Hackers Chronicle - Online Cyber Security News: SIP

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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh8qyrDKsJr96I5tdiSkEqQO2HNoT7USZ_JHOjCF_edBE_IyfzOC-MebaZu5ESzvJMuLHN6XJM-8N_2o-BT6BVYR860MyA9BKGTaO7hLd_36qOdkvcnGVuJ1JV8H6AK5OMjH46bCFsLNF8/s1600/asterisk-pbx.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img alt="Set Up Your Own IP-PBX with Asterisk" border="0" height="136" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh8qyrDKsJr96I5tdiSkEqQO2HNoT7USZ_JHOjCF_edBE_IyfzOC-MebaZu5ESzvJMuLHN6XJM-8N_2o-BT6BVYR860MyA9BKGTaO7hLd_36qOdkvcnGVuJ1JV8H6AK5OMjH46bCFsLNF8/s1600/asterisk-pbx.jpg" title="Set Up Your Own IP-PBX with Asterisk" width="400" /></a></div>
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<span class="Apple-style-span" style="font-family: "times new roman"; font-size: medium;">Set Up Your Own IP-PBX</span></h1>
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<span class="Apple-style-span" style="font-family: "times new roman"; font-size: medium;">Note: I know this is seriously out of date.  I have not been able to update it, and I don't expect that to change for the forseeable future.  Still, many people continue to find it useful so I will keep the page around for a while longer.  I cannot provide help or answer questions on it, however.</span></div>
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<span class="Apple-style-span" style="font-family: "times new roman"; font-size: medium;">The state of open source software has progressed to the point where you can set up your own IP PBX at home in a single evening, with a minimum of investment needed.  All you need is explicit instructions, which is what I provide here.</span></div>
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In past times, it would have been quite costly to have a "personal" PBX.  Using spare hardware, open source software, and low-cost service providers, it can now be done for almost nothing.  The key advance is the open source Asterisk IP PBX and the Asterisk@Home package that includes Asterisk and a web-based GUI configuration tool.</div>
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The Components</h2>
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You will have to set up three main components: the IP PBX itself, the phones (or softphones) to be used with <span class="GramE">it,</span> and the gateway service that lets you call other people on the PSTN.  I will describe how to set up each of these.  I assume you already have a home network and broadband access.  If you are behind a NAT firewall, it does not matter – you don't have to do anything special like running the IP PBX outside the NAT.  All that matters is that you have enough bandwidth (upstream and downstream) to carry voice traffic.</div>
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The IP PBX</h3>
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You will need a computer to run the IP PBX.  While you may already have a Linux server at home, I do not recommend using it to also run your phones.  If you want to do that, you will have to figure out how to load and configure the PBX software on your own, although the rest of the instructions here will be helpful.</div>
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If you have an old PIII class machine lying around, then use that.  If not, you can buy one on eBay for about $40.</div>
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I used a Dell OptiPlex GX1, a PIII 450MHz system with 128M RAM.  It has a built-in sound and Ethernet, so no additional hardware is required.</div>
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You will load Asterisk@Home on this computer.  It will take it over – it starts by formatting the hard disk, so make sure there is nothing on the machine that you want to keep.  Detailed instructions are given below.</div>
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Phones</h3>
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You can buy SIP phones from Grandstream, SNOM, Cisco, even Avaya.  Or you can buy an adapter from SNOM or Cisco and use an analog phone.  To get started, though, it is easiest to get a softphone and run it on another computer.  We will describe how to obtain and install the XLite softphone on a Windows machine to use for a phone.  You will also use this Windows machine to administer the IP PBX, through a browser.  If you want to use a Linux machine instead, you will need to get an appropriate softphone for it, but you can still use its browser to administer the IP PBX.</div>
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SIP Gateway</h3>
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I presume you want to communicate with others on the PSTN network, so you need to obtain gateway service.  Since part of the call is being carried on the circuit switched network, it costs real money.  This means you will have to pay for this part of the system. </div>
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I will describe how to set up access to two services, one for outgoing calls and another for incoming calls.</div>
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VoipJet is used for outgoing calls.  It is priced at 1.3 cents per minute (USA), with no monthly minimum or service charges.</div>
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BroadVoice is used for incoming calls.  You get a phone number (you can choose the area code and exchange).  They have various plans, but I signed up for one that is $5.95 per month, with unlimited incoming minutes. You can also make outgoing calls using this account (it includes 100 minutes per month, and 3.9 cents per minute additional).  There is a $9.95 one time activation fee to sign up for BroadVoice.</div>
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So if you have the hardware, you can set up your IP PBX for a total of $9.95 and run it for $5.95 per month.    </div>
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Home Network</h3>
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I assume you have a home network, and that you are setting up behind a Gateway Router (otherwise known as a NAT firewall).   You will need to pick a static IP address for your IP PBX that is on your home network. </div>
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How Much Will This Cost?</h2>
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I assume you have broadband service, a router, and a Windows machine to run the softphone. </div>
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If you already have a spare computer that you can dedicate to this project, there is no cost at all for equipment, unless you need to buy an audio headset for the softphone.  If you <i>do not</i> have a spare computer, then you can buy one on eBay for about $60, including shipping.  This does not include a monitor, but you don't need a monitor except briefly when you first set it up.  I assume you can borrow a monitor (or that you have a KVM switch).</div>
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Your only other initial cost will be the $9.95 activation fee to BroadVoice.</div>
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Your monthly cost will be $5.95 for incoming service from BroadVoice, and 1.3 cents per minute for outgoing calls to VoipJet.  If you only make a few outgoing calls, you could drop VoipJet, and stay within the 100 outgoing minutes that you get from BroadVoice.  If, say, you use 100 minutes of VoipJet outgoing calls, it would cost an additional $1.30.</div>
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So you can build and experiment with your own IP PBX for an investment of less than $100, and for an ongoing cost of less than $100 per year.  This is a lot less than I am now paying for my phone service from AT&T.  In fact, a whole year's service would be less than my monthly phone bill.  <span class="GramE">Hmmm…</span></div>
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Step 1 – Sign Up for Service</h2>
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This section takes you through signing up with VoipJet and BroadVoice.  I am using two <span class="GramE">service</span> because</div>
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<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">it is less costly, if you use it a lot, and</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;"><span class="GramE">it</span> illustrates how to set up two different kinds of trunks.</li>
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VoipJet</h3>
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Browse to <a href="http://www.voipjet.com/" style="color: blue; text-decoration: underline;">http://www.voipjet.com</a>.  <span class="GramE">Sign up for service.</span> Then log on and follow the line describing how to set up Asterisk. You will need to copy down your "VoipJet account number (username)," your "Authorization code (password)," and your server IP address (depending on your location). Ignore the rest of the setup instructions.</div>
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BroadVoice</h3>
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<img align="right" height="75" hspace="12" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image004.gif" width="295" />Browse to <a href="http://www.broadvoice.com/" style="color: blue; text-decoration: underline;">http://broadvoice.com</a>.  Sign up.  Say "I want to use my own SIP device".  When it asks what type of device, select "Not Listed (Generic SIP).  When it asks for details, just say "Asterisk".  Next, pick your phone number.  Next, pick your plan.  I recommend "BroadVoice BYOD Lite."  Once you have finished the signup process, log in, click on "Account" and follow the "Show Settings" link in the "Your Devices" section.  This section gives your phone number (in case you forget) and your password (you will need this later). </div>
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You should also click on the "Support Center" link on the left and bring up the "Step By Step Installation Guide" for "Bring Your Own Devices."  Select "Asterisk."  You must follow the instructions in Section 3 to find the "right proxy" to use.  This involves <span class="GramE">pinging</span> their proxies to find which has the lowest latency.  You will use this later.</div>
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            Detailed Instructions</div>
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1.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>Ping the following hosts: <span style="font-family: "courier new";"><a href="http://proxy.lax.broadvoice.com/">proxy.lax.broadvoice.com</a></span>, <span style="font-family: "courier new";"><a href="http://proxy.dca.broadvoice.com/">proxy.dca.broadvoice.com</a></span>, and <span style="font-family: "courier new";"><a href="http://proxy.mia.broadvoice.com/">proxy.mia.broadvoice.com</a></span>.</div>
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2.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>Pick the one with the lowest latency.  In my case, it was <span style="font-family: "courier new";">proxy.dca.broadvoice.</span>com.</div>
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3.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>Use <span style="font-family: "courier new";">nslookup</span> to get the IP address of <a href="http://proxy.dca.broadvoice.com/">proxy.dca.broadvoice.com</a>.  In my case it is <span style="font-family: "courier new";">147.135.0.128</span>.</div>
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Step 2 – Set Up Home Network</h2>
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Pick an IP address for the IP PBX.  You will need to find an unused address on your subnet outside the range assigned by your DHCP server.  I picked <span style="font-family: "courier new";">192.168.0.40</span>.  If you pick a different IP address, you will have to adjust the instructions accordingly.</div>
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Step 3 – Set Up SIP Softphone</h2>
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<img align="right" height="56" hspace="12" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image005.gif" width="220" />Download and install XLite from <a href="http://xten.com/" style="color: blue; text-decoration: underline;">http://xten.com</a>. </div>
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Run XLite.  Click on the "Menu" icon to configure it.  <span class="GramE">Click on "System Settings", then "SIP Proxy", then "Default".</span> </div>
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Fill in the following fields:<br />
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<span lang="EN" style="font-family: "courier new";">User name: 200<br />Authorization user: 200<br />Password: abc123<br />Domain/Realm: 192.168.0.40<br />
