similar to: Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP

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'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

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

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 List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

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

This post is a FWDout specific one, but may be of interest for many Asterisk users and may even make more of them use FWDout for the good of everyone :) On the one hand, I have seen many reports of people using the FWDout (http://www.fwdout.net/) service who don't get credits because the prefix they provide calls for to is not popular. On the other hand, I get plenty of unused credits, much

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

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, 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

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10

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.

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 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

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 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.

which ir is the * channel? I have decleared 1 peer one in veracruz: [aullox_gdl] type=peer username=aullox_gdl host=200.67.99.127 and aat gdl I have: exten => 200,Dial(IAX/aullox_gdl@200.64.35.58/200@aullox) 200.64.35.58 is my ip 200 is tehe xtension and aull is the context and at veracruz i have: [aullox] exten => 200,1,Wait,2 exten => 200,2,Playback(transfer,skip) ;

Hi all, Currently trying to get asterisk to dial out with an Internet Line Jack card, however, it does not use the pots line, only on the line it dials out of. This is similar to the previous thread/posting "Asterisk won't answer pstn ring", but I didn't find any follow up to get it working. My asterisk setup is like this: iptelephony:/etc/asterisk# cat phone.conf | grep -v

At the CLI, should the command "show channels" list the configured devices that may be used for calls? Or, is that command simply a way of viewing the channels on a T1/E1 if you have that type of interface? If the latter, is there an equiv. command to list the single FXO interfaces? If the answer is yes, all configured devices should have something in channels, how do you map a

The agenda for Astricon Europe in Madrid June 15-17 is now coming together. It will be an international conference, with speakers from both USA and Europe. Last year, we had over 25 nationalities participating in the first Astricon - the Astricon where Mark released Asterisk 1.0 on the conference floor, during his keynote! Many active members of the Asterisk community talks at the conference, one