similar to: Quicknet Internet Phonecard

I have just finished installing all the pieces of Redhat Linux 9 (2.4.20-8), Asterisk-0.9.1, Gnophone-0.2.4 on my IBM T30. Audio card is SoundMAX Integrated Digital Audio. Not sure what the chips are. But everything works on both W/XP and RH9. (Machine is obviously dual boot) Everything starts, Asterisk is up, sound/audio is great for CD player, Volume controls, Voice recorder, RealPlayer, etc.

I have an Internet PhoneCard for my IBM T30 running Linux RH9 2.4.20-31.9 and Asterisk 0.9.0. I have installed the appropriate Openh323 drivers. The card works and I can call to an extension that plays a recorded message, hang-up, call again and so forth. Suddenly, the subject error message starts appearing in the log, something goes into a loop, everything else in the system slows to a

Hi list, I have some problems to get the QuickNet Internet PhoneJack working. What .conf files do I have to edit to get a dialtone for the first test with the standard configs from asterisk? I have the ixj driver running and a cat /proc/ixj after asterisk start tells me one reader and one writer. But if I pick up the receiver I don't get a dialtone nor I'm able to dial a number.

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

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

I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called 612@fwdpulvercom No path to translate from SIP/fwdpulvercom-dd5a(2) to

If you have any quicknet cards you are not using, I may be interested in them. I'll discuss terms after I know what you have. For sake of high shipping costs, I'm not interested in overseas shipments to the United States (where I live). My best resources will be for those that have PCI PhoneJacks, or PCMCIA CardJacks. Please reply to my email address, and not the mailing list.

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

Hi, all. On a lark, I have gotten Asterisk 1.0.2 (from Debian testing) up and running, but I have found one problem when I try to use it with Free World Dialup: Dialing doesn't work properly. I have everything else working; it receives calls, ringing my phone and everything, but when I try to dial using the configs found on FWD's site for Asterisk at this URL:

(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

waited for the aroma of burnt electronics. There wasn't any so I went to the next room and plugged in a telephone on the wall outlet. I picked up the handset and it rang my asterisk box. I dont have any phone service. The phone is working fine and have used it for hours listening to podcasts. And I can plug it to any phone in the house. Now I am thinking if I can plug another phone

Greetings: We have been running a CVS HEAD version of asterisk from Mar 10, 2005 on ix86 (PIII-600) Linux 2.4.27 with ixj (chan_phone) hardware. In a hope of getting better 'chan_skinny' support (to attempt using a Cisco 7920 IP phone) I built asterisk 1.2.40 on this box. Initial tests verify that our previous dialplan is working (iax2 trunks, register sip phones, registering withour SER

Is there an interface (direct or indirect) in Asterisk that can be used by JTAPI to do third party call control and the other functionality supported by JTAPI? Does anyone have an example of such a thing? Jim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040528/0a688746/attachment.htm

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

Hi. I've found a large bug in Speex Codec implementation in Openh323 code. I've signalled this bug in the Openh323 site, in the section "Report bugs". THE BUG NUMBER IS 143! Everybody can verify that the report is arrived monday 7 July!!!!! But no one has analyze this bug....!!!! So the result is: Openh323 implements Speex Codec that runs with wrong bit rates and that

Hi, I have recently discovered the project along with the PhoneJacks produced by quicknet, they could be the answer to something I have been looking into. I would like to be able to test using a dial-in server & possibily also a Windows RAS server, however I only have 1 phone line. I was thinking that I create a setup like that illustrated below to solve the problem:- Ext 1000

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _________________________________________________________________ Expand your wine savvy — and get some great new recipes — at MSN Wine.