Добрый день!
пытаюсь понять, что надо сделать, что бы при использовании Page/Intercom телефоны сами "подняли трубку" \ auto answer (цель: жена на кухне подняла трубку, набрала номер пейджинга и произнесла "обед готов". Из динамиков IP-телефонов эта приятная информация разнеслась по детской, подвалу и гаражу...)
Нашел даже инструкцию под мои телефоны Линксис
ну куда в нашем астериксе это писать???
This works for Linksys SPAXXX and Snom phones. (confirmed working with Asterisk 1.2.7.1, 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)
[intercom]
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:10.1.1.171>\;answer-after=0) ; Change 10.1.1.171 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:10.1.1.171>\;answer-after=0) ; 4 digit extensions
exten => _XXXX,2,Dial(SIP/${EXTEN})
Here is how I got this to work for my polycom phones.
; if you cut and paste this make sure you include page under the context where your phones are
exten => *96,1,Goto(intercom,s,1)
[intercom]
exten => s,1,Answer
exten => s,2,Playback(beep)
exten => s,3,Set(TIMEOUT(digit)=5)
exten => s,4,WaitExten(10)
exten => *,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => *,2,Page(SIP/202&SIP/231&SIP/207) ;add all extensions here
exten => _XXX,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _XXX,2,Dial(SIP/${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./>
and
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="100" se.rt.4.ringer="11" ; you could also use 7 here