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

[Dave Fullerton]

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