<span class="caps">SIP </span>Proxy: 192.168.0.40</span><span lang="EN"></span></div>
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It should look something like this.</div>
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The phone will try to register, but for now it will fail.</div>
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Step 4 – Set Up Asterisk@Home</h2>
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<img align="right" height="37" hspace="12" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image007.gif" width="300" />The rest of the instructions explain how to install and configure Asterisk@Home.  From here on, I will refer to it as AAH. </div>
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Introduction</h3>
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AAH is a package consisting of several major components.  These were developed and supported relatively independently.  The "Asterisk" part is the core IP PBX, and the "@Home" part consists of applications, a provisioning system, an installer, and an operating system that, together, make a complete package.  The major components that make up AAH are</div>
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<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Asterisk, the core PBX</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Sugar, a CRM system</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Flash Operator Panel, a screen-based operator's console</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Web Meet Me Control, a meet me conferencing control application</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Asterisk Management Portal (AMP), a web-based provisioning tool for Asterisk</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">A report system, part of AMP, which provides CDR reporting tools</li>
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<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">CentOS, a version of Linux related to from Red Hat Enterprise Linux (but without Red Hat branding and support).</li>
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Get the Software</h3>
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AAH can be downloaded from <a href="http://asteriskathome.sourceforge.net/" style="color: blue; text-decoration: underline;">http://asteriskathome.sourceforge.net</a>.  Download the basic ISO file, and burn it to a CD as a bootable volume (exercise left to reader).</div>
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Load AAH</h3>
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<i>This is going to reformat your hard disk and load everything from the operating system on up, so make sure there is nothing on the hard drive that you want to save. </i></div>
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Boot your machine from the CD.  When it prompts, <span class="GramE">type ENTER</span>.  Then wait as everything loads and compiles.  This could take 30 minutes or more, depending on how fast your computer, hard drive, and CDROM are. At the end, it ejects the CD and reboots from the hard drive. </div>
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The initial login is:</div>
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<span style="font-family: "courier new";">username: root</span></div>
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<span style="font-family: "courier new";">password: password</span></div>
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Set Up Networking</h3>
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The machine probably got an IP address from DHCP, but it is not what you want.  Log in as root and run:</div>
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<span class="SpellE"><span style="font-family: "courier new";">netconfig</span></span><span style="font-family: "courier new";"></span></div>
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<br /></div>
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It will display setup information.  You should enter the following:</div>
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<br /></div>
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<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 3in;" valign="top" width="288"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">IP address: 192.168.0.40</span></div>
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<span style="font-family: "courier new";">Netmask: 255.255.255.0</span></div>
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<span style="font-family: "courier new";">Gateway: 192.168.0.1</span></div>
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<span style="font-family: "courier new";">DNS Server: 192.168.0.1</span></div>
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<br /></div>
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OK these <span class="GramE">changes,</span> then reboot the machine to make them take effect.</div>
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After reboot, long in once more.  You need to add the BroadVoice server IP address to /etc/hosts.</div>
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Edit /etc/hosts, and add the following line at the end:</div>
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<span style="font-family: "courier new";"><a href="http://sip.broadvoice.com/">sip.broadvoice.com</a> 147.135.0.128</span></div>
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<br /></div>
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Here the IP address was the one obtained when signing up with BroadVoice.</div>
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<br /></div>
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Once this is done, you can do the rest of the configuration through the web.</div>
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Browse <span class="GramE">To</span> AMP</h3>
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From your web browser, go to <a href="http://192.168.0.40/" style="color: blue; text-decoration: underline;">http://192.168.0.40</a>.  You should see the following:</div>
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<img border="0" height="310" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image009.jpg" width="576" /></div>
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Select "Asterisk Management Portal."  Log in as follows:</div>
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<br /></div>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 0.5in;"><tbody>
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<span style="font-family: "courier new";">username: <span class="SpellE">maint</span><br />password: password</span></div>
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<br /></div>
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Now you should be at the AMP main screen.</div>
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<img border="0" height="249" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image011.jpg" width="575" /></div>
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<br /></div>
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Click on "Setup" to bring you to the main setup screen.</div>
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<img border="0" height="418" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image013.jpg" width="576" /></div>
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<br /></div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Set Up BroadVoice Trunk</h3>
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<span class="GramE">Click on "Trunks" and then "Add SIP Trunk."</span>  You see a blank SIP trunk form.</div>
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<img border="0" height="863" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image015.jpg" width="523" /></div>
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You will need to fill out the main items on the SIP/Trunk screens. I am using BroadVoice for incoming service only, so I will not administer outgoing trunk information. </div>
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<span style="font-family: "courier new";">Outbound caller ID:  7237570239<br />Max channels: 1</span></div>
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The outgoing settings can be left as is, except to fill in the trunk name as <span style="font-family: "courier new";">BroadVoice</span>.</div>
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In Incoming Settings, fill in the following:</div>
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User Context: 7327570239<br />
User Details:<br />
<span style="font-family: "courier new";">callerid=7327570239<br />context=from-pstn<br />dtmfmode=rfc2283<br />fromdomain=<a href="http://sip.broadvoice.com/">sip.broadvoice.com</a><br />host=<a href="http://sip.broadvoice.com/">sip.broadvoice.com</a><br />insecure=very<br />secret=********<br />type=user<br />user=7327570239<br />username=7327570239</span></div>
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The format of the Register String is: <i>username:password:phone_number@provider_domain</i>.  For BroadVoice this is what you will want to use (replacing with <i>your</i> BroadVoice phone number & password):</div>
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<span style="font-family: "courier new"; font-size: 10pt;">7327570239@sip.broadvoice.com:**********:<a href="http://7327570239@sip.broadvoice.com/200">7327570239@sip.broadvoice.com/200</a></span></div>
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Once all of this information is entered, you can click the <span style="font-family: "courier new";">Submit</span> button. Once you do, a red line will appear at the top of the page. To "apply" your changes you <b>must</b> click this red line (this will cause AAH to tell the Asterisk engine to reload its config files).</div>
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<span class="Heading3Char" style="font-family: "arial"; font-weight: bold;"><span style="font-size: 13pt;">Set Up VoipJet Trunk</span></span><b><span style="font-family: "arial"; font-size: 13pt;"></span></b></div>
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<span class="GramE">Click on "Trunks" and then "Add IAX2 Trunk."</span>  You see a blank IAX2 trunk form.</div>
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<br /></div>
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Fill in the fields as follows:</div>
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<span style="font-family: "courier new";">Outbound Caller ID: 7327570239<br />Maximum channels: 1</span></div>
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The outgoing dial rules control how numbers are processed before they are sent to the trunk.  In the case of VoipJet, phone numbers must be in the form "1+Area <span class="SpellE">Code+Local</span> Number".  If the number appears to have an area code but not a "1", then it is added.  If it does not have an area code, then "1732" is added (I am in area code 732).  So the dial rules are:</div>
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<span style="font-family: "courier new";">1+NXXNXXXXXX<br />1732+NXXXXXX</span></div>
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This time we will fill in outgoing settings. </div>
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<span style="font-family: "courier new";">Trunk name:: voipjet<br />Peer Details:<br />auth=md5<br />secret=***********  (get this from voipjet config info)<br />host=216.118.117.46  (get this from voipjet config info)<br />type=peer<br />username=***** (get this from voipjet config info)<br />
<span class="SpellE">notransfer</span>=yes<br />context=default</span></div>
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<br /></div>
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You can leave the "Incoming Settings" and the "Register String" blank.</div>
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When you are done, the form appears as follows.</div>
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<br /></div>
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<img border="0" height="492" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image017.jpg" width="576" /></div>
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Set Up Extensions</h3>
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Now it is time to define the extensions.  I will set up two extensions, number 200 in the basement and number 201 in the study.  I have already shown how to configure XLite extension 200, and you can configure another extension like it for 201.  Or you can set up a SIP phone instead.</div>
