similar to: Newbie Questions . . .

A couple more data points. Current llvm 3.7svn with the two outstanding OPENMP patches can build the openmp support in gdl 0.9.5 (which completely passes its test suite) and apbs 1.4.1's limited openmp support. On Sat, May 2, 2015 at 11:11 PM, Jack Howarth < howarth.mailing.lists at gmail.com> wrote: > On a positive note, current llvm 3.7svn with the two outstanding > OPENMP

Hi all, i have installed two digium cards on my asterisk box a TDM04B & TDM22B. The channels are configured as show below: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06:

Hi Jeremy, 2013/9/13 Jeremy Bell <bell.jeremy at gmail.com>: > I looked into the archives as far back as 2009 and searched around for more > information about vmkit, but I still have some questions. > > First of all, what is that status of VMKit? Is there any active development? > A roadmap? Is it in maintenance mode? The development is inactive since two month because the

Hi I am interested in setting up an Asterisk PBX in my office with digium hardware, and I just have a few questions in regards to what I would need. It is my understanding that an FXO card is used to interface with an incoming/outgoing phone line, and an FXS card is used for interfacing with a phone within the system. Currently we have 4 incoming/outgoing phone lines and would like to have

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

Hello, I am having problems configuring a TDM400P 4 Port FXO card with groundstart signalling. The box has 2 X100P's, 1 TDM04B and 1 TDM40B. I can configure it for loopstart and it works, just not groundstart, which I need for this installation. What am I missing? Thanks! Mark /etc/zaptel.conf #two X100P FXO Cards # fxsls=1-2 # TDM400P 4 Port FXO (TDM04B) fxsgs=3-6 # TDM400P 4 Port

How are you? Fristly I would like to introduce myself in a shortly. I'm a newcomer at Newcomtech Co,. Ltd. Nowdays I'm working at TDM40B and TDM04B cards. I have installed Linux hedhat 9.0 and Asterisk software. I have configured the cards. I typed asterisk -vvvc command in the command line and then asterisk give us "asterisk is ready". Now I have to configure my asterisk

OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can receive and place calls with no issues, however, when I receive a call, the CID only shows "Analog Line" on the Grandstream 2000XP phone. Does anyone have any ideas even where to look to change this?? Is it a setting in the phone, Asterisk, or both?? Thanks, Marc

Hi: I have one TDM40B and one TDM04B on my Asterisk box. Both were working fine. Then, all of the FXS ports started to make echo sounds when I make FXS to IAX or SIP connection. All of the FXS ports fail to make bridging to the FXO channels. And when I try to make a call from the console to the FXO ports, I only hear static noise. Any suggestions about the source of the sudden change in

Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup

I've installed a TDM04B and a TDM40B. I haven't plugged any lines into them yet but I'm starting to see this in my logs... [root@webster asterisk]# grep alarm /var/log/messages Sep 20 09:13:22 webster kernel: Power alarm on module 1, resetting! Sep 22 11:07:07 webster kernel: Power alarm on module 1, resetting! Sep 22 16:10:55 webster kernel: Power alarm on module 1, resetting! Sep

I looked into the archives as far back as 2009 and searched around for more information about vmkit, but I still have some questions. First of all, what is that status of VMKit? Is there any active development? A roadmap? Is it in maintenance mode? Secondly, can VMKit generate binaries that can be used on android/jni? Finally, I understand that the .net/CLI support is no longer being

Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _____________________ Don't PANIC

Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks.

I'm debugging a TxFax problem whereby the fax transmission fails. I suspect interrupt latency--some interrupt routine is holding its interrupt too long. I have all unnecessary services switched off and X is not running when I perform these tests. Some transmission are successful while others fail at random points. I've noticed that after I boot Linux, load zaptel, wcfxo, and wcfxs,

I need to reopen this discussion because it's impossible to run spandsp (and VoIP) under these circumstances. With zaptel unloaded, I see the following "vmstat 1" output: no swapping, an occasional disk output, +/- 1003 interrupts/sec., less than 10 context switches/sec., CPU idle 100%. A very quiet system. I load modules zaptel and wcfxo, and the system utilization stays the

I have TDM40B, TDM04B cards, 4 analog and digital phones. First I want to use 4 analog phones with my TDM40B card. I would like to dial between 4 analog phones. The dialing numbers for 4 analog phone will be 800,801,802 and 803. These are my conf files. /etc/zaptel.conf fxsks=1-4 fxoks=5-8 loadzone = us defaultzone=us ;;;;;;;;;;; /etc/asterisk/zapata.conf [channels] relaxdtmf=yes

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

Hello List, I'm from Germany and I want to use a Asterisk System. I have a few Accounts at my SIP-Provider www.sipgate.de and now I want to use my ISDN-Phone on the Sip-System. My idea was i set up a Asterisk-System and i will put in an ISDN Card where I can plug a ISDN Phone, I will have to use an ISDN card with the NT-Mode. The Asterisk has to register is at the SIP Provider and if a Call

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed