2013/03/03 V1.0 Page 1 Dean Forest Railway - Asterisk Service The Asterisk service allows us to provide telephones on the DFR internal telephone network at home, over the internet. It's provided on a best efforts basis, and must not be used for operational, safety, or business purposes. It is maintained by volunteers who have full time jobs in Bristol. In the event of a fault it may be unavailable for quite some time until someone is able to visit site. Support is also best efforts and is provided via email. While we'll try our hardest to get you up and running no one involved has time to provide phone support, a home visit service, or to fix your broadband or computer problems for you! Equipment Needed To connect to the system, you will need the following equipment at home: A broadband internet connection with a spare ethernet port. A telephone which can do DTFM (tone) dialling. Most push button phones will work fine, unfortunately dial telephones such as those used on the railway don't work (at least, not with the ATAs documented here. An adaptor such as http://www.dialgizmo.com may help) An Analogue Terminal Adaptor (ATA) for example: Cisco SPA112 - This is our preferred model as it is available new, with a manufacturer warranty from retailers such as dabs.com http://www.dabs.com/products/cisco-2-port-phone-adapter-89b6.html ( 36.99) or found slightly more cheaply on ebay. Linksys/Cisco PAP2T - This was the model replaced by the SPA112. As it's no longer manufactured it's not our preferred option, but they are available second hand from ebay for 10 to 15. Be wary of buying one which is advertised as locked (which means it can only be used with a commercial provider) or auctions which ship from china as there are a lot of chinese fakes around! Other ATAs may work, but please ask before buying one so we can advise if it will be suitable. For example, do not buy a Zoom ATA! They're simply junk. We've had 3 so far and none of them work properly. A free ATA Alternative - the softphone A softphone is a computer program which will allow you to make/receive calls through the system, but you will only be able to receive calls if your computer is on and the program is running. We've found http://www.3cx.com/voip/softphone.html to be easy to set up, and it runs on Windows/iPhone/Android
2013/03/03 V1.0 Page 2 Requesting an account To request an extension on the system, please email thg@paulseward.com with the following information: Your Name (forename and surname for the directory) Your Email Address (so we can inform you of changes to the system) The name of your broadband provider (BT, Virgin, Pipex etc) as there are some settings at the server end which may need to be tweaked depending on your provider. Account creation is a manual process, and it may take a few days for all the required buttons to be pushed (so please be patient!) but once your account has been created we'll email you back the username and password details you'll need.
2013/03/03 V1.0 Page 3 Setting up your ATA Cisco SPA112 Step 1: Plug in your SPA112 as outlined in the instruction booklet, connected to your broadband router, and a telephone. Step 2: Turn on the SPA112 and wait until the light next to Line 1 is lit up. Then dial **** (four asterisks) this should give you the automated voice menu Step 3: Dial 110# The PAP2T will then read it's IP address out to you (eg 192.168.0.100) Write this down! Step 4: On your computer, open a web browser (internet explorer, firefox, chrome etc) and type the IP address into the address bar. This should bring up the login page for the ATA. The default Username is admin with a Password of admin Step 5: The first screen you'll see is the Quick Setup screen. Enter the following details Proxy Display Name User ID Password Dial Plan (optional) sip.dfrvoip.org.uk xxx
2013/03/03 V1.0 Page 4 Click Submit to save the settings Illustration 1: "Quick Setup" settings Step 6: Click on Network Setup (top menu) then Time Settings (left menu), and enter the following information: Time Zone (GMT) England Time Server Address Set the dropdown box to Manual and enter europe.pool.ntp.org Click Submit to save the settings
2013/03/03 V1.0 Page 5 Illustration 2: Time Settings All Done! Your ATA should now be configured to work with the DFR Asterisk system. How to test it We recommend the following tests, to make sure your ATA is working. Test 1 : Try dialling 400 to reach our speaking clock. Test 2 : Try dialling 401 which is an echo test number. This should echo your speech back to you so you can tell if audio is working in both directions. If both of those tests are successful you're good to go! If either of them fails, check your settings carefully and make a note of the date/time at which the test failed so that we can investigate further. Changing the admin password You may want to change the Admin password. To do this, click on Administration (top menu), User List (left menu) then click on the row for the admin user. I like to write the admin password on the bottom of the ATA so I don't lose it!
