similar to: problem with A400P01 OpenVox

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]:

hello, I have the A400P01 from OpenVox. Is necesary to install all this packages? http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz or just with asterisk and zaptel is enough. thanks a lot -------------- next part -------------- An HTML attachment was

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 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.

how can I fix this just started ...... Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid

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.

'ello Markus/folks@, I downloaded 3.0.2 today and went and looked at openbsd.html and found some errors. Patch attached. Patch for http://www.openssh.com/openbsd.html Hope that helps, Rgds -- Kevin Sindhu <kevin at tgivan dot com> Systems Engineer TGI Technologies Inc. 107 E 3rd Avenue Tel: (604) 872-6676 Ext 321 Vancouver V5T 1C7 Fax: (604)

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

Hello everybody, I have installed the Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new

Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] 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

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

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up "from" my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking

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