[Asterisk-Users] Re: Implementing Paging on the Linksys SPA9XX phones (working)

[JR Richardson]

I came up with this a few days ago, mostly used the wiki examples,
didn't have time to post on the wiki yet, maybe one of you guys with a
few minutes can throw it up there, really, I forgot my logon.

http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom

The agi script didn't work for me, wouldn't call the active hint
extensions, even though they were there, no time to debug it so I
setup direct paging in the dialplan by setting the SIPAddHeader cmd
for each phone type.

;pages a Polycom 601 (must setup polycom sip.cfg file according to wiki)
exten => *1001,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => *1001,2,Dial(SIP/1001)
exten => *1001,3,Hangup


;pages a Linksys SPA942
exten => *1003,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten => *1003,2,Dial(SIP/1003)
exten => *1003,3,Hangup


;pages a Snom 320
exten => *1005,1,SIPAddHeader(Call-Info: sip:10.10.11.255\; answer-after=0)
exten => *1005,2,Dial(SIP/1005)
exten => *1005,3,Hangup

;this group page only works for the linksys and polycom at the same time, due
;to the snom also looking for Call-Info header but a little different
than the linksys
;so can't set both.  Also Page cmd only fully functional in 1.2.7.1,
earlier versions
;the phones would not hangup after the caller hangs up, the phones kept ringing
;and other wierd issues, jsut wouldn't work correctly but 1.2.7.1
worked right away
exten => *22,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => *22,1,SIPAddHeader(Call-Info: sip:10.10.11.255\; answer-after=0)
exten => *22,2,SIPAddHeader(Call-Info:\;answer-after=0)
exten => *22,3,Page(SIP/1003&SIP/1001|)

The customer implementation is with a Polycom 601 with 2 operator
panels for the receptioninst phone and Linksys 942's for the office
staff.

The receptionist can push the buddy watch position on the Polycom
console to call the linksys extensions directly or *exten to page
them, or *22 to page all extensions, works quite well.  But of course
this is a hack, true paging needs to be multicast.

Maybe putting a a page variable in the sip.conf phone profile could
set the sip header properly when the page is called in the dial plan,
like:

[1001] ;Polycom 601
page=SIPAddHeader(Alert-Info: Ring Answer)

[1003] ;Linksys SPA942
page=SIPAddHeader(Call-Info:\;answer-after=0)

[1005] ;Snom 320
page=SIPAddHeader(Call-Info: sip:10.10.11.255\; answer-after=0)

This way, you would not have to set the sip header in the dialplan,
just call the page cmd and the sip header is set per the phone called
in the group.  This would allow for different phones to be called at
the same time.

Hope this helps.

JR

-- 
JR Richardson
Engineering for the Masses