similar to: VPB Driver Question

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 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

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",

So I've got a Voicetronix card and it looks like the kernel driver works. Other than the 0's for ID info. vpb: Driver Version = 4.0 vpb: major = 251 vpb: tmp [0xfc8fec00] dev->res3 [0xfc8fec00] vpb: tmp [0xfc8c0000] dev->res2 [0xfc8c0000] vpb: 1WS Write cycle vpb: Manufactured 00/00/0000 vpb: Card version 00.00 vpb: Serial number 00000000 vpb: Setting up udev... vpb:1 V4PCI's

HI All I've tried to do Call pickup /pickup group using VPB channel. I configure the same way as in zapata.conf ( callgroup=1 pickupgroup=1). The result is, the phone can only be picked up by other SIP phone under the same group. If we called that SIP phone and tried to pick up using VPB channel, it cannot pick up the call. Is anyone has the solution? Thanks.

Hello, list, I am using Asterisk with an OpenLine4 card. It worked well with Asterisk 1.0. Then I upgraded the system, Asterisk 1.4 had some problem to compile chan_vpb, but I managed to compile it manually. Still Asterisk does not work because it refused to load chan_vpb module. I had to revert back to install Asterisk 1.2. Then I get a error: asterisk: hip.cpp:793:

#I would like to use the below data to make a scatter plot matrix with #code similar to that below the data #conceptually this is the right approach I think #thanks in advance melt.gg <- structure(list(stream = c("Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower",

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

Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why please Help Altus

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

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

I am a serious Asterisk newbie: just installed asterisk last week and it is now running with our Voicetronix OpenLine4 hardware. All is working as expected with one exception, in the following sequence (extracted from my extensions.conf file): [GetConfirmation] exten => s,n,SetVar(TimeOut=0) ; if timeout and TimeOut=1 then user already timed out once, so hangup exten =>

Hi, I am fairly new to asterisk. I am currently testing my first setup. I've been able to debug most of the problems to make asterisk work with my hardware setup until this time. Currently I have the following issue: Voicemail is running but when I test to leave a voicemail thru my incoming PSTN channel (voicetronix / vpb), asterisk will not detect sound (according to the log) on that

Hello all, We had an experimental system which works on OpenLine4 telephony card and Asterisk 1.0.9. Customer asked to upgrade Asterisk to 1.4, then we found our problem: At first Asterisk 1.4 does not compile chan_vpb.so. The problem is it tries to compile chan_vpb.cpp to chan_vpb.o and chan_vpb.oo, then try to link them together. I manually compiled it, then make went smoothly. But it

Good day all I had a look at the extensions.conf sorting http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting What I'm trying to do is route all my cellphone number threw a channel and all other calls threw the other 3 channels Cellphone numbers are 10 number,i.o.w XXXXXXXXXX. This is what I tried but it doesn't seem to work,please help Thanks Altus extensions.conf

Hi list, I am having some difficulty implementing a certain dialplan where the following happens. If the first Dial() is not answered, I want to play a small greeting then ask the caller to either hold the line (try calling again) or press 1 to leave voicemail. exten => s,1,Dial(${BLAH},10,Tt) ; Dial 10 sec exten => s,2,Answer exten => s,3,Playback(greeting)

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

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 --

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. 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