Any one know asterisk?

I'm toying around with asterisk (at least as far as reading about it goes) and I wanted to know if anyone out there has an experience with setting up and using it. What I want to do seems, to me, fairly simple. But then again, I don't know.

I essentially want to use asterisk (or trixbox) in a single user environment. I want a call to come in over PSTN (Q Is it possible for it to come in over a standard phone modem?) and then forward that call out to all registered phones until one of them answers.

Anyone with any ideas? ++Karmels for help

I think you just hit SHIFT-8 to use asterisk

searching for --karmels....

Wow, no post from h3x yet...gotta fix that....

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[quote253fed0bff="Fugger"]I'm toying around with asterisk (at least as far as reading about it goes) and I wanted to know if anyone out there has an experience with setting up and using it. What I want to do seems, to me, fairly simple. But then again, I don't know.

I essentially want to use asterisk (or trixbox) in a single user environment. I want a call to come in over PSTN (Q Is it possible for it to come in over a standard phone modem?) and then forward that call out to all registered phones until one of them answers.

Anyone with any ideas? ++Karmels for help[/quote253fed0bff]

Let's break you in slowly and let you get a feel for Asterisk first

Well, as you know from reading about (li), it runs on Linux... So I recommend using it on a computer that is strictly dedicated to it. You could dual-boot, but it's not recommended (you're trying to roll your own PBX... so you want to have your PBX running all the time).

Assuming you've read up on installation procedures on installing Asterisk (which is available on their site)... I'll show you how it works

- First, let's run the Asterisk CLI...

asterisk -vvvvvc

This will run the command line interface for Asterisk... First thing you should do when you run it, is type "help" and to find out more about something, such as SIP (single initiation protocol... it's a voip protocol), type in "help sip"... The help file in there is pretty much self explainatory. Now to shut down Asterisk, type in "stop now" from the CLI.

Now after you get Asterisk running, you're going to want to use a softphone to try it out... I recommend X-Lite www.xten.com[]www.xten.com

It works on Linux and Windows.

Now, you want to configure FWD and Asterisk to work with eachother... So first you need an account with FWD, here's the link http//www.freeworlddialup.com[]http//www.freeworlddialup.com.

After you make a FWD account, lets go to the A@H portal (assuming you're using Asterisk at Home)

Click Setup, Trunks, Click Add Trunk, Click Add IAX2 Trunk

Add the following in

Outbound Caller ID Fugger <FWD>
Maximum Channels 1

Trunk Name fwd
PEER Details

allow=ulaw

auth=md5

disallow=all

host=iax2.fwdnet.net

qualify=yes

secret=(the password you used to sign up at FWD)

type=peer

username=(your fwd user number again)

USER Context iaxfwd

USER Details

allow=ulaw

auth=rsa

context=from-pstn

disallow=all

inkeys=freeworlddialup

secret=(your fwd password again)

type=user

Register String

(your fwd number)(your fwd password)@iax2.fwdnet.net

Then, go to DID Routes, then Add Route... Put this in

DID Number (your fwd number)

Ring Group 1

Almost done here... Next, go to Outbound Routing

... and put this in

Route Name outgoingFWD

Dial Patterns 393|.

Trunk Sequence IAX2/fwd

[b253fed0bff]Your extension for FWD will be 393.[/b253fed0bff]

Now to set up X-Lite, go to the Setup Screen and choose the Default profile

Set Enabled to Yes, Set Display Name to Fugger, Set User Name and Authorization name to your FWD number, Set Password to your FWD password, Set your Domain Realm, SIP Proxy and Outbound Proxy to your Asterisk's IP address. Save settings. Then go to Menu, then Advanced Settings, then Audio Settings, then Silence settings, and enable Transmit Silence (so it's always sending packets regardless of if there is no noise being transmitted on the line).