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Click on "Extensions" on the left, and "Add an Extension."  You will see the extensions page.</div>
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<img border="0" height="506" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image019.jpg" width="576" /></div>
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<br /></div>
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The information you will add is the following:</div>
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<span style="font-family: "courier new";">Extension: 200<br />Password: abc123<br />Full name: basement<br />Voice Mail Password: 1234<br />Email Address: <your email address></span></div>
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When you're finished adding the extension, you can see the details of the extension by clicking on its name on the right-hand side of the AMP interface (as shown above).</div>
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Add extensions 201 similarly.  Remember to click on the red bar, to make Asterisk take not of your changes.</div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Check for Phone Registration</h3>
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At this point, the XLite application should have registered with AAH.  If not, exit XLite (you have to stop it from the tray icon) and restart.  It should say "Logged in".  If not, see the troubleshooting section below.  To start with, make sure you have assigned the same password in the extension form and in the phone.</div>
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<img border="0" height="379" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image020.gif" width="275" /></div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
On XLite, dial <span style="font-family: "courier new";">*23</span> to test audio input and output levels. </div>
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Set Up Ring Group</h3>
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Now we set up a ring group, so that we can ring all the extensions at once.  This is useful, for instance, so that incoming calls <span class="GramE">can  alert</span> at all extensions.</div>
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<span class="GramE">On the left, click "Ring Groups" and on the right "Add Ring Group".</span></div>
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<img border="0" height="504" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image022.jpg" width="575" /></div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
Fill out the form as follows:</div>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 41.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 387pt;" valign="top" width="516"><div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
<span style="font-family: "courier new";">Group Number: 1<br />Extension List:<br />  200<br />  201<br />Ring Time: 18<br />Destination if no answer: Voicemail basement <200></span></div>
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<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0.5in; margin-right: 0in;">
<span style="font-family: "courier new";">></span></div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
Submit the changes, and click the red bar to make them take effect.</div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Set Up Digital Receptionist</h3>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
A digital receptionist is an application that answers incoming calls, interacts with callers, and forwards their call on.  It allows callers to dial an internal extension, to access the company directory, or to reach selected destinations according to predefined dial patterns.  We will set it up to dial by extension, to give the company directory, and to ring all extensions if the caller presses "1". </div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
The digital receptionist set up involves a series of screens and actions, not all of which will be illustrated.  Part of the setup includes recording a message to be played to callers.  You can record this independently, and download the MP3 file, or you can use XLite to dial in and make the recording.  The latter method will be illustrated.  In general, the setup instructions are clear and explicit, so this will only give the essentials.</div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
Click "Digital Receptionist" on the left.  Enter <span style="font-family: "courier new";">200</span> as the current extension number.  On XLite, dial <span style="font-family: "courier new";">*77</span> and record a message.  It goes something like this:</div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
<i>Hello.  You have reached Charles Hayden's experimental Asterisk phone system.  You can dial a three digit extension.  Dial 1 to ring all <span class="GramE">extensions .</span> Or dial pound for a directory.</i></div>
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Hang up and dial <span style="font-family: "courier new";">*99</span> to hear the message.  Rerecord if necessary.</div>
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Name the recording "<span style="font-family: "courier new";">top greeting</span>" and add a description.</div>
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On the next page, for "Number of options for Menu top greeting enter <span style="font-family: "courier new";">1</span>.</div>
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On the next screen, choose "<span style="font-family: "courier new";">Ring Group #1</span>" as the action.  This means that when the caller enters "1" it will go to Ring Group #1, which rings all extensions.</div>
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Set Up Outbound Routing</h3>
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Click on the "Outbound Routing" link and fill in the following:<br />
<br /></div>
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<span style="font-family: "courier new";">Route Name: outgoingVoipJet<br />Dial Patterns:  011.<br />                1NXXNXXXXXX<br />                732NXXXXXX<br />Trunk Sequence: IAX2/voipjet</span></div>
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The dial patterns given here will allow you to dial international calls, long distance with 1+area code, and calls within the 732 area code with only the area code but without the 1.  This is how it works here in the 732 area. If you can dial locally without the area code, then you could add "<span style="font-family: "courier new";">NXXXXXX</span>" as well.   As we have seen, the trunks cannot necessarily accept all these dialing combinations, so they have their own rules to screen and transform them.</div>
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<img border="0" height="478" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image024.jpg" width="576" /></div>
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If you've made it this far you should be able to dial "7777" on your SIP phone - this will simulate an incoming phone call - and you should hear your greeting. At this point the system should have enough functionality to dial out through VoipJet.</div>
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Go ahead and call a number and it <i>should</i> ring at the other end. Remember that you need to dial 1 followed by a 10 digit number. If it does then <b><i>pat yourself on the back!</i></b> Only a few more steps are needed to get incoming calls to work (you can try to call your BroadVoice number, but it won't work yet).</div>
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Set Up Incoming Calls</h3>
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Now we will set up incoming calls to go to the digital receptionist.  Choose the "Incoming Calls" link, and click the "Digital Receptionist" "top greeting".</div>
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<img border="0" height="503" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image026.jpg" width="575" /></div>
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Edit extensions.conf</h3>
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So far we've done everything through the AMP web GUI. Now we have to drop into editing text files. This is obviously not desirable, but this is how it is right now, so you have to go through this.  You can edit extensions.conf through the AMP web GUI.   Click the <span style="font-family: "courier new";">Maintenance</span> link at the top of the AMP page and then click the <span style="font-family: "courier new";">Config Edit</span> link in the left-hand menu. Then click on <span style="font-family: "courier new";">extensions.conf</span>.</div>
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Look on the left for <span style="font-family: "courier new";">from-sip-external</span> and click it.  By default, AAH sends all incoming SIP calls to a congestion status. This means that all incoming calls will ring fast-busy to the caller and they will never make it into our AAH system.</div>
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There are four lines under the <span style="font-family: "courier new";">[from-sip-external]</span> section. One is already commented out with a semi-colon (;). Comment out the remaining three lines by adding semi-colons. Now we need to add two new "exten" lines.</div>
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The first is:</div>
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<span style="font-family: "courier new";">exten => _.,1,Wait(1)</span></div>
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<br /></div>
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<span class="GramE">which</span> tells AAH to wait for 1 second once a call has been detected from an external SIP device. This makes sure we don't clip off the first part of our greeting message. The second line we need to add is:</div>
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<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 3.75in;" valign="top" width="360"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span class="GramE"><span style="font-family: "courier new";">exten</span></span><span style="font-family: "courier new";"> => _.,2,Goto(from-pstn,s,1)</span>.<span style="font-family: "courier new";"></span></div>
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When we're done the section should look like this:</div>
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<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 387pt;" valign="top" width="516"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">[from-sip-external]<br />;give external sip users congestion and hangup<br />;exten => _.,1,AbsoluteTimeout(15)<br />;exten => _.,2,Congestion<br />;exten => _.,3,Hangup</span></div>
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<span style="font-family: "courier new";"><br />exten => _.,1,Wait(1)<br />exten => _.,2,Goto(from-pstn,s,1)</span></div>
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When you're finished and you've saved your changes, you need to click the link at the top of the config edit screen <span class="SpellE">labled</span> <span style="font-family: "courier new";">Re-Read Configs</span>. This will cause Asterisk to reload its configuration files and make any changes take effect.</div>
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Test Incoming Calls</h3>
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Now is the moment of truth. If everything went smoothly, you should now be ready to test an inbound call to your AAH system. From a phone (POTS or cell, either will do) dial your BroadVoice number. You <i>should</i> hear your greeting! At this point you'll probably be giddy that it's all working. Go ahead and listen to your greeting a few times (dial 9 to repeat it). Then either dial "1" or your extension and your SIP phone should start ringing. You now have Asterisk@Home configured and working with your BroadVoice account.</div>
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Set up Passwords</h3>
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You need to change the default passwords, so that people do not break into your system.  There are a variety of passwords, used for different things.</div>
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AMP allows you to assign different login accounts to access the Maintenance section separately from the other sections (Setup, Reports, and Panel).  If you log into AMP with the maintenance account, you will be able to access everything, but if you log in with the admin account, you will have to log in also to get to the maintenance section.</div>
<div align="center">
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Username</div>
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Default Password</div>
