Asterisk cmd Page



Pages phones, i.e. transmit a message thru multiple phone (and/or their loudspeakers)



Places outbound calls to the given technology / resource and dumps them into a conference bridge as muted participants (if the 'd' option is not specified). The original caller is dumped into the conference as a speaker and the room is destroyed when the original caller leaves.
This requires a working MeetMe installation including an Asterisk timer.


  • d - full duplex audio (i.e. not a muted conference!!)
  • q - quiet, do not play beep to caller


New in Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is:

MulticastRTP/<type>/<destination>/<control address>

Type can be either basic or linksys. Destination is the IP address and port for the RTP packets. Control address is specific to the linksys type and is used for sending the control packets unique to them.


; Paging macro:
; Check to see if SIP device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page
exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for ANY call
exten => s,2,Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,3,SIPAddHeader(Call Info
Answer-After=0) ; This is for the Snoms and Others
exten => s,3,SIPAddHeader,Call-Info
exten => s,3,SIPAddHeader(Call-Info:<sip:domain>\;answer-after=0) ; enter your domain
exten => s,4,NoOp() ; Add others here
exten => s,5,Dial(${ARG1}||)
exten => s,6,Hangup
exten => s,102,Hangup

[page] ; Paging context
exten => 202,Macro(page,SIP/polycom)
exten => 208,Macro(page,SIP/cisoo1aa)
exten => _X.,1,Macro(page,SIP/${EXTEN})

The line below goes in the context where you have your extensions:

exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/202@page&Local/208@page&Local/210@page/n&Local/interal 223@page|)


This works for Linksys SPAXXX and Snom phones. (confirmed working with Asterisk, Linksys SPA941, SPA942 & Snom 360. May 29, 2006)

It implements both paging and intercom. Other phones would work as well but you would have to adjust the SIPAddHeaders for your brand of phone. NOTE: The Linksys SPAXXX phones already have *96 assigned so if you are going use *96 in Asterisk don't forget to first remove *96 from the phones Advanced Regional settings first! (The built in Paging feature of the Linksys phones only works with the SPA9000 so its safe to reuse it)

How to use it: Users pick up phone and dial *96. They hear a beep. Dial the extension of the person you want to intercom with OR dial * to page all phones.

exten => *96,1,Goto(intercom,s,1)

exten => s,1,Answer
exten => s,2,Playback(beep)
exten => s,3,Set(TIMEOUT(digit)=5)
exten => s,4,WaitExten(10)

exten => *,1,SIPAddHeader(Call-Info: <sip:>\;answer-after=0) ; Change to your Asterisk server's IP
exten => *,2,Page(SIP/3218x1&SIP/3219x1&SIP/3220x1) ; add all your devices here

exten => _XXXX,1,SIPAddHeader(Call-Info: <sip:>\;answer-after=0) ; 4 digit extensions
exten => _XXXX,2,Dial(SIP/${EXTEN})

Here is how I got this to work for my polycom phones.

[page] ; if you cut and paste this make sure you include page under the context where your phones are
exten => *96,1,Goto(intercom,s,1)

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,Page(SIP/202&SIP/231&SIP/207) ;add all extensions here

exten => _XXX,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _XXX,2,Dial(SIP/${EXTEN})

I also had to make these changes to the sip.conf file (actually I found this is in the sip.cfg polycom provisioning file, not the asterisk sip.conf file GTM)

<alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1./>
<RING_ANSWER"Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="100" se.rt.4.ringer="11" ; you could also use 7 here


As of Aug 16 2006, the following firmware versions seem to work when using SIPAddHeader(Call-Info: sip:\;answer-after=0) for auto-answer. While using SIPAddHeader(Call-Info: answer-after=0) does work for Grandstream it does not for Aastra or Snom;
Aastra - 480i - 1.4
Grandstream - GXP2000 -
Snom - 360 - 6.2.3

See also

Asterisk | Applications | Functions | Variables | Expressions | Asterisk FAQ

Created by: jamieg, Last modification: Sat 17 of Sep, 2011 (15:00 UTC) by JustRumours
Please update this page with new information, just login and click on the "Edit" or "Discussion" tab. Get a free login here: Register Thanks! - Find us on Google+