2013/03/03 V1.0 Page 6 Setting up your ATA Linksys PAP2T These instructions should work, but haven't been verified on a real PAP2T. Step 1: Plug in your PAP2T as outlined in the instruction booklet, connected to your broadband router, and a telephone. Then switch it on. Step 2: Turn on the PAP2T and wait until the light next to Line 1 is lit up. Then dial **** (four asterisks) this should give you the automated voice menu Step 3: Dial 110# The PAP2T will then read it's IP address out to you (eg 192.168.0.100) Write this down! Step 4: On your computer, open a web browser (internet explorer, firefox, chrome etc) and type the IP address into the address bar. This should bring up the web interface for the ATA. Step 5: Click on the Admin Logon button near the top right side of the screen, then click on the Line 1 tab. Illustration 3: PAP2T Menu The default Username is admin with a Password of admin Step 6: Scroll down to the Proxy and Registration and Subscriber Information sections. You only need to modify a few parameters from the default. They are: Proxy Display Name User ID Password Dial Plan (optional) sip.dfrvoip.org.uk xxx
2013/03/03 V1.0 Page 7 Illustration 4: PAP2T Settings Click Save Settings All Done! Your ATA should now be configured to work with the DFR Asterisk system. How to test it We recommend the following tests, to make sure your ATA is working. Test 1 : Try dialling 400 to reach our speaking clock. Test 2 : Try dialling 401 which is an echo test number. This should echo your speech back to you so you can tell if audio is working in both directions. If both of those tests are successful you're good to go! If either of them fails, check your settings carefully and make a note of the date/time at which the test failed so that we can investigate further. Changing the admin password You may want to change the Admin password. To do this, click on System (top menu), you should be able to enter a new admin password. I like to write the admin password on the bottom of the ATA so I don't lose it!
2013/03/03 V1.0 Page 8 Setting up your ATA Any other ATA or softphone Follow the instructions which came with your ATA, you'll need the following account information Proxy (or server ) Display Name User ID Password Auth ID Outbound Proxy sip.dfrvoip.org.uk Set this to the same as your User ID sip.dfrvoip.org.uk Most other settings can usually be left as the default. How to test it We recommend the following tests, to make sure your ATA is working. Test 1 : Try dialling 400 to reach our speaking clock. Test 2 : Try dialling 401 which is an echo test number. This should echo your speech back to you so you can tell if audio is working in both directions. If both of those tests are successful you're good to go! If either of them fails, check your settings carefully and make a note of the date/time at which the test failed so that we can investigate further.
2013/03/03 V1.0 Page 9 Setting up an Asterisk SIP Trunk If you have an asterisk server of your own, and want to use phones connected to it to make/receive calls from your DFR Asterisk account, the easiest thing to do is set up a SIP trunk. This effectively makes your Asterisk server behave as though it were an ATA. Step 1: Put the following in the [general] section of /etc/asterisk/sip.conf ; Register DFR trunk register => myusername:mypassword@sip.dfrvoip.org.uk Obviously you'll need to replace myusername and mypassword with your DFR username/password. Step 2: Add a user to the bottom of /etc/asterisk/sip.conf ; SIP trunk to the DFR Asterisk at Norchard [dfr trunk ] type=peer secret=mypassword authid=myusername username=myusername context=dfr incoming ; The context to use for incoming calls host=sip.dfrvoip.org.uk nat=yes qualify=yes dtmfmode=rfc2833 canreinvite=yes allow=all ; insecure=port,invite ; uncomment this if you need to fromdomain=sip.dfrvoip.org.uk Again you'll obviously need to replace myusername and mypassword with your DFR username/password. Step 3: Add a new context to /etc/asterisk/extensions.conf for incoming calls, eg: ;; Incoming calls from the DFR [DFR incoming] exten => s,1,log(notice, Incoming call from ${CALLERID(all)}) exten => s,n,dial(dahdi/g1/25,30) ; Ring ext25 on DAHDI trunk G1 exten => s,n,hangup() Step 4: Modify your main context in /etc/asterisk/extensions.conf to allow outgoing calls down the DFR trunk. For example: ;; Outgoing calls to the DFR start with 9, then 3 digits exten => _9XXX,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:3} through DFR 9xxx) exten => _9XXX,n,Dial(SIP/dfr trunk/${exten:1},60) exten => _9XXX,n,Playtones(congestion) exten => _9XXX,n,Hangup() You should be ok to core sip reload and core dialplan reload and dial when ready!