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Command to Change Password</div>
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Usage</div>
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<span style="font-family: "courier new";">root</span></div>
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<span style="font-family: "courier new";">password</span></div>
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<span style="font-family: "courier new";">passwd</span></div>
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<span style="font-family: "courier new";">log in to console</span></div>
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<span style="font-family: "courier new";">wwwadmin</span></div>
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<span style="font-family: "courier new";">password</span></div>
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<span style="font-family: "courier new";">password-amp</span></div>
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<span style="font-family: "courier new";">log into AMP GUI, access Setup, Reports, Panel</span></div>
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<span style="font-family: "courier new";">maint</span></div>
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<span style="font-family: "courier new";">password</span></div>
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<span style="font-family: "courier new";">password-maint</span></div>
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<span style="font-family: "courier new";">log into AMP Maintenance section</span></div>
</td></tr>
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<span style="font-family: "courier new";">meetme</span></div>
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<span style="font-family: "courier new";">??</span></div>
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<span style="font-family: "courier new";">passwd-meetme</span></div>
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<span style="font-family: "courier new";">log into Web Meetme Control</span></div>
</td></tr>
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<span style="font-family: "courier new";">admin</span></div>
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<span style="font-family: "courier new";">password</span></div>
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<span style="font-family: "courier new";">passwd admin</span></div>
</td><td style="border-style: none solid solid none; border-width: medium 1pt 1pt medium; padding: 0in 5.4pt; width: 153.75pt;" valign="top" width="205"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">for checking system mail (Maintenance / Web Mail login)</span></div>
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</div>
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To change these passwords, you must log into the console as root, either on the physical console or using a <span class="GramE">ssh</span> client such as putty (which can be obtained from <a href="http://www.chiark.greenend.org.uk/~sgtatham/putty/" style="color: blue; text-decoration: underline;">http://www.chiark.greenend.org.uk/~sgtatham/putty/</a>) from a windows platform or using ssh from a Linux platform.</div>
<h2 style="font-family: Arial; font-size: 14pt; font-style: italic; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Troubleshooting</h2>
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If things did not work the first time, you can track down the trouble using the Asterisk console.  This can be used to display SIP messages coming through the system, as well as steps executed in interpreting the dial plan instructions. </div>
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If you are more comfortable using Ethereal, you can use that instead (or also) to see the SIP messages.  I will not describe how to download or configure it, other than to point out that it can be obtained from<a href="http://www.ethereal.com/" style="color: blue; text-decoration: underline;">http://www.ethereal.com</a>.</div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
To use the Asterisk console, go to the Linux console or enter through <span class="GramE">ssh</span>.  Remember the username is <span style="font-family: "courier new";">root</span> and the password is <span style="font-family: "courier new";">password</span>.  Give the command:</div>
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<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 221.4pt;" valign="top" width="295"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">asterisk –<span class="SpellE">vvvvr</span></span></div>
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<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
This will attach to the console with verbose mode set.  Give the command</div>
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<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 221.4pt;" valign="top" width="295"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">sip debug</span></div>
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<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span class="GramE">to</span> enable SIP debugging.  You can turn off SIP debugging using the command</div>
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<span style="font-family: "courier new";">sip no debug</span></div>
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<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<br /></div>
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Other console commands are summarized at <a href="http://www.voip-info.org/wiki-Asterisk+CLI" style="color: blue; text-decoration: underline;">http://www.voip-info.org/wiki-Asterisk+CLI</a>.</div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
<span class="SpellE">Softphone</span> will not register</h3>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
XLite registers by sending a REGISTER message, which is challenged.  It then sends another REGISTER with credentials, which is accepted.  If you do not see these REGISTER messages, then XLite proxy setting is probably not right.  If you see the challenge but it still rejects the REGISTER with credentials, the passwords are probably mismatched.  The password <span style="font-family: "courier new";">abc123</span> should be used in both places. Also, be sure the phone "Authorization user" is set to 200.</div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Outgoing calls</h3>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
If you are having trouble making outgoing calls, check the username and secret in the IAX2 trunk.  Make sure you can ping the host address listed there.  The username is a 4 digit number that you find from your account settings on VoipJet. </div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
On a message trace, you should see an INVITE going out, followed by receipt of TRYING and OK (when the far end answers).  If there is no response, then perhaps you have the wrong IP address.  If the INVITE is immediately followed by a 4XX or 5XX error, maybe the username or password is wrong.</div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Incoming calls</h3>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
If the incoming BroadVoice trunk is set up properly, Asterisk will register successfully, and will renew the registration every 10 seconds.  Look for an outgoing REGISTER, a challenge, another outgoing REGISTER with credentials, and an incoming 200 OK.  If you get no response, or either REGISTER yields an error reply, then look at the host, secret, and user fields from the SIP trunk. </div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
If the REGISTER succeeds, then when you call (from a POTS phone) you should see an INVITE being received.  If you have not properly changed <span style="font-family: "courier new";">extensions.conf</span>, then the system will reply with an error rather than with TRYING.  Look at the debug output to see how the INVITE is being processed.  You should see the <span style="font-family: "courier new";">Wait</span> and the <span style="font-family: "courier new";">GoTo</span> that you entered into <span style="font-family: "courier new";">extensions.conf</span>.  If you see these, and the call is still not answered, you might want to change <span style="font-family: "courier new";">Incoming Calls</span> to route the call to a specific phone, for instance to "<span style="font-family: "courier new";">basement <200></span>".  This will bypass the Digital Receptionist and the Ring Group for now.  If this fixes things, look at those two screens. If not, make sure the "<span style="font-family: "courier new";">context=from-pstn</span>" is present in the incoming trunk user details.  If this still does not fix things, you will have to follow the trace as it steps through the contexts and steps in <span style="font-family: "courier new";">extensions.conf</span>.</div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
More</h3>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
This section describes some things you might want to do to further customize your system.  I am experimenting myself, and presenting the results here.  These instructions might not be the most effective way to achieve the desired results.  If you find a better way, please let me know.  The instructions in this section are briefer, reflecting your increased level of experience with AAH.</div>
<h4 style="font-family: 'Times New Roman'; font-size: 14pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Weather for external callers</h4>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
You can modify the digital attendant so that it can read the weather to callers.  Remember to modify your digital attendant announcement recording if desired ("<i>press 2 for weather</i>").</div>
<ol start="1" style="margin-bottom: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Modify the Digital Receptionist.  For "Number of options" give "2".  On the next screen, for option 2, give "Custom App" and enter</li>
</ol>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 77.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 3.5in;" valign="top" width="336"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">custom weather,s,1</span></div>
</td></tr>
</tbody></table>
<ol start="2" style="margin-bottom: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Go into "Maintenance", "Config Edit", <span class="GramE">"</span>extensions_custom.conf".  At the end, add:</li>
</ol>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 77.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 365.4pt;" valign="top" width="487"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">[custom-weather]<br />exten => s,1,</span> <span style="font-family: "courier new";">Goto(from-internal-custom,*61,1)</span></div>
</td></tr>
</tbody></table>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<br /></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
You can customize the weather for your own city <span class="GramE">be</span> editing <span style="font-family: "courier new";">/var/lib/asterisk/agi-bin/weather.agi</span>.  You need to edit <span style="font-family: "courier new";">$<span class="SpellE">custpath</span></span> and <span style="font-family: "courier new";">$filename</span>.  Don't forget to modify the announcement a few lines down.  You can ftp to <span style="font-family: "courier new";"><a href="http://weather.noaa.gov/">weather.noaa.gov</a></span> and <span style="font-family: "courier new";">cd</span> to <span style="font-family: "courier new";">data/forecasts/city</span> to see the state/city combinations that are available.</div>
<h4 style="font-family: 'Times New Roman'; font-size: 14pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Outgoing BroadVoice</h4>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
You can set up the BroadVoice trunk to carry outgoing calls.  Since you get 100 "free" minutes per month, you might want to use these up first, before switching over to VoipJet.  It would be great if the system could do this for you, but for now there is no easy way to do this.  So instead, let's up a dial-9 trunk for BroadVoice.</div>
<ol start="1" style="margin-bottom: 0in; margin-top: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Open the BroadVoice trunk.  Add "Peer Settings" of:<br />
<br />
<span style="font-family: "courier new";"></span></li>
</ol>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 77.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 279pt;" valign="top" width="372"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">allow=ulaw<br />authuser=7327570239<br />callerid=7327570239<br />disallow=all<br />dtmfmode=inband<br />fromdomain=<a href="http://sip.broadvoice.com/">sip.broadvoice.com</a><br />fromuser=7327570239<br />host=<a href="http://sip.broadvoice.com/">sip.broadvoice.com</a><br />insecure=very<br />qualify=yes<br />secret=*******<br />type=peer<br />username=7327570239</span></div>
