similar to: zulty's ZIP 2 IP phone

Does asterisk support the intel 537/md3200 chipset? I don't want to start any flames here, I know all about using generic crap in asterisk,[*] which I really don't approve of other than for testing, but I have a customer demanding a generic chipset for his one backup analog line. He will not spend the money for a Digium card and says he will find another company if I can not provide a

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

Is there something special that needs to be done to allow a T100P/T400 to respond to a remote loop request? I am having some issues with a Point to point T1 line using zaptel ppp. This line gave us small problems when we had a pair of Cisco 2600's on either end but now with the zaptel ppp it is going down every couple of minutes for 15 intervals. The phone company says the line is good to

I had a hard time coming up with a subject. I have some SP IP500s that I'm using with *. When I access voicemail or get the busy announcement, I'm only hearing the last half of the first word or coming in on the second word. As example, when accessing voicemail, I hear ' have 1 new message.' It's like * is starting to play the announcement before opening the channel. This is

Anybody have a guide to the Cisco 79xx phones? One that I can give the 7 or 8 ppl in my office so that they can stop asking me questions. I was going to type up a basic guide but then decided I don't want to reinvent the wheel, one of you may already have one. I tried to use Cisco's guide but it's for their own protocol, a lot of options are different or rearranged. I need a basic

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

I'm trying to get a Zultys Zip 2 phone working with Asterisk. The phone seems to be failing registration (see sip debug output below). However, I can place calls TO the Zip2 from other SIP phones (Grandstream BT-101, Xten X-Lite, and eStara Softphone) and from Nortel PBX extensions coming in to Asterisk over a PRI T1. The problem is that I cannot dial any extensions from the Zip 2. Any

> Message: 1 > Date: Fri, 21 Jan 2005 17:38:27 -0600 > From: "Henry Devito" <hdevito@qwest.net> > Subject: [Asterisk-Users] SPA-2000 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <000d01c50012$4ea49f30$4300000a@homeacxa7jw2xn> > Content-Type: text/plain; format=flowed;

I have my first Zultys here but can't seem to get to first base. Does anyone have a basic configuration file they might share? Mine is a Zip 2+. If I understand the Zultys User Manual, the default passwd should be a null string--but the web interface isn't accepting that (nor 'admin' nor a couple others). Anyone know what the default passwd actually is? Thanks for any and all

Now that most of you have worked overtime to show why most people are continually pissed at Nix Users (all except two of course). The problem I can see is the downright technosnobbery involved. There is nothing wrong with Linux. I play around with RH9 and FreeBSD and find that most things run fine. But you get into a problem where it keeps asking for the same blamed libraries over and over on

Hi peoples Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they have configured one before for iax. If you have a sample config file that would be great. Any assistance would be nice Scott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/6c243053/attachment.htm

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

I have had the opportunity to test many IP phones in the last 6 months and I thought you might enjoy a quick review of what I have found. Grandstream Budgtone 200 - Poor Quality for business use- Looks good, and the handset feels nice, buttons have a decent feel, but the disply is difficult to read when you are not directly over head of the phone, plus the sound quality of the handset, and

I currently have a Toshiba Strata DK424 with a Stratagy voicemail system (4 ports). I am looking to go from having a receptionist answering the phone to an automated attendant. It appears that Asterisk can be the solution, but I have some questions. Do I just replace the Stratagy with the Asterisk or do I need to reconfigure the routing of the phone lines? Currently, the POTS lines come in to

Nadav, We see multiple regressions after r172868 in ISPC compiler (based on LLVM optimizer). The regressions are due to spill/reloads, which are due to increase register pressure. This matches Zach's analysis. We've filed bug 17285 for this problem. Is there any possibility to avoid splitting in case of multiple loads going together? Dmitry. On Wed, Jul 10, 2013 at 1:12 PM, Zach

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

I've narrowed this down to a single kernel (kernel.ll), which does a fixed-size matrix-matrix multiply: # ~/llvm-32-final/bin/llc kernel.ll -o kernel32.s # ~/llvm-33-final/bin/llc kernel.ll -o kernel33.s # ~/llvm-32-final/bin/clang++ harness.cpp kernel32.s -o harness32 # ~/llvm-32-final/bin/clang++ harness.cpp kernel33.s -o harness33 # time ./harness32 real 0m0.584s user 0m0.581s sys 0m0.001s

I have searched both the wiki and googled looking for a solution to a square key configuration. I need to have C.O. lines to appear on the buttons to facilitate a small office. All of the users can see each other and calls are put on hold and picked up by the other users instead of transferred. Has anyone done this? Can it be accomplished and how is it accomplished? Thanks in advance.

I tried this with a fresh tree, so I know it works. Based on your last message, I think you may have missed step 1. My needs are for both CBMysql and MeetMe2, so I bundled the patch file. If you are only interested in one, then delete the sections of the patch that are not needed. 1. Download the source for MeetMe2.c from http://www.areski.net/asterisk-meetme/about.php?s=0 and put it in