similar to: Asterisk, VoiceTronix & UK Caller ID

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

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 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

#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",

>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?

Hi, I hope some can help me out, as I have a problem with the vpb driver in *, and I am hoping someone has hit these issues and can offer some advice. First off, I'm connected to a British Telecom SystemX exhange with the local Tel Co. I can ring out fine, but when the callee on POTS answers, the vpb driver detects a loop drop and immediantly disconnects the call. I have currently set

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>

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

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

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the

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

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 would like to have some more informations on asterisk, I want to setup a linux based pbx and asterisk seems to be the best solution, I have some questions for configuration: 1) I have a PRI, so I must buy a digium card to interface with PRI, right? 2) If I connect an ethernet card from the pc (equipped with a digium card connectd to the PRI) to a switch I can connect users to this

Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the "Qs about FXO/FXS cards" thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi

Hi all, I'm new to the list, and I wonder if someone could offer some advice please or point me to a previous thread about this in the archives? I work for a Radio station on the Costa Del Sol in Spain, and currently we're using DarkIce to take an audio feed (over USB) from our desk into DarkIce and then sending up to a hosted IceCast2 server. This works perfectly, and I have a cronjob

I have 2 problems with openswitch12: 1) I can not make work "ignorepat => 9" i do not get dialtone after the number is dialed, the system ignore the number and i can go on dialing the rest of the number.... but when i want to take the line teh dialtone do not stay. 2) when i tray to leave a message on the voicemail of an user i get the following error Sep 3 17:04:55

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

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

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