</td></tr>
</tbody></table>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.25in;">
<br /></div>
<ol start="2" style="margin-bottom: 0in; margin-top: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Go to "Outbound Routing".  Add a new route.  Name it "outgoingBroadVoice."  Add the following dial patterns.  These mimic the local dialing rules, which permit "1" followed by a ten digit number, or "732" followed by a seven digit number.  An initial "9<span class="GramE">" selects</span> the broadvoice trunk, but it is stripped off before the number is dialed.<br />
<br />
<span style="font-family: "courier new";"></span></li>
</ol>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 77.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 207pt;" valign="top" width="276"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">9|1NXXNXXXXXX<br />9|732NXXXXXX</span></div>
</td></tr>
</tbody></table>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<br /></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
Set the "Trunk sequence" to SIP/broadvoice."</div>
<h4 style="font-family: 'Times New Roman'; font-size: 14pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Set Up Voice Mail</h4>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
By default, when you set up an extension, and add an email address in the "email address" field, then when the system records a message, it sends email containing a link that will let the recipient listen to the voice message on their computer (through the browser).  For this to work, you must open up your Asterisk machine to browser traffic from outside the router's firewall.  I leave it to the reader to figure out this one.</div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
Once you do, you still need to change the email, which contains a link to the mail reading web page. </div>
<ol start="1" style="margin-bottom: 0in; margin-top: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Go to "Maintenance" and then "Config Edit" and choose "<span class="SpellE">vm_email.inc</span>". </li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Replace the string "192.168.1.101" with the <i>external</i> name or IP address of your Asterisk system.  In other words, you need to form a URL that can be used to browse to the Asterisk system from where you will be reading the mail, which is probably outside your NAT router.</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">After you click "Update" do not forget to select "Re-Read <span class="SpellE">Configs</span>".</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Test it by calling extension 200, leaving voice mail, receiving the email, and following the link.</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">You will be prompted to log in.  Use extension number 200, and its password "1234".  You can then retrieve and listen to the email.</li>
</ol>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
You may notice a few things wrong with the voice mail reader application.  I guess this means that it is not really finished, although it appears to basically work.  I noted that the initial login does not fill in your extension, as it should.  Also the image links are broken.  Maybe the next version will clean up these details.</div>
<h4 style="font-family: 'Times New Roman'; font-size: 14pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Add Free World Dialup IAX Trunk</h4>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<img align="right" height="72" hspace="12" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image027.jpg" width="338" />Free World Dialup (FWD, at <a href="http://www.freeworlddialup.com/" style="color: blue; text-decoration: underline;">http://www.freeworlddialup.com/</a>) is a free VoIP service provider.  Using it, you can call other FWD users, and through cooperative arrangements with other service providers, subscribers on many other VoIP networks.  In addition, you can use their gateways to call to the PSTN (but only toll free numbers, since someone still has to pay for other kinds of calls).</div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<br /></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
You first need to sign up for a FWD "phone number," which you can do at the web site listed above.  Be sure to enable IAX service while you are there.  My number is 679263.  I will use this in the examples below, but you should replace it with your own number and password, as appropriate. </div>
<ol start="1" style="margin-bottom: 0in; margin-top: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Open the Trunks page, select "Add Trunk" and select "Add IAX2 Trunk".  Enter the following information.  I have chosen to use a prefix of "393" (FWD) to select the FWD trunk.  You can pick a different trunk prefix if you want.<br />
<br />
</li>
</ol>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 41.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 5.25in;" valign="top" width="504"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">Outbound Caller ID: Charles Hayden <679263><br />Maximum <span class="SpellE">Chanels</span>: 1</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">Trunk Name: fwd<br />PEER Details:</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">allow=ulaw</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">auth=md5</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">disallow=all</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">host=<a href="http://iax2.fwdnet.net/">iax2.fwdnet.net</a></span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">qualify=yes</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">secret=<your FWD password></span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">type=peer</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">username=679263</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">USER Context: <span class="SpellE">iaxfwd</span></span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">USER Details:</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">allow=ulaw</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">auth=<span class="SpellE">rsa</span></span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">context=from-pstn</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">disallow=all</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span class="SpellE"><span style="font-family: "courier new";">inkeys</span></span><span style="font-family: "courier new";">=<span class="SpellE">freeworlddialup</span></span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">secret=<your FWD password></span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">type=user</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">Register String:</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt 0.5in;">
<span style="font-family: "courier new";">679263:<your FWD password>@<a href="http://iax2.fwdnet.net/">iax2.fwdnet.net</a></span></div>
</td></tr>
</tbody></table>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 12pt 0in 0.0001pt;">
<img border="0" height="864" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image029.jpg" width="504" /></div>
<ol start="2" style="margin-bottom: 0in; margin-top: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Open the "DID Routes" page, and select "Ad Route".  The following information:<br />
<br />
</li>
</ol>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 41.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 5.25in;" valign="top" width="504"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">DID Number: 679263</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">Ring Group: #1</span></div>
</td></tr>
</tbody></table>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 12pt 0in 0.0001pt;">
<br />
<img border="0" height="441" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image031.jpg" width="575" /><br />
<br /></div>
<ol start="3" style="margin-bottom: 0in; margin-top: 0in;" type="1">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Open the "Outbound Routing" page and select "Add Route".  Enter the following information:<br />
<br />
</li>
</ol>
<table border="1" cellpadding="0" cellspacing="0" class="MsoTableGrid" style="background-attachment: scroll; background-color: #e0e0e0; background-image: none; background-position: 0% 50%; border-collapse: collapse; border-style: none; border-width: medium; margin-left: 41.4pt;"><tbody>
<tr><td style="border: 1pt solid windowtext; padding: 0in 5.4pt; width: 5.25in;" valign="top" width="504"><div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">Route Name: <span class="SpellE">outgoingFWD</span></span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">Dial Patterns: 393|.</span></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span style="font-family: "courier new";">Trunk Sequence: IAX2/fwd</span></div>
</td></tr>
</tbody></table>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 12pt 0in 0.0001pt;">
<img border="0" height="461" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image033.jpg" width="575" /><br />
<br /></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 12pt 0in 0.0001pt;">
After you submit and apply these changes, you can go to the "Maintenance/Asterisk Info" page and check the IAX2 Sip Registry and the IAX2 Peers.  These should both show the trunk you set up with State<span style="font-family: "courier new";">Registered</span> and Status <span style="font-family: "courier new";">OK</span>.  You can test FWD by dialing "393612" to hear the time.  You can make an incoming call to your Asterisk system using the FWD page that used to sign up for IAX service.</div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<br /></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<br /></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
<span class="SpellE"><b>ArtDio</b></span><b></b></div>
<div class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">
There are many choices for phones that you can use with the Asterisk system.  Among the possibilities are IAX phones, SIP phones, SIP softphones, and analog phones with a SIP/analog converter.  Such a converter should provide a FXO interface.  Sources of phones and converters are listed below.</div>
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<img align="right" height="99" hspace="12" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image035.jpg" width="152" />I purchased some <span class="SpellE">ArtDio</span> phones, since they were the least expensive SIP phones I could find.  Adding new extensions for the phones is straightforward – just like the others.  Administering the phone itself was a little more difficult.</div>
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The procedure for setting up the phone consisted of two steps.</div>
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<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Plug the phone in, log on to it, and set its IP address, net mask, and default gateway parameters.</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">Browse to the phone and set up its SIP gateway, its name and password, and other SIP parameters.  You will know these are right when the phone can register.  In the case of the <span class="SpellE">ArtDio</span> phone the critical step <span class="GramE">was knowing</span> that there were two passwords, and that only the "super user" password ("12345678") would allow access to the key SIP configuration parameters. </li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">The fields that I had to change on the <span class="SpellE">ArtDio</span> are as follows (assuming the extension is 201 and the IP address of the phone is 192.168.0.60):</li>
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<span style="font-family: "courier new";">Use service: checked<br />Service type: common<br />Service <span class="SpellE">addr</span>: 192.168.0.40</span></div>
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<span style="font-family: "courier new";">Service id: 192.168.0.40<br />Phone number: 201<br />Account: 201<br />PIN: abc123<br />Call type: advanced</span></div>
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If you do not see the screen below when you browse to the phone's IP address, but instead see a simpler screen, then it means that you have logged in with the phones user password, not its <span class="SpellE">superuser</span> password.</div>
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<img border="0" height="350" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image037.jpg" width="575" /></div>
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<b>Sipura</b></div>
