[asterisk-users] Paging: Does anyone have a simple howto for Polycoms?

[Doug]

At 08:19 8/17/2007, Dave Fullerton wrote:
 >Doug wrote:
 >> I've looked at the following pages, and they are
 >> just so garbled.  I keep going around in circles:
 >>
 >> <http://www.voip-info.org/wiki/view/Polycom+auto-answer+config>
 >> <http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page>
 >> <http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom>
 >> <http://lists.digium.com/pipermail/asterisk-users/2006-May/152764.html>
 >> <http://threebit.net/mail-archive/asterisk-users/msg23241.html>
 >> <http://www.aussievoip.com.au/wiki/freePBX-Paging>
 >> <http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card>
 >>
 >> Can anyone just show some simple working examples on
 >>
 >>    1.  The Asterisk side
 >>    2.  The Polycom side
 >>
 >
 >Here's what I use on my production system with 1.2.24. This for one of
 >our four page zones but it happens to page through the polycoms in the
 >office.
 >
 >In extensions.conf I have the following:
 >
 >[pagezones]
 >; Office page zone through phones
 >; I don't want to see page calls in my cdr reports.
 >exten => _631,1,SetAMAFlags(omit)
 >; There are actually several more phones in here but I cut them
 >; out for readability
 >exten => _631,n,Page(Local/13309 at intercom&Local/13302 at intercom)
 >exten => _631,n,Hangup
 >
 >[intercom]
 >exten => _133XX,1,Macro(pageextension,SIP/${EXTEN:1})
 >
 >[macro-pageextension]
 >; Paging macro:
 >; Check to see if device is in use and DO NOT PAGE if they are
 >; ${ARG1} - Device to page
 >;
 >exten => s,1,ChanIsAvail(${ARG1}|js) ; j for jump and s is for ANY call
 >exten => s,n,Set(_ALERT_INFO="page") ; This is for the PolyComs
 >exten => s,n,Dial(${ARG1}||)
 >exten => s,n,Hangup
 >exten => s,102,Hangup
 >
 >
 >The [pagezones] context is included in each phones context to make it
 >available. What happens is the page zone extension is dialed, the page
 >app is called with several local channels in the [intercom] context. All
 >of these channels will be dumped into a meetme conference where everyone
 >except the person paging is muted. Since it uses meetme you will need a
 >zaptel timing device or ztdummy loaded. The line in the intercom context
 >simply calls a macro (which I borrowed from voip-info.org I believe).
 >The macro first checks to see if the phone is in use. If it is not, then
 >the _ALERT_INFO header is set to "page" (more on this below) and the
 >phone is then dialed. If the phone is in use then that local channel is
 >hungup and will not be paged to. Paging a phone that is in use causes
 >some odd things to happen on both the phone and asterisk side sometimes.
 >
 >
 >On the polycom side, here's what I have set in the sip.cfg (I'm using 1.6.7)
 >
 >You must fill in values in for the <alertInfo> tag. It's near the top of
 >the config file in the <voIpProt><SIP> section. See section 4.6.1.1.3.2
 >(page 74) of the SIP 1.6 Admin Guide for details. Here is how I filled
 >mine out:
 >
 ><alertInfo voIpProt.SIP.alertInfo.1.value="page"
 >voIpProt.SIP.alertInfo.1.class="4"/>
 >
 >Notice the alertInfo.1.value is set to "page", the same as what I set
 >_ALERT_INFO to in my macro. The class is set to the ring type I want to
 >use on the phone. RingType is discussed in section 4.6.1.7.2 (page 91).
 >Mine is set to 4 which corresponds to:
 >
 ><RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer"
 >se.rt.4.timeout="500" se.rt.4.ringer="13" se.rt.4.callWait="6"
 >se.rt.4.mod="1"/>
 >
 >When one of my phones is paged it rings for 1/2 a second and then
 >automatically answers the incoming call. I have set se.rt.4.ringer="13"
 >because I have created a custom page beep ring tone. You can use one of
 >the predefined ring tones or if you don't want any page beep set the
 >ring class in the alertInfo tag to 3 which is auto-answer.
 >
 >
 >Hope that answers your question.
 >
 >-Dave

Thanks, Dave.  This looks a bit more clear
than what's up on the wiki.