similar to: Voicetronix Openswitch/6

Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it

Does anyone have trouble with dialing in to an Asterisk Server and having the DTMF digits recognized? We have some clients who are calling in with cell phones, notably those with SprintPCS service, who's DTMF is just never recognized. I have tried relax_dtmf on and off, with no improvement. My rxgain is currently set to 3. Can anyone suggest possible solutions? Incoming calls are

Good people, Please find attached a first shot at a driver for the Voicetronix OpenSwitch hardware. It does work (ish). After paying $900-odd for it I figured I might try to make it work with Asterisk (could never get enough documentation on Bayonne!) I have problems with the audio quality, and since am new to Asterisk, I could really do with some input. First is that because the

Fails on loading several of the chan_*.so modules with "undefined symbol __use_ast_pthread_create_instead__". Notably, these same modules complain during compilation "implicit declaration of function __use_ast_pthread_create_instead__". Ideas? -- Alok K. Dhir <adhir@symplicity.com> Symplicity Corporation http://solutions.symplicity.com 703 351 6987 (w) | 703

Seems strange to me that I should be having this problem but I'm stumped. I'm still playing with a Voicetronix OpenSwitch6 card and Asterisk (development CVS). I using a dialplan that's derived from one on another running installation that's using Digium FXS/FXO interfaces instead of the OpenSwitch card. It seems that as soon as a user on an FXS interface dials 7 digits, a call

Hey all - we have two TDM400P cards in an SMP Redhat9 box, with 4 FXO ports each running thel latest Asterisk CVS. Users connect to this to share POTS lines using analog phones connected to Sipura SPA-2000 boxes. All works reasonably well, except every day (or more) a line will get "locked up". No other way to describe it really -- the line just stops working, so if someone tries

Hello, Hacker I install VoiceTronix OpenSwitch 12 port PCI Telephone Card, and setting vpb.conf, extensions.conf following My problem is: When i dial to fxo(channel 9-12), it is ok, but when i continue press exten 102, the channel crach with error messages following exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 Do i ignore some setting for VoiceTronix OpenSwitch12

Hello, I would like to know if voicetronix card (specially openswitch6 and 12) can be used with asterisk. Is there any driver for this card? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

Attached is the updated driver, thanks to Mark's advice. It now does pretty good audio, although when you dial another channel via Asterisk you get some echo. This I have not been able to solve quite yet (using voicetronix echo cancelling doesn't seem to do it). If anyone can figure it out, would appreciate. Thanks P. Ps. Unpack inside asterisk code directory. Replaces your makefile

Skipped content of type multipart/alternative-------------- next part -------------- [root@LINUXVM root]# asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= == Parsing

Hi all, I think I've got the V4PCI card working with asterisk (can't tell for sure, can't dial out since the card doesn't appear to understand 2 or 3 dtmf codes, if you've seen that before, let me know). Has anyone else had problems/successes with those cards before though? Thanks, Andrew Griffiths

I'm working on putting together some Ideas about using Asterisk in our environment, one of the things I want to consider is DID trunks (analog), what hardware do I need to terminate these trunks? I'm looking at the voicetronix openswitch6 or openswitch12. On the openswitch, I'd like to use some of the lines for analog sets for the breakroom, kitchen, etc where they don't need all

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Hi. I need help with my brand new Voicetronix OpenLine4 board that I installed into Asterisk. After building the Linux device driver and inserting the module, I modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the US settings and comment out the Austrailian ones. I made the appropriate entries for routing in vpb.conf and extensions.conf.... All appears to be well, except

One of the suggestions I got for dealing with my analog FXO woes was to look into using a Voicetronix card instead of the Digium. They're more expensive but if it worked better it may be reasonable. They indicated they were compatable with the CVS HEAD version of *. Can someone share with me their experiences with Voicetronix OpenLine and/or OpenSwitch cards with *? I'm specifically

In my search for a better analog FXO interface, I'm looking into the Voicetronix cards. Can someone let me know if they've been using the OpenSwitch cards with Asterisk? Also, can anybody comment on the differences in FXO performance between the OpenCall and OpenSwitch cards? Thanks in advance, Paul -- Paul A. Dugas Dugas Enterprises, LLC email:

hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a "transfer"

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I normally don't write emails like this, but the last couple of days have been very frustrating. We built a system for a customer with the following hardware: Asus Vintage (sis chipset) P4 2.8 512M Ram LSI SATA Raid card (2x80g drives) Digium TDM400 we had originally purchased a tdm2400 with echo cancellation but couldn't fit it in the

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

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