => s,3,Set(TIMEOUT(digit)=5)
exten => s,4,WaitExten(10)
exten => *,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => *,2,ChanIsAvail(SIP/5053|js)
exten => *,3,Page(SIP/5053) ;add all extensions here
exten => *,4,Hangup
exten => *,102,Hangup
exten => _XXX,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _XXX,2,ChanIsAvail(SIP/${EXTEN}|js)
exten => _XXX,3,Dial(SIP/${EXTEN})
[mynumbers]
exten => 3500,1,Goto(intercom,s,1)
But by doing it this way, I can't easily to an office page of 60 people.
I'd have to have to create an extension that says
Page(SIP/1000&SIP/1001&SIP/1002), etc. How do I have it do everyone in a
certain context (since my phones our in their own context)?
Thanks
C F wrote:
Quote: | You can use app_page.
If you call a local channel that uses app_chanisavail first then you
should be able to call as many as you need to. You can actualy break
it down in groups that way.
On Jan 9, 2008 1:28 PM, Rob Schall <rschall at callone.net> wrote:
Quote: | I've been able to page to a specific phone (intercom type of thing), but
I'd like to have a macro or agi that pages all phones but first checks
if their on the phone. It looked like there used to be a pageall.agi
type of script on the wiki, but that link isn't valid anymore. Does
anyone have that script, or something else that would work? I would just
do SIP/1000&SIP/1001, but there are about 60 phones involved.
Rob
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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_______________________________________________
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asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
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George.Pajari at netvo... Guest
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Posted: Fri Jan 11, 2008 12:12 pm Post subject: [asterisk-users] Intercom & Paging with Polycoms |
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C F wrote:
Quote: | You can use app_page.
If you call a local channel that uses app_chanisavail first then you
should be able to call as many as you need to.
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How do you make chanisavail work for SIP devices? On a system we tried
to get paging to work it always returns status 0 (Unknown) for all SIP
phones, whether in use or not. As a result, we cannot tell which phones
are being used to ensure they are not paged.
--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca
www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) |
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