similar to: Panasonic KXTD 1232 6

Hi, Anybody have an idea how to integrate * with a Panasonic td-1232? We one at the main office, and are installing * in a branch office. We'd like to be able to make calls from * extensions to Panasonic extensions and the other way around. Making outgoing calls from extensions one one side to lines on the other would be nice too. I can put another * machine at the main office, but what is

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Can anyone point me to a possible source of a linux cups driver specifically for this machine or to a work-alike driver for a similar piece of equipment. Failing this, does anyone have any suggestions as to any open source software guys who might be willing to develop one for a fee? Regards, P.S. I have already googled, yahooed and sniveled on other venues in search of an answer. -- ***

We have a setup running Asterisk interconnected to a Panasonic TDA200. The Asterisk server has a two port E1 card, one connected to the phone company and the other to the Panasonic. Everything is running fine and we can send and receive calls from the Panasonic and phone company. We are using MFC/R2 for both links on Asterisk 1.4.4 and Zaptel 1.4.3. The only detail we have is that we cannot

I have a Panasonic Video Camera and need to install the software [Quick Movie Magic and Motion DV Studio {plus the USB driver for the camera}] so I can actually edit my movies on an Ubuntu system [10.04] I have tried to install it through WINE with no success as my system informs me that the software is read only and from an untrusted source. Attempting to switch the executable bit through the

Hello! I have following connections over isdn pri: avaya definity---pri--asterisk--pri-panasonic 500 Just because panasonic 500 can't send user's names. I also want to have reverse callerid for avaya users. But if there is no answer in dial plan: exten => _XXXX,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})}) ;exten => _XXXX,n,Answer exten => _XXXX,n,Dial(DAHDI/g4/${EXTEN})

I'd like to have my Asterisk system pick up a certain extension on an existing Panasonic PBX when it rings. (It's connected to some proprietary Panasonic doorphones that I haven't replaced yet.) I connected that extension to an FXO port on a Digium AEX410 card, and set that channel to have the context "doorphone". The problem is that the extension is never executed. With

Hello all, did anybody try to use the application under Wine? I need help in reproducing and researching the bug. The program requires modem and I don't have one. For details see: http://bugs.codeweavers.com/show_bug.cgi?id=411 The page has description how to reproduce and gather information about the issue. Please, add your comments to the bug. In general - you are welcome to submit

I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia

Hi all, There is someone with some experience interconnecting a Panasonic digital PBX (KX-TD1232) with Asterisk? Thank you and best regards, Dan

Hi, at the university department we have quite old Panasonic 2+6 PBX (1BRI + 6 POTS for outputs) and 25 local analog extensions. We would like to add Asterisk with 1 fresh BRI line and possibly integrate with existing equipment (we would like to crossover between both pbxses). What would be most efficient way to do this ? Thanks in advance, regards, Rob.

Hi, all ! I have a situation like this: [SIP Terminals] <-> [*] < -ISDN-PRI-> [Panasonic D500] <-> Telecom (conn to Telecom is with second PRI card in Panasonic and 16 POTS lines). Panasonic has 2 ISDN PRI cards (one to Telco, and second to Asterisk), 16 POTS lines to telco and 32 (advanced hybrid telephone type) extensions. Idea is to have possibility to have some users on

Hello, We use Asterisk with PAP2 and today we connected the FXS ports of PAP2 to CO ports of our Panasonic KX-TD1232. Problem is that Panasonic doesn't ring - that is doesn't ring every time the PAP2 is ringing. When we reset either Asterisk or the PAP2 it usually rings, but after couple of minutes it stops and only the automatic operator is answering - after 2 rings. We tried changing

Thank you for response. So a "standard" quad bri card can serve the KX-TD1232, no need for anyt special fxs similar equipment. Would like not to invest much more money into the Panasonic, I wan't it out :P BR Amund -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Peter Svensson Sendt: 14. juni

> Date: Thu, 23 Jun 2005 08:50:50 +0200 > From: "Robert Rozman" <rozman@fri.uni-lj.si> > Subject: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * > - Euroisdn Italy > > I'm pulling my hair down and getting bold :-) ..... I have Asterisk between > Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff > Asterisk).... Plenty of

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I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not familiar with that phone. Is it single pair? -----Original Message----- From: broadbandvoice@comcast.net [mailto:broadbandvoice@comcast.net] Sent: Sat 4/22/2006 12:13 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3 I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work

Hi! I want to make a call to/from Panasonic IP pbx thru asterisk via H323. H323 is working nice, I can call/receive using netmeeting -- Executing Dial("OH323/WindowsUser@195.184.x.x-305e28cf", "OH323/11@80.98.x.x|15|tr") in new stack -- H.323 call to 11@80.98.188.159 with codec(s) alaw -- Outbound H.323 call to destination '11@80.98.x.x', channel

Sorry someone screwing with permissions on my server bounced the 2 days worth of email after I posted this, any and all those lovely people who replied with suggestions from my post could you sent them again :-) James -----Original Message----- From: James Bean On Behalf Of Asterisk Mailing List Sent: Tuesday, 30 May 2006 12:54 PM To: 'Asterisk Users Mailing List - Non-Commercial