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<img align="right" height="74" hspace="12" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image039.jpg" width="163" />Once I had a few phones set up, I realized that there are places in my home where I wanted a phone, but where I did not have Ethernet wiring.  There are several ways to address this:</div>
<ul style="margin-bottom: 0in; margin-top: 0in;" type="disc">
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">rewire my house, putting Ethernet jacks all over</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">use an analog phone adapter and set up cordless phones as extensions</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">get WiFi SIP phones, used in conjunction with my existing wireless infrastructure</li>
<li class="MsoNormal" style="font-family: 'Times New Roman'; font-size: 12pt; margin: 0in 0in 0.0001pt;">use an analog phone adapter and existing phone wiring and phones</li>
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The simplest, least <span class="GramE">expensive,</span> and least disruptive choice was to use an analog phone adapter with a cordless hone.  The Sipura adapter is widely available on eBay.  It was distributed by various IP telephony service providers. I guess people who tried it and did not like it are selling their used adapters.  Just make sure you get one that is "unlocked".  Apparently Vonage adapters are set up so that you cannot modify the critical SIP parameters, and are not usable with Asterisk.  Other service providers such as <span class="GramE">BroadVoice,</span> do not lock their devices.</div>
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I bought a used Sipura SPA-2002.  I have also tried this with a Sipura 1000 and it works about the same.  Setting it up was pretty easy. </div>
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1.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>Set up an extension, as before, in the AMP Setup/Extensions screen. </div>
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2.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>Plug it into Ethernet and connect a phone.  When you apply power, it gets an IP address from DHCP. </div>
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3.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>You need to get the IP address.  Pick up the phone, dial "<span style="font-family: "courier new";">****</span>" and then dial "<span style="font-family: "courier new";">110#</span>". It will speak the IP address.</div>
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4.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span> Browse to the address as follows: <span style="font-family: "courier new";"><a href="http://%3Cipaddres%3E/admin/advanced" style="color: blue; text-decoration: underline;"><span style="text-decoration: none;">http://<ip_addres>/admin/advanced</span></a></span></div>
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5.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>If you have purchased a used unit, it would be a good idea to clear out any old settings first.  On Sipura devices, you can do a factory reset by:</div>
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a.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">       </span>Dial "<span style="font-family: "courier new";">****</span>" and verify that you hear a voice prompt.</div>
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b.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>Dial "<span style="font-family: "courier new";">RESET#</span>" (<span style="font-family: "courier new";">73738#</span>).</div>
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c.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">       </span>Dial "<span style="font-family: "courier new";">1</span>" to confirm.</div>
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6.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>On the "System" tab, enter the fixed IP address you want to assign the device.<br />
<img border="0" height="328" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image041.jpg" width="575" /><br />
When you are done, submit the changes.</div>
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7.<span style="font-family: "times new roman"; font-size: 7pt; font-style: normal; font-variant: normal; line-height: normal;">      </span>Using the new IP address, browse to the advanced admin screen again.  This time, select the "Line1" tab.  Enter the proxy information (the IP address of the Asterisk system) and the subscriber information (the extension number and password you set up in step 1).<br />
<img border="0" height="318" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image043.jpg" width="575" /><br />
You should be able to use the phone at this point.  Go to AMP's Maintenance/Asterisk Info screen to check that the new extension is registered.</div>
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Router</h4>
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<img align="right" height="182" hspace="12" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image044.jpg" width="170" />Once you have your system set up and are using it for phone service, you may notice that the sound quality is terrible when you are doing other things on your broadband connection, such as large FTP transfers.  This depends a lot on your connection speed.  In my case, which I imagine is pretty common these days, I have only 128K bits per second uplink speed.  For comparison, with the standard G711 coder, one phone call takes 64K bits, half the bandwidth.  Unless you do something, FTP will take enough bandwidth to make telephone calls impossible.  What can you do about <span class="GramE">this ?</span>  Your router may be able to help, by giving traffic coming to and from the Asterisk box priority. I have a <span class="SpellE">Linksys</span> WRT54G, a widely available combo router, 4 port switch, and wireless access point.  It has a way to assign priority to different traffic in various ways.  I set it up to give priority to a given switch port, where I have the Asterisk system plugged in. </div>
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<span class="GramE">To set this up, browse to your router, log in, and go to "Applications and Gaming" and then to "<span class="SpellE">QoS</span>".</span>  I plugged my Asterisk system into port two, so I set up:</div>
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<span class="SpellE"><span style="font-family: "courier new";">QoS</span></span><span style="font-family: "courier new";">: </span><span style="font-family: "courier new";">Enable</span><span style="font-family: "courier new";"><br />
</span><span style="font-family: "courier new";">Port</span><span style="font-family: "courier new";"> 2: Priority High</span></div>
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<img border="0" height="360" src="http://chayden.net/Asterisk/SeUpAsteriskAtHome_files/image046.jpg" width="575" /></div>
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Next</h2>
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The AAH people have provided you with a ton of pre-configured features. You can start reading through the main configuration files to see some of the cool things you can do with AAH and your BroadVoice and VoipJet accounts. You might want to set up the BroadVoice trunk to do outgoing calls, using a prefix.  It would be nice if it could use BroadVoice for outgoing calls until the 100 minutes are used up, <span class="GramE">then</span> switch to VoipJet.  If you figure out how to do this, let me know.</div>
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You can make a backup of your configuration files by backing up the files in the <span style="font-family: "courier new";">/etc/asterisk</span> directory of your AAH server. The AMP "<span style="font-family: "courier new";">Backup</span>" command on the left can be used to set up periodic backups.</div>
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If you are interested in making international calls, add some more patterns in "<span style="font-family: "courier new";">Outbound Routing</span>".  Commonly used dial patterns are preconfigured, so you only have to select them.</div>
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Read through the <span style="font-family: "courier new";">extensions.conf</span> and see what all is already configured for you to test and work with.</div>
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Further Reference</h2>
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Here are some places to go to find out more about AAH, Asterisk, and the components that make it up.</div>
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<span class="Heading3Char" style="font-family: "arial"; font-weight: bold;"><span style="font-size: 13pt;">Service Providers</span></span></div>
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BroadVoice <a href="http://broadvoice.com/" style="color: blue; text-decoration: underline;">http://broadvoice.com</a><br />
VoipJet <a href="http://voipjet.com/" style="color: blue; text-decoration: underline;">http://voipjet.com</a><br />
VoicePulse <a href="http://connect.voicepulse.com/" style="color: blue; text-decoration: underline;">http://connect.voicepulse.com</a><br />
TelaSip <a href="http://www.telasip.com/index.html?PARTNER=coupon4voip" style="color: blue; text-decoration: underline;">http://www.telasip.com/index.html?PARTNER=coupon4voip</a><br />
Free World Dialup <a href="http://www.freeworlddialup.com/" style="color: blue; text-decoration: underline;">http://www.freeworlddialup.com</a><br />
LibreTel <a href="http://www.libretel.com/" style="color: blue; text-decoration: underline;">http://www.libretel.com</a></div>
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Software</h3>
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Asterisk@Home and Asterisk:</div>
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Asterisk@Home <a href="http://asteriskathome.sourceforge.net/" style="color: blue; text-decoration: underline;">http://asteriskathome.sourceforge.net/</a><br />
Asterisk <a href="http://www.asterisk.org/" style="color: blue; text-decoration: underline;">http://www.asterisk.org/</a><br />
Asterisk@Home Handbook <a href="http://asteriskathome.sourceforge.net/handbook/" style="color: blue; text-decoration: underline;">http://asteriskathome.sourceforge.net/handbook</a><br />
Asterisk@Home support <a href="http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home" style="color: blue; text-decoration: underline;">http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home</a><br />
Asterisk Support <a href="http://www.voip-info.org/tiki-index.php?page=Asterisk" style="color: blue; text-decoration: underline;">http://www.voip-info.org/tiki-index.php?page=Asterisk</a><br />
Asterisk@Home info <a href="http://geekgazette.com/index.php?option=com_frontpage&Itemid=1&limit=4&limitstart=8" style="color: blue; text-decoration: underline;">http://geekgazette.com/index.php?option=com_frontpage&Itemid=1&limit=4&limitstart=8</a></div>
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AMP (Asterisk Management Portal)</div>
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AMP Home <a href="http://amp.coalescentsystems.ca/" style="color: blue; text-decoration: underline;">http://amp.coalescentsystems.ca/</a></div>
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FOP (Flash Operator Panel)</div>
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FOP Home <a href="http://www.asternic.org/" style="color: blue; text-decoration: underline;">http://www.asternic.org/</a></div>
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Sugar CRM</div>
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What it is and how to set it up <a href="http://www.sugarcrm.com/home/" style="color: blue; text-decoration: underline;">http://www.sugarcrm.com/home</a></div>
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CentOS</div>
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            A free enterprise OS <a href="http://centos.org/" style="color: blue; text-decoration: underline;">http://centos.org</a></div>
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Phones</div>
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<span lang="SV">XLite <a href="http://xten.com/index.php?menu=products&smenu=xlite" style="color: blue; text-decoration: underline;">http://xten.com/index.php?menu=products&smenu=xlite</a><br />SipXPhone </span><a href="http://www.sipfoundry.org/sipXphone/index.html" style="color: blue; text-decoration: underline;"><span lang="SV">http://www.sipfoundry.org/sipXphone/index.html</span></a><span lang="SV"></span></div>
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Tools</div>
