similar to: troubles working with Voicetronix Openswitch12

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

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 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"

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

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --

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,

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 we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

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 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

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

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

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

Hi all, Trying to get fax reception going using a voicetronix openline4 card, however, there are two issues (as far as I can see) (1) Currently the voicetronix world (cards, firmware, drivers, channel prog etc.) does not do fax tone detection. I have spoken to the voicetronix guys , understand what has to happen there and may do some hacking if I can resolve the second issue. Meanwhile, I

Any guidance on how to get my openline4 to get recognized by asterisk@home? I've got my vpb drivers running, but not sure how to add it as a trunk, should it be via zap? or is there another way? Thanks, Chuck

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

Hi list, anyone know what is going on here? I don't get any sound from the out clip & I get the following when I dial in to asterisk (after which it just times out): -- Event [0=>[06] Ring] on vpb/1-7 -- Executing Wait("vpb/1-7", "2") in new stack Read_channel ## vpb/1-7: Setting record mode, bridge = 0 -- Executing Answer("vpb/1-7",

>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?

I am really struggling getting Asterisk up and running utilizing these two cards. Has anyone had success with this combination? If so, would you be ever so kind to submit sample vpb.conf,zaptel.conf,zapata.conf and extension.conf files??? With great thanks, Ronald R. McDaniel Southern Computer Services, Inc. rmcdaniel@southerncomp.net (251) 444-3136 office (251) 446-3137 fax (251)