similar to: Should configured devices show up with "show channels"

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

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

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

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, 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 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 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

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 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 All... I have searched in the archive and also followed Zara's instruction on getting incoming calls to work with Asterisk...but I still can't get Asterisk to answer incoming call on Linejack's LINE port. I attached a phone set to the PHONE port, and telco line to the LINE port on the Linejack(ISA) card. I have downloaded, compiled and installed the newest driver for

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

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

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

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

Hello, I have ISA card LineJack. I could not find any information if this card can work as fxo with Asterisk. If it can work, can somebody point me how to install it on my Asterisk box. Or maybe there is some documentation about it how to install LineJack. I will be very thankful for any help. Regards Bartosz Jozwiak ------------------------------------------------- This mail sent through IMP:

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

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

I'm staring with Asterisk and want to setup it up (at the moment) as simple answering machine different message will be plaid between 8am - 5pm and different after hours. I'll add extensions later on. I've Supira 3000. I was reading wiki page and to my understanding I have to place device=/dev/phone0 in phone.conf (actually it is there so I uncommented it). Though I

I want * to answer the phone when call comes-in. I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a command: mknod /dev/phone0 c 100 0 Though, when I start * I get: Parsing '/etc/asterisk/phone.conf': Found Sep 12 00:18:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' Sep 12 00:18:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable

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