similar to: A400P01 from OpenVox

Hello Has someone used the OpenVox A400P01 (ie. a supposedly Digium-compatible A400P board with a single FXO module www.openvox.com.cn/products_detail.php?genre_id=9&id=28) successfully? I've put it in an older PC with a Gigabyte GA-7ZX motherboard, then a more recent PC with an Asrock K8NF4G-SATA2: dmesg returns nothing :-/ Is there something specific that needs to be done in either

Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line connected. I am new in Linux and Asterisk, my steps are theese: 1. Install CentOS 4.4 (basic instalation). 2. Command line: yum -y update yum install gcc kernel-devel bison openssl-devel yum install openssl-devel 3. Download the source: wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz

Hello, I'm thinking of ordering an OpenVox A400P01 (A400P + 1 PORT FXO Bundle) for use in a old IBM 8364 thin client: http://www.openvox.com.cn/products_detail.php?genre_id=9&id=28 http://silicon-verl.de/home/flo/software/netstation-8364/ Has someone already used this hardware with Asterisk, especially on a small piece of hardware like this, and could offer some feedback? Thank you.

Hello If someone had the opportunity of trying those two analog cards, how do they compare? Digium's sells for $150 while OpenVox's sells for $95. Thanks.

hello friends, I am configurin my Billion ISDN and when I start asterisk (asterisk -vvvc) I have this error message: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 29 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line:

Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

hello friends, is there any way to simplify that extensions.conf file? [miprimerejemplo] exten => 20000,1,Dial(SIP/20000,30,Ttm) exten => 20000,2,Hangup exten => 20000,102,Voicemail(20000) exten => 20000,103,Hangup exten => 20100,1,Dial(SIP/20100,30,Ttm) exten => 20100,2,Hangup exten => 20100,102,Voicemail(20100) exten => 20100,103,Hangup exten =>

Hello I'm looking at my options to build a compact, silent, headless Asterisk server to handle one or two FXO ports. Out of curiosity, I got one of those babies on eBay for 20E: http://silicon-verl.de/home/flo/software/netstation-8364/ Before I spend time on this, can someone tell me if... 1. it's easy to install Linux on those things (am willing to add a CF-to-IDE adapter to use a

hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/86e4ad95/attachment.htm

Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

Hi I am looking for a source for an OpenVox card. Has anyone purchased through http://www.voiplink.com or do you normally use another vendor or OpenVox.cn directly? thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090705/6d90b996/attachment.htm

Hi, This is tech-support from OpenVox, would you mind to send email to tim.june at openvox.cn for more details about G400P issue? Or contact me via IM below for better communication. Regards, MSN: tim.june at msn.cn Gtalk: tim.june666 at gmail.com Skype: tim.jjune OpenVox Communication Co. Ltd. Quick Support: http://wiki.openvox.cn/index.php/OpenVox_Quick_Support

Hi i need to install a server with this hardware: 1 OpenVox B800P 1 OpenVox A800P01 4 OpenVox FXS-100 FXS100 4 OctWare SoftEcho SOFTECHO Do you suggest 1.2 or 1.4 branch? Is now 1.4 stable ? I've tried 1.4 the last year but i've experienced many problems with misdn drivers. Thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 > >> - downloaded the file opvxa1200.c > >> - copied in zaptel-1.4.7.1/ > >> - edited makefile adding opvxa1200 in the modules and the voice > >> opvxa1200.o : zaptel.h wctdm.h > >> - edited zaptel.sysconfig adding MODULES="$MODULES

Hello all- My client has purchased these two OpenVox cards and I'm configuring a system with Asterisk 1.6. In the past I have used bristuff and libpri with older versions of Asterisk, but now I would like to upgrade to Asterisk 1.6. Question, should I be using mISDN or libpri for these cards when they are in the same system, or does DAHDI now support both cards under asterisk 1.6

I installed a new machine with CentOS 5.2, Zaptel 1.4.11 and Asterisk 1.4.21.2 and an OpenVox A1200P card. This card has its own driver and Zaptel has been patched to use it. The problem is that from the moment I load Zaptel I get this messages on the console: buffer re-sync occur from 2 to 1 buffer re-sync occur from 1 to 0 buffer re-sync occur from 0 to 1 buffer re-sync occur from 1 to 0

B.H. Hello, all :-) We have some OPENVOX D410P PRI cards and we are successfully using them with Asterisk boxes which are based on stock ubuntu 12.04 DAHDI and Asterisk packages. The card is recognized by DAHDI as 'Wildcard TE410P (2nd Gen)' and it uses wct4xxp driver. Now, i'm trying to run this hardware with DAHDI 2.6.2 package which is available from asterisk.org site and looks

What is the difference between these to cards? Any feed back good or bad would be great. Jonn