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/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]) >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. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
