Please Note: The content on this page has been out of date for a long time: More current info can be found at http://voip-info.org
by Shawn L. Djernes shawn@djernes.org
Linux machine running Asterisk from CVS
Cisco ATA186 with at lease version 2.14 of the firmware
Time!
I am going to make some assumptions here:
You are on a network with a DHCP server
Your Asterisk box is on the same network
If not, then make adjustments where necessary. For more information on configuring the system go to http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/
Connect your ATA186 to the network, power and a phone as described in the instructions. As soon as the unit receives power it will look for the DHCP server for an address. While it is doing this the red light on the top of the unit will blink. Now lets find out what address it got.
Lift the handset and press the function button on the ATA186
Enter 21# on the telephone keypad.
This will read back your units IP address. Please write it down we will need it a lot as we go on.
Now for the actual configuration you will need a web browser. You could set all these options via the keypad of a phone but that would be very tedious. So open a new browser window or another console and lets start.
Goto the URL http://<ip_of_device>/dev
If you are asked for a password enter it.
You should now see a Cisco ATA 186 Configuration with many colored boxes.
First set ToConfig to 0 (we are configuring it now)
Set UseTFTP to 0 (don't need it to try to configure itself again)
If from this point on you wish the ATA186 to have a static IP then set DHCP to 0 and fill in the appropriate Static and DNS boxes. Otherwise leave this at 1
Set UID0/1 to ata1/2 or other unique names. This is important for the Asterisk sip.conf
Set PWD0/1 to ata1/2 or other more secure values.
Set GtkOrProxy to the IP address of your Asterisk box.
Make sure UseLoginID is 0
Set UseSIP to 1
Make sure SIPregON is 1 and SIPport is 5060
Set LBRcodec to 3 (this truly does not not matter because we are disabling it later)
Set RXcodec and TXcodec to 2 (G.711u)
Set AudioMode to 0x11241124 (this turns off LBRcodec and sets up DTMF out of band)
You can set TimeZone to the correct value for your area EST = 20, CST = 19, MST = 18 PST = 17. To get this number if you are in the Western Hemisphere you take your time offset plus 25 or for Eastern Hemisphere this is your time offset from GMT.
You may want to set NTPIP to 194.43.244.18 (time.nist.gov) or another NTP time server. This will work along with TimeZone to give correct feedback on CallerID time.
Click the Apply button at the bottom of the page.
NOTE: This configuration is safe if your Asterisk Box and ATA186 are on a private/NAT network or are behind a firewall. If this is not the case you should set a password in UIpasswd and use more complex UID and PWD values.
By now you should have Asterisk compiled and installed on your machine. If this is not the case then go to www.asterisk.org and follow the documentation. Once that is done you are ready to continue here.
Open sip.conf in your favorite editor
Make sure the basic stuff is setup like follows if you are using the DevKit Lite. Otherwise change the context to your incoming call context. This will allow other unregistered SIP phones and soft phones to call your box in the form of s@hostname and be treated just as if they were on the PSTN phone line,
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = bell ; Default for incoming calls
Now lets create the SIP definitions so Asterisk knows who we are when we call in.