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            Ethereal <a href="http://www.ethereal.com/" style="color: blue; text-decoration: underline;">http://www.ethereal.com</a><br />
            Putty <a href="http://www.chiark.greenend.org.uk/~sgtatham/putty/" style="color: blue; text-decoration: underline;">http://www.chiark.greenend.org.uk/~sgtatham/putty/</a></div>
<h3 style="font-family: Arial; font-size: 13pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Phones</h3>
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Buy phones <a href="http://www.voiplink.com/" style="color: blue; text-decoration: underline;">http://www.voiplink.com/</a><br />
Buy phones <a href="http://www.voipsupply.com/" style="color: blue; text-decoration: underline;">http://www.voipsupply.com/</a><br />
 Buy phones <a href="https://www.thevoipconnection.com/" style="color: blue; text-decoration: underline;">https://www.thevoipconnection.com/</a><br />
Buy Phones <a href="http://store.sipphonestore.com/" style="color: blue; text-decoration: underline;">http://store.sipphonestore.com/</a><br />
XLite softphone <a href="http://www.xten.com/" style="color: blue; text-decoration: underline;">http://www.xten.com/</a><br />
SNOM phones <a href="http://www.snom.com/phones.html?&L=1" style="color: blue; text-decoration: underline;">http://www.snom.com/phones.html?&L=1</a><br />
ArtDio phones <a href="http://www.artdio.com.tw/en/ipf.htm" style="color: blue; text-decoration: underline;">http://www.artdio.com.tw/en/ipf.htm</a><br />
<span class="SpellE">Grandstream</span> phones <a href="http://www.grandstream.com/y-product.htm" style="color: blue; text-decoration: underline;">http://www.grandstream.com/y-product.htm</a><br />
Cisco phones <a href="http://www.cisco.com/en/US/products/hw/phones/ps379/" style="color: blue; text-decoration: underline;">http://www.cisco.com/en/US/products/hw/phones/ps379/</a><br />
<span class="SpellE">Avaya</span> phones <a href="http://www.avaya.com/gcm/master-usa/en-us/products/offers/4600_series_ip_telephones.htm&View=ProdTechSpec" style="color: blue; text-decoration: underline;">http://www.avaya.com/gcm/master-usa/en-us/products/offers/4600_series_ip_telephones.htm&View=ProdTechSpec</a></div>
<h2 style="font-family: Arial; font-size: 14pt; font-style: italic; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Acknowledgements</h2>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
A preliminary version of this set of instructions was produced by Jacob Cazell (<a href="http://www.cazz.org/" style="color: blue; text-decoration: underline;">http://www.cazz.org/</a>).  It helped immensely in getting my own system set up properly the first time, and inspired this document.  A few of the sections here incorporate material that first appeared there.</div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
I will be adding to these instructions from time to time as I experiment with additional features, service providers, and subsequent releases.  If you have suggestions, corrections, or comments on these instructions, I welcome your feedback.  Please send commands to <a href="mailto:cchayden@gmail.com" style="color: blue; text-decoration: underline;">cchayden@gmail.com</a></div>
<h1 style="font-family: Arial; font-size: 16pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Appendix 1 – Feature Codes</h1>
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*411</div>
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Directory</div>
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*43</div>
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Echo Test</div>
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*60</div>
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Time</div>
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*61</div>
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Weather</div>
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*62</div>
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Schedule wakeup call</div>
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*65</div>
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Festival test (your extension is XXX)</div>
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*69</div>
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Last Called ID</div>
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*70</div>
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Activate Call Waiting (deactivated by default)</div>
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*71</div>
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Deactivate Call Waiting</div>
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*72</div>
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Call Forwarding System</div>
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*73</div>
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Disable Call Forwarding</div>
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*77</div>
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IVR Recording</div>
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*78</div>
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Enable Do-Not-Disturb</div>
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*79</div>
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Disable Do-Not-Disturb</div>
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*8</div>
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Pickup</div>
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*90</div>
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Call Forward on Busy</div>
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*91</div>
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Disable Call Forward on Busy</div>
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*97</div>
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Message Center (does no ask for extension)</div>
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*98</div>
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Enter Message Center</div>
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*99</div>
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Playback IVR Recording</div>
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666</div>
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Test Fax</div>
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700</div>
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Park</div>
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7777</div>
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Simulate incoming call</div>
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888</div>
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Barge In</div>
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<br /></div>
<div style="font-family: 'Times New Roman'; font-size: 12pt; margin-left: 0in; margin-right: 0in;">
<br /></div>
<h1 style="font-family: Arial; font-size: 16pt; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
Appendix 2 – Asterisk CLI commands</h1>
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This is a selected summary of the Asterisk CLI commands.  These commands can be entered on the Asterisk console, which you access by running:<br />
<br /></div>
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<span style="font-family: "courier new";">asterisk –r</span></div>
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<br /></div>
<h2 style="font-family: Arial; font-size: 14pt; font-style: italic; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
General commands<span style="color: #436976;"></span></h2>
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<b><span style="color: black;">!<command></span></b><span style="color: black;">: Executes a given shell command</span></div>
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<span class="GramE"><b><span style="color: black;">abort</span></b></span><b><span style="color: black;"> halt</span></b><span style="color: black;">: Cancel a running halt</span></div>
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<span class="GramE"><b><span style="color: black;">add</span></b></span><b><span style="color: black;"> extension</span></b><span style="color: black;">: Add new extension into context</span></div>
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<span class="GramE"><b><span style="color: black;">add</span></b></span><b><span style="color: black;"> ignorepat</span></b><span style="color: black;">: Add new ignore pattern</span></div>
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<span class="GramE"><b><span style="color: black;">add</span></b></span><b><span style="color: black;"> indication</span></b><span style="color: black;">: Add the given indication to the country</span></div>
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<span class="GramE"><b><span style="color: black;">debug</span></b></span><b><span style="color: black;"> channel</span></b><span style="color: black;">: Enable debugging on a channel</span></div>
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<span class="GramE"><b><span style="color: black;">dont</span></b></span><b><span style="color: black;"> include</span></b><span style="color: black;">: Remove a specified include from context</span></div>
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<span class="GramE"><b><span style="color: black;">help</span></b></span><span style="color: black;">: Display help list, or specific help on a command</span></div>
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<span class="GramE"><b><span style="color: black;">include</span></b></span><b><span style="color: black;"> context</span></b><span style="color: black;">: Include context in other context</span></div>
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<span class="GramE"><b><span style="color: black;">load</span></b></span><span style="color: black;">: Load a dynamic module by name</span></div>
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<span class="GramE"><b><span style="color: black;">logger</span></b></span><b><span style="color: black;"> reload</span></b><span style="color: black;">: Reopen log files. <span class="GramE">Use after rotating the log files.</span></span></div>
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<span class="GramE"><b><span style="color: black;">no</span></b></span><b><span style="color: black;"> debug channel</span></b><span style="color: black;">: Disable debugging on a channel</span></div>
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<span class="GramE"><b><span style="color: black;">pri</span></b></span><b><span style="color: black;"> debug span</span></b><span style="color: black;">: Enables PRI debugging on a span</span></div>
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<span class="GramE"><b><span style="color: black;">pri</span></b></span><b><span style="color: black;"> intense debug span</span></b><span style="color: black;">: Enables REALLY INTENSE PRI debugging</span></div>
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<span class="GramE"><b><span style="color: black;">pri</span></b></span><b><span style="color: black;"> no debug span</span></b><span style="color: black;">: Disables PRI debugging on a span</span></div>
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<span class="GramE"><b><span style="color: black;">remove</span></b></span><b><span style="color: black;"> extension</span></b><span style="color: black;">: Remove a specified extension</span></div>
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<span class="GramE"><b><span style="color: black;">remove</span></b></span><b><span style="color: black;"> ignorepat</span></b><span style="color: black;">: Remove ignore pattern from context</span></div>
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<span class="GramE"><b><span style="color: black;">remove</span></b></span><b><span style="color: black;"> indication</span></b><span style="color: black;">: Remove the given indication from the country</span></div>
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<span class="GramE"><b><span style="color: black;">save</span></b></span><b><span style="color: black;"> dialplan</span></b><span style="color: black;">: <i>Overwrites</i> your current extensions.conf file with an exported version based on the current state of the dialplan. A backup copy of your old extensions.conf is <i>not</i> saved. The initial values of global variables defined in the [globals] category retain their previous initial values; the current values of global variables are not written into the new extensions.conf. Using "save dialplan" will result in losing any comments in your current extensions.conf.</span></div>