Everything should be set up, try running the Echo Test from your X-Lite softphone... Dial the extension 393 first, and then 613 (so 393613)... Then speak when it tells you to, you should hear yourself echo.. If so, then you got a working Asterisk box!

As for PTSN numbers, here's what I suggest

http//www.ipkall.com[]http//www.ipkall.com (you'll get assigned a free [425, 206, 253, 360] number to use. This are Washington State PTSN phone numbers.

http//www.calluk.com/fwd/[]http//www.calluk.com/fwd/ (you'll get assigned a free [0870] number to use. This is a British (U.K.) number.

To use a standard telephone, get an analog phone adapter, such as IAXy

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That's my recommendation for a beginning li user...

Let's get your Asterisk fully operational first before we dive into the more complex stuff.

If you have any more questions, let me know.

Okay... Lets say I wanted to use this at my home and just wanted to use Cisco phones for the pure shits and giggles of using them at my home and I wanted to keep AT&T as my carrier. So my AT&T PTSN rings and is received through some FXO card like this one[=http//cgi.ebay.com/Professional-OEM-X100P-FXO-PCI-Digium-Asterisk-IP-PBX_W0QQitemZ160044078301QQihZ006QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem]this one and simply wanted all of the phones I have to ring then the PTSN line rings.

I have A@H/Trixbox working and have been playing around with some softphones. Just tried SJphone and I'll be trying the one you recommended. Any thoughts on this?

[quotec465b90927="Fugger"]Okay... Lets say I wanted to use this at my home and just wanted to use Cisco phones for the pure shits and giggles of using them at my home and I wanted to keep AT&T as my carrier. So my AT&T PTSN rings and is received through some FXO card like this one[=http//cgi.ebay.com/Professional-OEM-X100P-FXO-PCI-Digium-Asterisk-IP-PBX_W0QQitemZ160044078301QQihZ006QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem]this one and simply wanted all of the phones I have to ring then the PTSN line rings.

I have A@H/Trixbox working and have been playing around with some softphones. Just tried SJphone and I'll be trying the one you recommended. Any thoughts on this?[/quotec465b90927]

Sorry, I'm having difficulty understanding what it is you want to do exactly...

You want someone to make a call to your current PTSN number (via AT&T POTS) and have the FXO card pick up the call and Asterisk route it to the Cisco phone?

... and what kinds of Cisco phone is it? 7940? 7960?

The phone at this time is just softphone... Eventually it would end up being a few 7940's and maybe some sort of wireless phone.

I know this isn't an everyday use for li, which is why virtually everyone I've asked has just been confused to hell about what I'm trying to do.

A friend of mine recenty set up asterisk for a caller id spoofing website. What's your email address? I'll see if he's got a minute to help you out.

For the Cisco 7940, unlock the phone by pressing lili# and enable DHCP, then set Alternate TFTP Server to ON.

Update your 7940 firmware and drop a copy in the /tftpboot directory

Then run these commands

setup-cisco
chmod 777 /tftpboot/li

Then access your phone's web interface and then go to Maintenance, then Config, then Create

Label the first phone Cisco1 (or however you want to order the phones)

Then put in your phone's MAC address, For Line Name, put in 200 and Display put in 1. Remember to click Add when you're doing all of this.

Click Edit Phone Config Files and click Edit. Scroll to Line 1 and put in 200 under Authorization Name. Then put in a password.

Then you want to create an extension using the same number as the Authorization Name (extension 200) and again use the same password.

Then unplug and replug the phone. It [bba0f9b7053]should[/bba0f9b7053] update the firmware with the copy you dropped in tftpboot...

Hope this helps.

Oh, and when you boot the phone back up, use these commands to get rid of the XML generated config files

rm /tftpboot/xmlDefault.CNF.XML
rm /tftpboot/XMLDefault.cnf.xml

Though if you're going to use multiple Cisco phones, I recommend assigning the phone an available IP address rather than using DHCP. It makes things a lot less complicated.