similar to: Asterisk Beginner

Displaying 20 results from an estimated 10000 matches similar to: "Asterisk Beginner"

2003 Apr 29
10
Creating a phone channel
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
2003 Apr 28
9
Dialing using X100P
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
2003 Apr 25
9
Dialplan question
First, here's what I want to do / what I have: X100P and a Quicknet PhoneJack. I want to be able to pick up the analog phone (connected to the phonejack) and dial another computer (with the same hardware) or just make a regular phone call which will be decided by asterisk depending on the phone number dialed. I know that this won't be taking full advantage of asterisk, but I'm
2005 Mar 27
8
Asterisk on a dialup connection?
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
2003 Apr 28
2
VoIP Gateway
hello, I would like to realize a VoIP Gateway, with some extra-features. The aim is to get the phone number of the caller, to make research in our database, and to put him automatically through the good employee. The company is equipped with a VoIP network : software : - PSTNgw - Ohphone - OpenGatekeeper hardware : - Quicknet phonejack - gateway : Voicetroniw OpenLine4 Is it possible to
2004 May 31
3
Quicknet PhoneJack Configuration
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
2003 May 01
6
No Dialtone
So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the
2003 Jul 14
2
Using 2 PhoneJacks with Asterisk for Data calls.
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
2003 Apr 14
7
Trouble installing
I am trying to run the make command to install Asterisk, but I get the following error: make ... ... checking for tgetent in -ltermcap ... no checking for tgetent in -ltinfo ... no checking for tgetent in -lcurses ... no checking for tgetent in -lncurses ... no configure: error: termcap support not found I am running Mandrake 9.1 on a Pentium II 200MHz. Could this be a hardware issue? I
2004 Dec 19
2
QuickNet Internet PhoneJack problem
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.
2003 Dec 27
3
Setting up asterisk on Rh 9
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.
2005 Mar 28
2
AGI STREAM FILE command
Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after
2003 May 27
1
Duplicate numbers with outbounding calls
I've a problem with my X100P card. I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I call an PSTN phone number, some digits are duplicated, so I'm unable to call the right person. Not very clear ? I'll try to do better (sorry, I'm french...) example : I use ohphone (with quicknet hardware), I call asterisk (*192*168*1*204#), asterisk answers, I choose
2003 May 09
4
SIP Confusion
Ok. I am confused. I now have conflicting answers to my question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B. I would like to connect to iConnectHere, which uses SIP. Has anybody done this before (using similar equipment to what I have listed above)? And if it is not possible, could somebody please explain why. I don't understand why this wouldn't
2003 Oct 02
0
Help with ISA PhoneJack.
The device is seen in linux pnp: isapnp: Scanning for PnP cards... isapnp: Card 'Quicknet Internet PhoneJACK' isapnp: 1 Plug & Play card detected total and I've installed the drivers from the openh323 dev... but I can't get * to see it. Does anyone have experience with this? THanks. -Dave. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Sep 05
1
Number of digits
Perhaps this will help... I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I get a dial tone. When I dial a certain number of digits, the call is processed by Asterisk. The question: How does Asterisk determine how many numbers to let me dial? I'm banging my head against the desk here... _9XXXXXXX lets me make an outbound call, but _9X. only lets me dial 9 plus
2004 May 02
1
phonejack and linejack in the same system
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
2003 Jun 30
1
Beginner Questions
I've done a quick search through the old mailing list archives to see if I could dial an outside line through * using a quicknet linejack setup as an FXO. Is it posible to do so? It appears from the old mailinglist from marko.net that this not possible? Thanks. Bradley Greep
2003 Apr 30
9
TDM10B problem
Ok. I just got a TDM10B and it is in with my X100P. So as it says in the provided instructions, I used the command modprobe tor2 I get an error message saying that there is no such device. My zaptel.conf looks like this: fxsks=1 fxoks=2 So I load the X100P first. (modprobe wcfxo) Then I load the TDM10B (modprobe tor2) Then I'm told that the device doesn't exist. Please help
2003 Nov 16
0
* is crashing, when the call is accepted (H.323 -> SIP)
I'v got the following scenario: Two clients (ohphone) are calling (one at a time) the host with asterisk, which then connects to the SIP client. One of these hosts let's asterisk crash with a segmentation fault (i can provide the core file, if needed) in the second, the SIP client accepts the call. However .. if that client get's to the voicemail instead, because the SIP client is