How to configure Asterisk without zaptel/dahdi hardware
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How to configure Asterisk without zaptel/dahdi hardware
Hello! I am using SAMSUN R-519 laptop with 4 GB RAM and 2.1 GHz dual core CPU and I intend to configure asterisk on this machine for voice conferencing with 4 other PCs. I am unable to configure asterisk. I don't have any hardware regarding voip such as Digium card. Can anyone tell me how do i should start??? Thanx in advance.
I think you can just install asterisk. You don't need any dahdi or zap devices. If I bring up my asterisk server without bringing up dahdi first, it doesn't complain about anything, it just runs as a pure SIP server.
My standard asterisk install on fresh centos (5.4) goes like this, I'm 95% sure you can skip all the dahdi stuff and still have a running asterisk service
I just ran thru that setup procedure on a standard centos install with no telephony hardware, not even a modem. Just a typical office workstation - it's running asterisk right now, processing whatever SIP traffic I throw at it. Just install it, it'll work just fine.
Nothing, dont start dahdi. Like you said, you have no dahdi hardware. Just start asterisk, it'll use whatever it has, ie ethernet / SIP. You don't need to do the addons either, unless you want to use mysql or other non-essential features. Asterisk right out of the box is enough to run a SIP server for typical smaller VoiP setups.
I don't know a whole lot about asterisk, but am curious how you would run a VoIP setup without any of the hardware it uses. IP phone? If you can just point to some type of tutorial or example configuration or something that would be helpful I'm interested
The hardware asterisk would be using is the ethernet port on the server. There's a million IP phones out there, they're completely ethernet - no phone circuits whatsoever. You set their ip addresses, they download their configs via a web server, they contact the asterisk (or other sip pbx) 'phone switch' over ethernet, etc.. We use a few models of avaya 4600 series ip phones, and have a few various polycom phones lying around. And if your PC has speakers and a mic, you can set up a soft-phone - xlite, zoiper, diax, twinkle for linux users, there's a ton of those too.
Asterisk will obviously also support telephony hardware - Digium makes cards with E1/T1 ports, and FXO/FXS ports, etc. But they are not required if you don't use POTS (Plain Old Telephone Service) telephony. You can run your entire phone system over ethernet.
To connect to the outside world, you can subscribe to SIP providers instead of getting a trunk to AT&T or Ameritech or whatever your local POTS provider is.
When i try to start asterisk with "asterisk -vvvc"
it gives the following error and does not start i have waited for hours but still it remains on loading:
chan_usbradio.c:1264 find_desc: null dev
[Aug 22 14:49:59] WARNING[29914]: chan_usbradio.c:1269 find_desc: could not find <--no-device-->
and everything else is commented out. I do not use any kind of usb or radio devices, if you aren't either, and it sounds like you aren't, check this file and comment out everything except the [general] line.
If asterisk isn't started, start it with
asterisk
on the command line, that will start the service in the background and return your prompt right away. Then start a console with
I have started the asterisk with only "asterisk" command it starts good and no error occur but there is another problem that i don't have all the commands such as reload, stop, etc. What should i do? my asterisk version is 1.6.1.18. Bundle of thanx
and get to an asterisk console. Once you're there, you can issue tons of asterisk commands to see what's what.
help
will show you all the available commands
help sip
will show you some SIP related commands.
help sip unregister
will show you the details of that command.
dialplan show
will show your dialplan, all the extensions your asterisk server is managing, and what they do.
Leave that console up, and try to register a SIP phone to it. You'll see that it registers or you'll see an error. Make a call, you'll see lines appear in the console relating to that.
in the CLI, if it doesn't exist, you need to get it, or re-compile asterisk with SIP support. The instructions I posted above would include SIP support, how did you install asterisk?
I have started the asterisk with only "asterisk" command it starts good and no error occur but there is another problem that i don't have all the commands such as reload, stop, etc. What should i do? my asterisk version is 1.6.1.18. Bundle of thanx
Per the instructions I posted above:
cp /files/asterisk-1.6.2.2/contrib/init.d/rc.redhat.asterisk /etc/init.d/asterisk
chkconfig asterisk on
service dahdi restart
rc.redhat.asterisk is the init.d script that starts / stops / reloads.
I have installed the asterisk using yum. I dont know why the asterisk is not working the module reload chan_sip.so also generates error. I dont know what to do.............
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