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<span class="GramE"><b><span style="color: black;">set</span></b></span><b><span style="color: black;"> verbose</span></b><span style="color: black;">: Set level of verboseness</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> agents</span></b><span style="color: black;">: Show status of agents</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> applications</span></b><span style="color: black;">: Shows registered applications</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> application</span></b><span style="color: black;">: Describe a specific application</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> channel</span></b><span style="color: black;">: Display information on a specific channel</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> channels</span></b><span style="color: black;">: Display information on channels</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> codecs</span></b><span style="color: black;">: Display information on codecs</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> conferences</span></b><span style="color: black;">: Show status of conferences</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> dialplan</span></b><span style="color: black;">: Show dialplan</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> image formats</span></b><span style="color: black;">: Displays image formats</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> indications</span></b><span style="color: black;">: Show a list of all country/indications</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> locals</span></b><span style="color: black;">: Show status of local channels</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> manager command</span></b><span style="color: black;">: Show manager commands</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> manager connect</span></b><span style="color: black;">: Show connected manager users</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> parkedcalls</span></b><span style="color: black;">: Lists parked calls</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> queues</span></b><span style="color: black;">: Show status of queues</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> switches</span></b><span style="color: black;">: Show alternative switches</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> translation</span></b><span style="color: black;">: Display translation matrix</span></div>
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<span class="GramE"><b><span style="color: black;">soft</span></b></span><b><span style="color: black;"> hangup</span></b><span style="color: black;">: Request a hangup on a given channel</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> voicemail users</span></b><span style="color: black;">: List defined voicemail boxes</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> voicemail zones</span></b><span style="color: black;">: List zone message formats</span></div>
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Server management<span style="color: #436976;"></span></h2>
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<span class="GramE"><b><span style="color: black;">restart</span></b></span><b><span style="color: black;"> gracefully</span></b><span style="color: black;">: Restart Asterisk gracefully</span></div>
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<span class="GramE"><b><span style="color: black;">restart</span></b></span><b><span style="color: black;"> now</span></b><span style="color: black;">: Restart Asterisk immediately</span></div>
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<span class="GramE"><b><span style="color: black;">restart</span></b></span><b><span style="color: black;"> when convenient</span></b><span style="color: black;">: Restart Asterisk at empty call volume</span></div>
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<span class="GramE"><b><span style="color: black;">reload</span></b></span><span style="color: black;">: Reload configuration</span></div>
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<span class="GramE"><b><span style="color: black;">stop</span></b></span><b><span style="color: black;"> gracefully</span></b><span style="color: black;">: Gracefully shut down Asterisk</span></div>
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<span class="GramE"><b><span style="color: black;">stop</span></b></span><b><span style="color: black;"> now</span></b><span style="color: black;">: Shut down Asterisk immediately</span></div>
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<span class="GramE"><b><span style="color: black;">stop</span></b></span><b><span style="color: black;"> when convenient</span></b><span style="color: black;">: Shut down Asterisk at empty call volume</span></div>
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<span class="GramE"><b><span style="color: black;">extensions</span></b></span><b><span style="color: black;"> reload?</span></b><span style="color: black;">: Reload extensions and <i>only</i> extensions</span></div>
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<span class="GramE"><b><span style="color: black;">unload</span></b></span><span style="color: black;">: Unload a dynamic module by name</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> modules</span></b><span style="color: black;">: List modules and info about them</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> uptime</span></b><span style="color: black;">: Show uptime information</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> version</span></b><span style="color: black;">: Display Asterisk version info</span></div>
<h2 style="font-family: Arial; font-size: 14pt; font-style: italic; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
AGI commands<span style="color: #436976;"></span></h2>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> agi</span></b><span style="color: black;">: Show AGI commands or specific help</span></div>
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<span class="GramE"><b><span style="color: black;">dump</span></b></span><b><span style="color: black;"> agihtml</span></b><span style="color: black;">: Dumps a list of agi command in html format</span></div>
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Database handling commands<span style="color: #436976;"></span></h2>
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<span class="GramE"><b><span style="color: black;">database</span></b></span><b><span style="color: black;"> </span></b><b><span style="color: black;">del</span></b><span style="color: black;">: Removes database key/value</span></div>
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<span class="GramE"><b><span style="color: black;">database</span></b></span><b><span style="color: black;"> deltree</span></b><span style="color: black;">: Removes database keytree/values</span></div>
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<span class="GramE"><b><span style="color: black;">database</span></b></span><b><span style="color: black;"> get</span></b><span style="color: black;">: Gets database value</span></div>
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<span class="GramE"><b><span style="color: black;">database</span></b></span><b><span style="color: black;"> put</span></b><span style="color: black;">: Adds/updates database value</span></div>
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<span class="GramE"><b><span style="color: black;">database</span></b></span><b><span style="color: black;"> show</span></b><span style="color: black;">: Shows database contents</span></div>
<h2 style="font-family: Arial; font-size: 14pt; font-style: italic; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
IAX Channel commands<span style="color: #436976;"></span></h2>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> debug</span></b><span style="color: black;">: Enable IAX debugging</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> no debug</span></b><span style="color: black;">: Disable IAX debugging</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> set jitter</span></b><span style="color: black;">: Sets IAX jitter buffer</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> show cache</span></b><span style="color: black;">: Display IAX cached dialplan</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> show channels</span></b><span style="color: black;">: Show active IAX channels</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> show peers</span></b><span style="color: black;">: Show defined IAX peers</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> show registry</span></b><span style="color: black;">: Show IAX registration status</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> show stats</span></b><span style="color: black;">: Display IAX statistics</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> show users</span></b><span style="color: black;">: Show defined IAX users</span></div>
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<span class="GramE"><b><span style="color: black;">iax2</span></b></span><b><span style="color: black;"> trunk debug</span></b><span style="color: black;">: Request IAX trunk debug</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> debug</span></b><span style="color: black;">: Enable IAX debugging</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> no debug</span></b><span style="color: black;">: Disable IAX debugging</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> set jitter</span></b><span style="color: black;">: Sets IAX jitter buffer</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> show cache</span></b><span style="color: black;">: Display IAX cached dialplan</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> show channels</span></b><span style="color: black;">: Show active IAX channels</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> show peers</span></b><span style="color: black;">: Show defined IAX peers</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> show registry</span></b><span style="color: black;">: Show IAX registration status</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> show stats</span></b><span style="color: black;">: Display IAX statistics</span></div>
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<span class="GramE"><b><span style="color: black;">iax</span></b></span><b><span style="color: black;"> show users</span></b><span style="color: black;">: Show defined IAX users</span></div>
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<span class="GramE"><b><span style="color: black;">init</span></b></span><b><span style="color: black;"> keys</span></b><span style="color: black;">: Initialize RSA key passcodes</span></div>
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<span class="GramE"><b><span style="color: black;">show</span></b></span><b><span style="color: black;"> keys</span></b><span style="color: black;">: Displays RSA key information</span></div>
<h2 style="font-family: Arial; font-size: 14pt; font-style: italic; font-weight: bold; margin: 12pt 0in 3pt; page-break-after: avoid;">
SIP channel commands<span style="color: #436976;"></span></h2>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> debug</span></b><span style="color: black;">: Enable SIP debugging</span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> no debug</span></b><span style="color: black;">: Disable SIP debugging</span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> reload</span></b><span style="color: black;">: Reload sip.conf  </span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> show channels</span></b><span style="color: black;">: Show active SIP channels</span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> show channel</span></b><span style="color: black;">: Show detailed SIP channel info</span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> show inuse</span></b><span style="color: black;">: List all inuse/limit</span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> show peers</span></b><span style="color: black;">: Show defined SIP peers (clients that register to your Asterisk server)</span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> show registry</span></b><span style="color: black;">: Show SIP registration status (when Asterisk registers as a client to a SIP Proxy)</span></div>
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<span class="GramE"><b><span style="color: black;">sip</span></b></span><b><span style="color: black;"> show users</span></b><span style="color: black;">: Show defined SIP users</span></div>
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Set Up Your Own IP-PBX Note: I know this is seriously out of date. I have not been able to update it, and I don't expect that to change for the forseeable future. Still, many people continue to find it useful so I wil…

Set Up Your Own IP-PBX (Asterisk)

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