similar to: RE: Help! VoiceTronix Multi FXO/FXS Problem (Jacky)

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 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

Good day all I'm getting this error while trying to dial out on my asterisk server using a openline4 card "exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872" Please Help me

Hi All, Just a quick question, but I could really use some help on this one. I've got the CVS-Head of * installed and running, and am using a VoiceTronix OpenSwitch12 to connect to 12 analouge lines. I've got callerid activated by the Telco, and can get callerid using a std phone. However, using *, I always get an error 'Cannot decode callerid'. Does anyone know if I need to

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,

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn

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

I'm using a Voicetronix OpenSwitch12 with Asterisk. I need the ability to hook-flash a channel while a caller is on the line so I can transfer a call through a PBX. I need the equivalent of the FLASH() application, which only works with Zap channels. Does anyone know how I can hook-flash then send DTMF on a vpb channel? Rodney Hassell -------------- next part -------------- An HTML

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

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

>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?

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:

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

Hello all - I'm trying to get Asterisk working with an Openswitch/6 card with all ports set to FXO for PSTN in/out, and 10 SIP phones on the LAN with the Asterisk server. The SIP phones are working properly to dial one another, but I can't seem to get a call to go out to the PSTN through the voicetronix, and I similarly can't get incoming calls to the lines hooked up to the

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

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

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"

Is it possible that cheap phones (Budgetone) cause echo's? I had a digium X100p and i managed to get rid of all the echo problems i was having. Recently i got a Voicetronix Openswitch12, and getting terrible noises when i use an IP phone (budgetone) to call analog phones or PSTN. I have tried all the possible things (rxgain, txgain, echocancel, i even changed the codec to g711) Is it

Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are