similar to: Voicetronix Tones

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"

Any one has documented how-tos for making voicetronix openline 4 to work with Asterisk. I have been contacting Australian Digium resellers and Digium cards are not approved in Australia. So I suppose Australian users are interested into putting Voicetronix in use. Any expereience to share will be most appreciated. David Kwok -------------- next part -------------- A non-text attachment was

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

Ron, VoiceTronix worked with me last week and backported the new chan_vpb that's in the current HEAD CVS tree to the 1.0 stable branch. So if you are using 0.7.2 or 0.9.0 you can go to the VoiceTronix website now (note that this is very new) and download an Asterisk package. When you download it there is an new chan_vpb.c that needs to be put in your asterisk source code tree. Also there

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

Hiya, here is a patch to fix that: [root@mailmx2 channels]# cvs diff chan_vpb.c Index: chan_vpb.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_vpb.c,v retrieving revision 1.9 diff -r1.9 chan_vpb.c 100,102c100,102 < static VPB_TONE Dialtone = {440, 440, 440, 0, 0, 0, 5000, 0 }; < static VPB_TONE Busytone = {440,

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

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

>From what i've read i think that asterisk is suppose to generate the dialtone. But when i pickup my analog phone it seems that the Openswitch12 card is the one playng the dialtone. Im almost positive the card is playing the dialtone, can anyone confirm this? And if so how can i change the dialtone?

When I dial a PSTN (via * with Digium Quad-E1) number, I hear two sumultaneous ring tones. One is coming from *, the other I assume is from the ATA. Is there an intelligent way to get around this?

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

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

Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE

Hi all, Just thought I'd pass along some pointers when outdialing with Voicetronix's OpenLine4 card. I was having a tough time dialing out from *, it probably has something to do with chan_vpb.c not waiting to hear the dialtone before telling the card to dial. A quick fix was to insert a "," in the dialstring telling the card to pause before dialing. However when the

Hi there, Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.

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

Hi, Firstly it appears you are not passing the extension to be dialed to the Dial command, so the driver is dialing badly. It should be something like dial(vpb/1-2/102) since this is a line port. Admittedly the driver needs to check better that the number it is trying to dial is legit but it doesn't :) P. --- You Wrote --- From: "Jacky" <jacky.taiwan@msa.hinet.net>

Hi, I am using a voicetronix OpenLine4. I downloaded a recent asterisk CVS from voicetronix webpage but have had no luck to reduce echo on outgoing calls and for it not to crach on incoming calls. I dont think both problems are related though. Here is an output of what happens when a new call comes in and my voicetronix tries to pick it up and crashes asterisk: > vpb/1-1: Event

Is it possible after dialling a trunk prefix to wait for dialtone (or wait 2 seconds) before continuing to dial the outgoing number ? I need to "hunt" for one of several lines, and when one successfully provides dialtone, to continue dialling.. A simple NO will do, but if there's a workaround, I have not found it. -------------- next part -------------- An HTML attachment was