Thomas, You've got some good advice so far, let me tell you what I'd check out and the order that I'd do it in.
Make sure that you've got that phone cord plugged into the right jack on your modem, you mentioned that there are two. One is probably for the line, and the other is for a phone. First things first, check your cables. Zapata.conf just sets up the hardware for your zaptel card (the x100p). It indicates what signalling is to be used on which channels. In your case you've only got one ZAP channel. Your /etc/zaptel.conf file probably only needs one line saying 'fxsks=1'. That means channel one is using FXS Kewlstart signalling. Running 'ztcfg -vv' will give you an idea as to whether that card is recognized and configured. If you don't see channel 1 listed with FXS Kelwstart next to it, don't go any further. You've got to resolve this first or you'll never get a call.
From there, check your /etc/asterisk zapata.conf. This file tells
asterisk how it's going to interact with ZAP channels. As others have said, zaptel is a particular technology used by asterisk on traditional telephony (TDM) interfaces. Here is where you specify all kinds of parameters for your channel. Off the top of my head, the minimum you could probably get away with is: ; /etc/asterisk/zapata.conf [channels] context=default ; or whatever your context is in extensions.conf signalling=fxs_ks ; note that fxo cards use fxs signalling group=0 ; probably not required strictly channel=1 Keep that file as small as possible at first because settings carry downward and it's simpler to debug if you've just got the basics. You should be able to dial out on that line using either dial(zap/g0/4165551212) or dial(zap/1/4165551212)
From there, you should be able to start asterisk -vvvvvvgc and see
some activity when the phone rings. You seem to have extensions.conf working. The call will be delivered to the s extension in whatever context you defined in zapata.conf. You should see something like " detected ZAP/1-1 ringing" on the console, even if extensions.conf is totally messed up. You can try running zttool from your shell and see if the card status is OK once asterisk is running.
From the asterisk CLI try zap show channels to see if asterisk has
recognized your card and it's config. Post back with results and we'll take it from there. BTW I totally understand your frustration. Once you've got through the same articles over and over and the same configs over and over it really drives you nuts. I ended up doing exactly what you're saying. I have a zapata.conf with an explanation/warning for what each line does, in my own words. Dave On 8/1/06, Thomas Keats <[EMAIL PROTECTED]> wrote:
Ok, I am feeling a little confused about how to get my asterisk system to start answering the phone. 1. I have updated to the latest build of asterisk 1.2.10 .. 2. Compiled with make, make all, make install, and make samples (after backing up /etc/asterisk for the old version) 3. I've read and read and re-read what seems to be the same articles about the same topics that do not seem to help... I swear, if I ever get this working, I am going to put sample configs online with LOOK DUMMY, THIS DOES THIS on EACH line of the file ;) (in MY simplistic terms) My problem right now is I must be confusing some things. As this system is not intended to have internet access, nor access to another asterisk or voip server, I get confused as most of the sample files already have that hooked in. Zap/1 Zap/2 ? What is Zap, and how do I know what device its using, I am assuming its a device... How does the modem show in the /dev/ is it the atypical /dev/modem/ or /dev/ttyS# ? (Consequently when I look through my sysinfo in /proc/ the modem reports as a Intel 537.. which I understand to be the X100P, tho it has 2 RJ's on the back of it) At this point I don't even know what files I should be seeking help on for editing, I think I understand that for setting up my softphones I edit sip.conf (or the [protocol being used].conf). and extensions.conf. (I have working extensions at this point... and have connected with typing 8500 and did the tests with digium...) When I am in CLI, and a call comes in, CLI gives me NO feedback. None at all. Nothing in the /var/log/asterisk files either... Can someone supply me with a inkling so I might discover the rest of the clue I need to continue? Thanks Thomas Keats P.S. /dev/dsp does not seem to exist, or is unable to be re-opened. I have no 'console' functionality, however I am under the understanding that this doesnt matter unless you plan to have a intercom, is this correct? --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
-- David Donovan Consultant Fulcrum Solutions
