similar to: paging/intercom

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his

Has anyone else had problems with the Page() application not working under Asterisk 1.2.3? We use Cisco 7960 phones and set one of the lines to auto answer. When someone dials the paging extension it calls the page app and invites all the lines on the phones that are set to auto answer into a meetme conference where all the members are muted except the original caller. When I try to use the

I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and

Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

I am trying to page my Grandstream GXP-2000 phones and keep getting the error message: Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete destination '' supplied. How can I fix this error? The two contexts below do either one-way paging or two-way paging to all Grandstream phones in a list. [One_Way_Page_GROUP] ; one to many page exten =>

Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom); SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(P-Auto-Answer: normal); Any other ideas? Leandro PS I have set

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option "d" (full-duplex), but I need to make ringing the phone in intercom. Now I set auto-answer=6 but after first ring the phone hangup the call. There is a way to using

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

Hello list, this works : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Dial(SIP/${SIPACCOUNT}) The phone auto-answers the call... this does not work : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Page(SIP/${SIPACCOUNT}) The phone rings and does not auto-answer the call... Can you tell me

There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. Do any of the (cheaper) ip phones have a way to support intercom or paging? I presume that it's not part of the SIP or IAX protocols. Chris.

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Hi, I'm testing paging using snom 360. Can someone correct my dialplan? Regards, Jason. ================================================== ;exten => _99XXXX,1,SIPAddHeader(Call-Info: Answer-After=0) ;exten => _99XXXX,n,SIPAddHeader(Call-Info: <sip:192.168.1.113>\;answer-after=0) ;exten => _99XXXX,n,Dial(SIP/${EXTEN:2}) exten => _99XXXX,1,Set(__SIPADDHEADER=Call-Info:

I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how

Hey guys, I have run into one last issue before I do my full * conversion this evening. I can't seem to get paging to work. I have the chan_oss module loaded as per the wiki, and I have the following in my dial plan ;here is our intercom exten => 6000,1,Dial,console/dsp when I dial it here is the output from the console -- Executing Dial("SIP/3062-4f07",

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>