similar to: Beginner Questions

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

I don't have any experience of your problem - but I thought this might help. http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html <http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html> The UK (and some of it's former colonies) use a system called 3-wire ringing. Some equipment from overseas requires an adaptor to make it work. I don't know if the LineJack is one

(PLS, also CC: to 7ukwn@wanadoo.fr, which is my private address) Hi, I'm trying to setup 2 boxes with Asterisk; one with one quicknet PhoneCard, the other with two Quicknet cards: a PhoneCard and a LineJack; either boxes have a phone connected to the PhoneCards, a regular line is connected to the LineJack They'll be both on the internet (by adsl) and the purpose is to be able to make

Hi, I am trying to get a prototype working based on Asterisk and Quicknet cards. I currently have to systems set up each with a LineJack card. I have the systems working, but can't get the voicemail demo to work properly. Messages to the user telephone set from the voicemail system are clear, but recordings left through the phone are distorted. Sounds like the audio has lots of echo and

I have a Quicknet Linejack. Actually, I want to make a standalone(no internet connections) call to my asterisk box running GNU/Linux. When I call the system it says: -- Executing BackGround("Phone/phone0", "demo-congrats") in new stack NOTICE[14349]: File channel.c, Line 1212 (ast_set_write_format): Unable to find a path from 2 to 1 WARNING[14349]: File file.c, Line 553

I have a Quicknet Linejack in /dev/phone0. My phone.conf is: [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device => /dev/phone0 Only I can mark 7 digits, soon asterisk tries dial automatically. I cannot mark 8 or more digits. 6 or less digits work ok. -------------- next part -------------- An HTML attachment was

I'm looking at getting the Dev light applications from digium and I have 2 Createive Labs voip blasters. The voip blaster supports the G.723.1 codec. After looking at Gnome meeting it does not talk unless you have a quicknet card for it. Can I make calls using asterisk and the digium cards to the Voip blaster. ie. From PSTN to through the X100P through * into the Voip Blaster. (Voip Blaster

Hi folks, First off all, I'm new to asterisk. I'd like to make a basic installation of * with a Linejack, as recommended in this mailing list, I downloaded and setup successfuly the pwlib, openh323 lib, asterisk and chan_323. I have made little changes to default configuration files packed with asterisk and I managed to call with the telephone plugged in my Linejack a friend

Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple interfaces? The QuickNet Internet LineJack meets the description I believe, but it only has a single FXS or FXO. Are there any cards that have more than one FXS? Thanks. __ Dana Olson Disclaimer: The information transmitted in this message is intended only for the person or entity to which it

Hi all, I am still confused about the way to use asterisk with QuickNet Phonejack. If I am not wrong, The phonejack card should be using the phone.conf as the asterisk channel. I was initially confused with the ZAP channel (The digium card), now that I have found out that Phonejack should use the Linux Telephony Devices and its configuration file is phone.conf, but the question is I do not know

Hi there, I?ve been out for some months now, haven?t been checking the list at all. Does anyone know if the problem with the Quicknet Linejack (FXO) card dial out to PSTN with asterisk was solved? Is anybody working on it? Cheers, -Z -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this

Hi, I'm having problems with setting up my ISA LineJack card on Linux machine... I've done everithing according to documentation available, by compiling the new ixj driver (v1.2.1), loading it, adding device node /dev/phone0 with major number 100 and minor number 0, adding aliases into the /etc/modules.conf: alias char-major-100 phonedev alias char-major-100-0 ixj and running depmod

Anyone tested this card in Greece? I would like to buy several of these but I don't know if it works well with the PSTN in Greece. The Quicknet Linejack card seems to be unable to understand the telephone network's tones (busy, ringing,...). TIA, Michael.

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

Before you get too far.... The internet line jacks dont allow outbound calling. they cannot be used as trunk lines to the PSTN. the outbound code has not been written yet. I had to go buy FXO card from digium (that works much better than the Linejack) to get outbound calling to work Dave >>> fplandae@hotmail.com 4/11/2003 5:35:24 PM >>> Hi, I am a newbie. I have been

Hi, I have been experimenting with Asterisk and the linejack card and have discovered the follow: 1) The linejack in FXS mode works well with asterisk as long as the format in the phone.conf is set to ulaw. With mode set to slinear, gsm recorded messages played by voicemail appear to be slightly choppy (sounds like a click at the end of each frame). 2) Using the configuration above, the

I found a message from you to the asterisk users mailing list from 2001. I was wondering if you got (or still have) an asterisk system working with the linejack? If so, would you be willing to assist me with mine? I seem to have things working, and * says that caller ID is coming in, but I can't get * to actually answer the call. Thanks, Greg -- NetIO.org

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.