similar to: Paging possible on an ATA?

Displaying 20 results from an estimated 10000 matches similar to: "Paging possible on an ATA?"

2003 Aug 08
5
ip phones and intercom/paging
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.
2004 Aug 20
1
Sipura partners with Linksys for new combo router/SIP ATA
Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net
2007 Dec 07
2
Sidetone with Snom 370
Hi all, I'm not getting any sidetone on my Snom 370. I searched the web and the snom wiki, but I don't see any place to enable/adjust it. Callers say I sound great on the other end, but I don't hear myself so it is a little off-putting. Any suggestions would be appreciated. On a related note, some times (maybe 1 out of 10 calls) I get the side tone, but its delayed by a second or
2004 Jan 17
6
Zone Paging
I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike
2007 Nov 27
5
SIP port 5060 closed - how do I open it?
Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0
2008 Jan 25
1
Need sample configuration files for sipura/linksys ata
Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Oct 16
1
OT - SIP Auto-Answer for Cisco 7940/7960!!
I've been digging around with some cisco engineers for about a week & I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: =============== == FROM CISCO == =============== Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if you
2007 May 31
1
linksys pap2 version2 ata DTMF issue
My asterisk box doesn't recognize DTMF from my analog phone, plugged into my ATA(linksys pap2 version2). I can make/receive calls fine... it's just that, for example, I cannot login to my asterisk voicemail. Softphones (such as x-lite) are fine. I've turned up a few articles via google where some people have this trouble, but have not seen suggestions on how to fix. I presume
2014 Sep 17
1
Polycom DND + Intercom/Paging Override?
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
2004 May 07
4
Cisco 7940 Phones as paging system?
Hi all; I have been searching for an answer to a question that a customer asked me and I have only found a few older answers. So, wanting to find out if anyone has any experience with this issue and can help provide me with some advice. I have a customer which is strongly interested in using Asterisk as a PBX. One of the core requirements, however, is that the system MUST be able to
2003 Feb 27
3
Intercom and Paging
What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson
2004 Sep 19
6
new ATA box for sale by Linksys
Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a "User Guide" meant for end-users that is very well written [no
2005 Mar 14
1
School design question
My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or
2011 Jun 14
1
Page() bumps user out of a call
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
2005 Feb 01
3
Linksys PAP2 / RT31P2 + multiple G.729 calls
Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the
2004 Aug 18
1
paging/intercom
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",
2007 Jan 27
1
How to fix error when paging
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 =>
2006 Nov 04
4
SPA3k wired to PAP2 for echo testing
In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with
2006 Feb 15
3
Fwd: Which ATA device do you recommend?
---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me
2007 Jan 07
1
snom 360 auto answer
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: