One of the reasons VoIP is such a positive evolutionary step for telecommunications is that it employs a highly distributed, software-centric design philosophy. It has the extensibility and programmability of the Internet, putting telephony power back into the hands of the users, not the phone companies. It is programmability—software—that makes IP telephony such a killer application.

Yet, critical parts of voice telecommunications are entirely in the domain of hardware. This chapter focuses mainly on hardware hacks: projects with a decidedly piquant, earthy flavor; projects that deal with analog telephone adapters (ATAs), phone line gateways, and “bat phones.” You won’t need much in the way of ‘l33t coding skills, but you’ll use some basic Perl. Have some Ethernet patch cables and Velcro handy. Oh, and it might help to have a bottle of XML on ice if the mood is right.

In this chapter, you’re going to be looking at VoIP and legacy telephony hardware—everything from state-of-the-art Internet Protocol (IP) phones to vintage rotary-dial candlestick phones—and how to get them working together. You might also pick up some handy tips for your cell phone, as well.

Digital and IP phone handsets are analog inside, which means you can use a transducer microphone to record a phone call.

It’s fairly easy to record from a standard telephone using an inline recorder switch. These devices allow you to record the analog audio signal on a standard Plain Old Telephone Service (POTS) line or a handset-to-deskset line using an analog recording device like a microcassette recorder or a personal dictation recorder. An easy way to use one of these recorder switches to produce a digital recording of a call is to connect the mono audio output to your computer’s microphone line-in.

Most of these switches (such as Radio Shack’s model 43-1237) offer a 1/8-inch male audio connector, which is perfect for use with a PC sound card or a Mac line-in, which both tend to be 1/8-inch female connectors. Just plug the recorder switch into the phone line and your computer’s audio input, and you’ve got an instant call recorder in the form of your favorite audio recording program (such as Windows Sound Recorder).

Since inline recorder switches work only with analog lines, you can’t use them to record calls on digital or IP telephones. If you want to use your PC to record from these devices, you’ll need something a little more James Bondish, like a transducer pickup. This is a microphone that you stick to the outside of your telephone handset, on the back of the receiver, that is sensitive enough to record the audio inside the handset. Since even digital and IP telephones use purely analog handsets, a transducer pickup can record them all. Some pickups (such as Radio Shack’s model 44-533) include a built-in suction cup that adheres easily to the handset. Like a recorder switch, these pickups provide a 1/8-inch mini plug that you can mate with your sound card’s audio line-in jack to make digital recordings.

With minimal effort, Grandstream’s BudgeTone series of IP phones can make and receive calls on your network—even without a PBX server.

In most enterprise VoIP setups, you have a PBX that connects all of the phones on the network. The PBX acts as a centralized signaling authority and access-control server for all of the telephone users. But some IP phones don’t need a PBX at all. They can call each other directly by way of an IP address. You’re about to make a direct IP-to-IP call with a BudgeTone hardphone.

The Grandstream BudgeTone 100 phone model has a Menu key, an LCD display, and two arrow keys that you use to navigate its configuration menu options: DHCP, IP Address, Subnet Mask, Router Address, DNS Server Address, TFTP Server Address, Codec Selection Order, SIP Server Address, and Firmware Versions (called Code Rel on the phone’s screen). When you get to the option you want, you press the Menu key to select it, and then enter the numeric data required for each option using the keypad. Use this menu only to set up the IP address, subnet mask, and router (default gateway) address, because you’ll be doing the rest of the phone’s configuration using its web interface.

To get the phone enabled for the next configuration step, turn DHCP off and assign an IP address, subnet mask, and router address.

You can perform more advanced configuration using the BudgeTone’s built-in web configuration tool. When you use your web browser to access the IP address you assigned, you’ll be prompted to log in to the phone. The default password is admin.

Then, you’ll be confronted with a big page of configuration options, many of which are available only through this interface, not from the phone’s keypad metu. For this project, the only settings we’re concerned with are the codec selection ones. Configure the first (highest-priority) codec to be "PCMU” if you’re in North America or "PCMA” if you’re elsewhere in the world. That’s all we’re going to cover about codecs for now. After you apply any configuration changes, you need to power-cycle the BudgeTone.

Some IP phones offer a Telnet interface rather than a web-based one. To use these tools, you must connect to the phone with a Telnet client rather than a web browser. In any event, once the network configuration is set on the IP phone, ping its address from another host on the same network subnet to make sure it’s speaking Transmission Control Protocol/Internet Protocol (TCP/IP).

	010 001 001 103

Mounting the Grandstream on the Wall

For practical, day-to-day use, Grandstream, shown in Figure 5-1, has a few shortcomings. At the top of my “bug list” for the Grandstream 101 is its half-baked support for being hung on the wall (in addition to sitting flat on the desktop). I say half-baked because Grandstream provides screw holes for hanging the phone on the wall, but it doesn’t provide a notch to keep the handset on the phone when it’s hanging. So, the handset just slips off the phone when you attempt to set it upright.

Figure 5-1. The Grandstream BudgeTone is a great, cheap SIP phone

This won’t do. I’ve envisioned two ways to deal with this problem. First, you can go the Velcro route. Apply about a square centimeter of Velcro adhesive hook strip to the handset, at the point where a normal wall-hanging handset’s notch would be. At the corresponding position on the phone itself, put the same amount of Velcro latch strip so that when you hang up the handset, it actually stays in place.

The second way to deal with the wall-hanging problem, which is probably a longer-lasting or more durable approach, is to drill a small hole in the phone base at the point where the wall-hanging notch should be. The hole should be about one-third of an inch to three-quarters of an inch in diameter. Then, again, at the corresponding spot on the handset, screw into the plastic casing a very short, round-headed screw. The head of the screw, if small enough to fit into the hole you drilled, should keep the handset firmly latched onto the phone’s base. Not pretty, but it works.

Sure, your cell phone has a custom ringtone, but does your IP phone? With a little help from Perl, you’ll able to load any sound you like onto the Grandstream phone.

If you carry a cell phone, you’ve no doubt changed your ringtone once or twice. From a sample of a vintage mechanical ringer to a recording of a C-3PO line from Star Wars, ringtones have become central to pop-culture communication. So why can’t you customize the ringtone on a Grandstream IP phone, one of the cheapest and most popular SIP hardphones available?

Well, since you asked, you can. It just takes a little hack job.

The Grandstream’s firmware stores the ringtone in its own odd format, a uLaw sound file with a custom header at the beginning of it. It’s simple enough to make a uLaw sound file; just use SoX [Hack #24] . But to add the header, a little Perl magic is needed.

	#!/usr/bin/perl
	$filename = shift or die "need output filename
";
	undef $/; # slurp whole file at once…
	$audio = <>; # … like this
	$filesize = 512 + length $audio;
	if ($filesize & 1) {
		# length odd, add a zero byte (should never happen)
		$audio .= chr(0);
	}
	die "Audio file too large
" if $filesize > 65536;

	# this is the format for the header
	$headerfmt = "n n n C4 n C C C C a22 n x216 n n x36 a216";

	# get the current date and time
	($min, $hour, $day, $month, $year) = (localtime)[1..5];
	$year += 1900;
	$month += 1;
	
	# create the header, with zero for the checksum
	$header = pack $headerfmt,
			0, # 0000
			$filesize/2,
			0, # put checksum in later
			1,0,0,1, # version
			$year, $month, $day, $hour, $min,
			$filename,
			0,		  # 0000 or 00C8 - why?
			256,		# 0100
			$filesize/2,
			"Grandstream standard music ring";

	# sanity check
	$headerlen = length $header;
	die "header length wrong ($headerlen)
" unless $headerlen == 512;

	# add the audio
	$header .= $audio;

	# compute the checksum
	$checksum = unpack "%16n*", $header;
	#printf "checksum before = %04x
", $checksum;

	# insert it in the correct place
	substr($header,4,2) = pack "n",-$checksum;

	# ensure the new checksum is zero
	$checksum = unpack "%16n*", $header;
	#printf "checksum after = %04x
", $checksum;
	die "checksum failed
" unless $checksum == 0;

	# write the file
	open F, ">$filename" or die "can't open output file $filename: $!
";
	print F $header;
	close F;

	$ sox my_sound -r 8000 -c 1 -t ul - rate | makering.pl ring1.bin

If you own a Sipura ATA, you’ve got a veritable softPBX hiding in that slick plastic enclosure. If only you knew how to set it up!

Sipura Technology, now a division of Cisco, makes some very powerful telephony devices. With hundreds of options and many potential combinations, literally thousands of possible configurations are available. While I can’t cover them all here (for obvious reasons), I can give you a few examples to get your mind working.

Various Tweaks

After you have the Sipura connected to your network and you know its IP address, you can get to its web interface. If you’re used to dealing with web interfaces for Network Address Translation (NAT) firewalls and the like, the Sipura web interface, shown in Figure 5-2, is probably unlike anything you’ve seen. Most people that use the web interfaces of small-office/home-office (SOHO) NAT/firewall/router devices are shocked when they see the web interface on a Sipura. Here, I will attempt to point out the most common and useful, yet often overlooked, web interface parameters.

To reach the web interface, simply enter the Sipura’s IP address in your web browser. Once you see the gray status screen, click the Admin link in the top right-hand corner. When the page refreshes, click Advanced. You should see several more tabs appear. Now we are ready!

While I fully encourage you to review the user guide and browse the configuration pages, I have summarized in Table 5-2 my “Top 10 Sipura options” for your hacking pleasure.

Figure 5-2. The Info tab on the Sipura ATA’s web interface

Table 5-2. Top 10 Sipura options

Tab title	Option name	Recommended value	Explanation
System	Primary NTP Server	pool.ntp.org	Sets the SPA’s clock automatically.
System	Admin Password	Make it up!	Sets the admin password.
SIP	SIP T1	.5–2	Sets the SIP timeout value. Crank this up for high-latency network connections.
Regional	Time Zone	Your time zone	Sets the SPA’s time zone.
System	Syslog Server	IP address of syslog server on your network	Very useful for debugging.
Provisioning	Upgrade Enable	Yes	Will use the URL from “Upgrade Rule” to upgrade the SPA’s firmware automatically.
Provisioning	Provision Enable	Yes	This requires the Sipura profile compiler. If you have more than 10 SPAs, you should be able to obtain this tool to aid in configuration. Contact Sipura for more information.
X Line	RTP TOS/DiffServ Value	Varies	This controls the IP TOS value for Real-time Transport Protocol (RTP) or audio packets from the SPA. When used in conjunction with intelligent switches and routers, this can ensure excellent voice quality on your network.
X User	VMWI Ring Splash Len	0	This is a common request. It will disable the “splash ring” for voicemail notifications. Otherwise, your analog phone will chirp every so often if you have a voicemail. Very annoying.
X Line	Preferred Codec	Varies	Sets the preferred voice codec to use. Various codecs are available with quality/bandwidth trade-offs.

	Line 1 Dial Plan: (7xxx)
	Interdigit short time: 3 seconds

	Line 1 Dial Plan:
	([2-9]xx[2-9]xxx|[2-9]xx[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx|011[2-9].|7xx|7xxx)
	Interdigit short time: 3 seconds
	Interdigit long time: 10 seconds

	(<111:[email protected]:5061)

	(<111:[email protected]:5061>|<112:[email protected]:5061>|<113:
	[email protected]. 0.13:5061>)

	(<:1847>[2-9]xx[2-9]xxx|1[2-9]xx[2-9]xxxxxx|011[2-9].|7xx|7xxx|xx.)

	([49]11<:@gw0>|*xx<:@gw0>|[2-9]xxxxxx<:@gw0>|[2-9]xx[2-9]xxxxxx<:@gw0>
	|1800xxxxxxx<:@gw0>|18[6-8][6-8]xxxxxxx<:@gw0>|7xx|7xxx|1[2-9]xx[2-9]xxxxxx|011[2-9].)

	([49]11<:@gw0>|[2-9]xxxxxx|[2-9]xx[2-9]xxxxxx|1800xxxxxxx|18[6-8][6-
	8]xxxxxxx|7xx|7xxx|1[2-9]xx[2-9]xxxxxx|<9:>xx.<:@gw0>)

Do you think Bruce Wayne uses VoIP to receive emergency calls from the Mayor of Gotham? Of course he does. He’s that cool (his car is OK, too).

If you’ve worked your way through “Tweak Your Sipura ATA” [Hack #62] , you know Sipura Technology makes some very powerful and flexible ATAs. So powerful, in fact, that you can use them to set up a point-to-point “hot line” with no SIP proxies or registrars.

A "bat phone” (or automatic ring-through in the telco world) is best known from the popular Batman television series. Batman would have such a burning desire to speak with the commissioner that he didn’t even have time to dial. The simple act of picking up the phone automatically connected him to the designated remote station.

Here is what you will need to get this going with your two Sipuras:

	(S0<:[email protected]:5060>)

	(S0<:[email protected]:5060>)

With a little tweaking, a very common fax/modem card can become a clone of the single-line X100P interface card.

The Digium X100P foreign exchange office (FXO) card, used to connect a single phone company line to an Asterisk server, is actually an Intel V.92 Data/Fax/Voice modem card. One visual comparison between an Intel V.92 Winmodem PCI card and an official X100P, and it’s obvious that the two cards are identical. So, using the less-expensive modem card in place of an X100P card is not only possible, it’s downright easy.

The Intel 537EP chipset is a V.92 PCI modem chip family. Many modems are built on the Intel 537EP chipset, but this hack is known to work only with the Intel V.92 Winmodem card.

The critical thing about using an Intel V.92 modem card that has not been purchased from Digium as an X100P, but otherwise looks the same, is that the vendor ID encoded into the card will read differently, breaking the original Zaptel driver and rendering the card useless. Fortunately, there are two ways around this. The most obvious solution is to hack the code of the driver. Before you compile Asterisk and Zaptel from the Digium CVS archive [Hack #41] , you’ll need to edit the zaptel/wcfxo.c file.

Here’s the existing code snippet you’ll need to change:

	static struct pci_device_id wcfxo_pci_tbl[] __devinitdata = {
	{ 0xe159, 0x0001, 0x8085, PCI_ANY_ID, 0, 0, (unsigned long) &wcx101p },
	{ 0x1057, 0x5608, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long) &wcx100$ };

Change this section in zaptel/wsfxo.c to this:

	static struct pci_device_id wcfxo_pci_tbl[] __devinitdata = {
	{ 0xe159, 0x0001, 0x8085, PCI_ANY_ID, 0, 0, (unsigned long) &wcx101p },
	{ 0xe159, 0x0001, 0x8086, PCI_ANY_ID, 0, 0, (unsigned long) &wcx101p },
	{ 0x1057, 0x5608, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long) &wcx100$ };

The line added in the middle will allow the Zaptel wcfxo driver to work with standard Intel V.92 Winmodem boards (while still keeping the driver compatible with official Digium X100P cards). Recompile the Zaptel drivers [Hack #41] , and your Intel V.92 cards can be used as FXO interfaces. Pretty neat, eh?

The other, more difficult way to enable this feat is by modifying the boards themselves. This means re-creating the same modification that Digium does when it modifies Intel cards to create so-called genuine X100P cards. Remove the resistors marked R13 and R19 by unsoldering them. But be careful, and don’t expect to return your Intel V.92 card, as its warranty will now be invalid.

Cisco’s 7900 series IP phones have some powerful programmable firmware that you can harness for your own unique purpose, answering the age-old question, “Doesn’t this LCD display seem a bit large for just caller ID?”

That expensive Cisco phone on your desk has some great hidden capabilities. Additional tools and toys that lurk beneath the 79xx’s gray exterior could increase your productivity and foster some innocent fun. I’m talking about things like automated weather reports on the phone’s display, simple menu-driven applications (like a time card, say), and just about anything else you can program using an XML web site.

In fact, you’ll probably build so many cool tools and toys that you’ll need a way to sort through them, like a directory or a menu. Using the Cisco VoIP phone’s XML application capability, you can set up such a directory.

For this hack, you’re going to use XML to create your own custom menus that access hidden features of the Cisco phone. To make your menu appear on the phone, you’ll need to configure the phone to look for your custom menu file.

Cisco phones, like most IP phones, have a Flash storage device onboard that is checked and optionally updated at every startup. During startup, the phones contact a TFTP server and attempt to download settings stored in files on the server [Hack #80] . By setting the services_url property on the phone’s console to the URL of your menu, you can configure the phone to load your custom menu upon its next reboot. To set this property, press the Settings button on the Cisco phone. Then, select Unlock Config and use the keypad to enter the password (the default is cisco). Next, select the XML URL option, followed by the Services URL option. Enter the URL, like the following example, using the phone’s keypad (you enter slashes and colons by cycling through the # and * keys):

	http://lathama.com/services/cisco/

Now, when the phone boots, it will attempt to load a menu for the Services button by accessing an XML file at the URL. You can also set this URL in a configuration file on the TFTP server. (If you do neither, the Services button will be rather useless, as it doesn’t default to any built-in settings.) Edit the SIPDefaults.cnf file on the TFTP server to set the Services URL. Use an entry like this:

	           services_url: "http://lathama.com/services/cisco/"

Leave the ending slash in there. It will tell the web server to grab the default or index file. Testing this with a web browser will be difficult, as the XML has no headers to tell the browser what is going on. Create a file called index.html (default.htm for Windows web servers) on a nearby web server that’s accessible at the URL you supplied to the phone, and fill it with XML content like this:

	<CiscoIPPhoneMenu>
		<Title>My New Menu</Title>
		<Prompt>Prompt This</Prompt>

	<MenuItem>
		<Name>Menu Item</Name>
		<URL>http://domain/cisco/services/menuitem.xml</URL>
	</MenuItem>

	<SoftKeyItem>
		<Name>Soft Key</Name>
		<URL>http://domain/cisco/services/softkey.xml</URL>
		<Postition>1</Posititon>
	</SoftKeyItem>

	</CiscoIPPhoneMenu>

This sample shows you a simple yet slick method of loading a menu. When you press the Settings button on the phone, the menu shows up as simple text bars and allows you to arrow-down or up to the option you choose, using the phone’s navigation controls and softkeys. After you make a selection, the listed URL is queried. This convention is the formula behind what Cisco calls services. You can think of services as the little applications (stock tickers, games, etc.) on modern cellular phones. The Cisco phones split the phone directory off as its own service (available through the Directory button rather than the Services button), but that shouldn’t stop you from making your own directory with this hack.

The menus are simple to create, but you’ll need to become familiar with the strange tags. If you’re not familiar with XML, the <CiscoIPPhoneMenu> tag, which is used to enclose Cisco phone menu structures, might seem a little strange to work with. I would have just used <Menu>, but that might not work in newer firmware. Menus contain one or more menu items denoted with the tag <MenuItem>. A softkey type of menu that uses the <SoftKeyItem> tag also is available. Softkey menus use the buttons on the side of some of the phones (the 7960, for example, has six on the right and four at the bottom of the display). Some of the older Cisco phones do not have the extra buttons, so for compatibility with all of Cisco’s IP phones, stickto simple menus.

Inside <MenuItem> are easy tags, such as <Name> and <URL>. These are casesensitive on some current firmware versions. <Name> is what shows up as the item’s name, and <URL> is what it will request if the user selects the menu option. Notice that the http:// part is in the URL, but FTP URLs will work, too.

Now, on to your first menu. Start the following example and adjust it to suit. Let’s say you use the extension of the phone for the title, to allow you and your friends or users to know the extension at any time. Break up the menu into chunks that you can handle:

	<CiscoIPPhoneMenu>
		<Title>EXT 1234</Title>

	<MenuItem>
		<Name>PIM</Name>
		<URL>http://domain/cisco/services/pim.xml</URL>
	</MenuItem>

	<MenuItem>
		<Name>Work</Name>
		<URL>http://domain/cisco/services/work.xml</URL>
	</MenuItem>

	<MenuItem>
		<Name>Play</Name>
		<URL>http://domain/cisco/services/play.xml</URL>
	</MenuItem>

	</CiscoIPPhoneMenu>

So you now have three sections to add things into. Work on PIM first, and add a list of favorite restaurants (hey, you’ll want to reward yourself with a pizza when this is through!). Go to the pim.xml file on the web server. Add some local and not-so-local places, and then allow the menu to dial the number for you. To keep my favorite places safe, I’ll suggest fictional restaurants and phone numbers. (Actually, if you’re in the Detroit area, nothing beats National Coney Island!)

	<CiscoIPPhoneMenu>
		<Title>EXT 1234 - PIM</Title>

	<MenuItem>
		<Name>Restaurants</Name>
		<URL>http://domain/cisco/services/pim.xml</URL>

	</MenuItem>

	</CiscoIPPhoneMenu>
		
	<CiscoIPPhoneDirectory>
		<Title>EXT 1234 – PIM - Restaurants</Title>

	<DirectoryEntry>
		<Name>Larrys Latkes</Name>
		<Telephone>18005551212</Telephone>
	</DirectoryEntry >

	<DirectoryEntry >
		<Name>Timbos Nacho World</Name>
		<Telephone>18665551212</Telephone>
	</DirectoryEntry >

	<DirectoryEntry >
		<Name>Drunken Cow Steak House</Name>
		<Telephone>18775551212</Telephone>
	</DirectoryEntry >

	</CiscoIPPhoneDirectory>

Look at all that’s changed. You are now using the <CiscoIPPhoneDirectory> tag to show the dial and other buttons at the bottom of the screen. You also are now using <DirectoryEntry> rather than <MenuItem>, and <Telephone> rather than <URL>. To complete the hack, edit the work.xml and play.xml files.

—Andrew Latham

To avoid lock-in with Cisco-only phones and switches, learn how to power Cisco phones from non-Cisco switches.

IP phones can be powered through their Ethernet connections. The standard for this inline power is called 802.3af, and many equipment manufacturers support it—except for Cisco, which uses its own proprietary inline power method. Because of this, you can match Cisco IP phones only with Cisco-powered switches (unless you use Cisco’s only phone model to support 802.3af—the 7970). This is an unfortunate form of vendor lock-in, but all is not lost. You can do a couple of things to get Cisco IP phones to draw power from non-Cisco switches.

If your budget permits, the obvious (though proprietary) solution to this problem is to use Cisco PoE switches to power the phones. Some other switch makers, like Foundry Networks, also support Cisco’s proprietary PoE standard. If you can’t afford to forklift your switches, you might instead want to power your Cisco phones by way of a power injector, which is a patch panel that adds inline power to a CAT5/CAT6 cable connection. Consider Cisco PoE-compatible injectors like those made by PowerDsine (http://www.powerdsine.com/).

But, if you can’t do that either, do the next best thing: hack.

By changing some wires on a standard UTP Ethernet patch cable, you can make a compatibility cable that lets you plug Cisco IP phones into any 802. 3af source, as shown in Figure 5-3. Essentially, you are flipping wires 4 and 7, and 5 and 8. Be advised, this technique could void the warranty of your phone and your switch.

Figure 5-3. The wiring diagram for a hacked PoE cable

Make sure your switch lets you program, port by port, which ports get power and which ones don’t, because in a native Cisco PoE solution, Cisco IP phone power requirements are “auto-detected,” so power can turn itself on and off as necessary on each port. There’s no such provision when using a hacked cable to supply 802.3af power to a Cisco PoE-using phone. If this is a problem, and 802.3af won’t work with the hacked cable, try using a device that does the two-pair flip but also works with auto-detection, such as 3Com’s 48-volt IntelliJack switch converter, part number 3CNJVOIPCPOD.

Change the logo on your Cisco IP phone, and reflect your inner geek’s refined sense of monochrome style.

Have you ever wished you could change the boot-up logo on your cell phone? Have you ever wanted to use custom graphics on your appliances’ LCD screens? Most Linux geeks love to plaster Tux the Penguin, the official mascot of Linux, all over the place—and what better place than a hackable display? If you’re like me and you have a thing for the penguin, allow our underdressed friend to show himself on your Cisco VoIP phones.

First, the facts: when most IP phones boot, they look for configuration files on a nearby TFTP server and download them to configure the phone further [Hack #80] . The configuration files allow the specification of a logo, along with other tweakable goodies. By editing or adding the logo_url setting in a phone’s configuration file, you can dictate which logo the phone should use. The storage or location for this logo varies depending on the version of the firmware that’s loaded on your phone, but you should be able to specify a standard HTTP URL to point the phone toward its logo. (This kind of hack is also possible on other phones, so look it up.) Here’s the specific setting for a Cisco configuration file:

	logo_url: "http://domain/cisco/logo.bmp"

As you can see, the URL points to a bitmap picture (logo.bmp). So far, this looks to be a simple hack, and with a few notes, it will stay that way. The image that the phone downloads is on a server, so it needs to find the server somehow. In other words, make sure the phone’s DNS server setting is right so that it can resolve the hostname you provide in place of the domain place-holder in the URL.

The image size and color are also important. If you use an image with the wrong size or aspect, it will come out looking a bit funky on your Cisco’s LCD. The default size for the display on the Cisco 7960 is 133 x 65 pixels, so that would be a good place to start. The image should be monochrome, at least for the 7960. Color is supported on newer models, like the 7970G. (You can always specify a different URL in your color phone’s config files that points to a color version of the same image if you need to support color and monochrome displays.) The older and current phones will alter an image to fit the screen and color if it does not match. This auto-correction might not be perfect, so you might want to run your image through Photoshop or the GNU Image Manipulation Program (GIMP) to meet the size and color requirements.

When I did this hack, I used a PNG-format picture of Tux the Penguin. Tons of great images like this are available at http://images.google.com/. I opened my image in the GIMP to have a look. Then I resized the happy fellow so that his height matched the height of the LCD, 65 pixels. Finally, I converted to grayscale and saved. Figure 5-4 shows the finished product. Cute, isn’t he?

Figure 5-4. The Tux logo, as he appears on a Cisco 7960’s display

I then simply uploaded the file to my web server, at the URL specified in my 7960’s TFTP configuration file. The next time I booted up my 7960, there was Tux, happy as usual.

—Andrew Latham

Uniden’s IP phones, like the UIP200, are excellent business-grade telephones that can be mass configured by TFTP—that is, if you know how.

As you might have already gathered, you can configure IP phones in three ways: directly, using the phone’s LCD and buttons; through a web or Telnet interface; or via a configuration file the phone downloads from a TFTP server during boot-up. Those three methods are presented in order of the level of detail to which you can administer the phone, with TFTP configuration allowing the most precise control. Once you’ve got a TFTP server running [Hack #80] , you need only drop the right text files onto your TFTP server to mass-configure your IP phones.

The Uniden IP phones don’t offer the trademark high-end look and feel of the Cisco 7900 series of phones, but they do provide a good value nonetheless. At half the price of a typical Cisco SIP phone, the Uniden UIP200 SIP phone supports all of the standards fundamental to a VoIP LAN: inline power, SIP, and several of the key audio codecs. Getting a UIP200 onto the network and doing useful things with your softPBX is a snap, with the help of mass configuration via TFTP.

	              ProxyServer			10.1.1.10
	ProxyServerPort		0
	OutboundProxy1		10.1.1.10
	OutboundProxy1Port	0

	              Registrar1			10.1.1.10
	Registrar1Port		10.1.1.10

	DnsServer_1			10.1.1.10

	RegisterExpireSec	3600
	RegisterRetrySec	90

	              SipPort				5060

	G711MuTxPacketLength		20
	G711MuJitterBufferLength	10
	G711ATxPacketLength			20
	G711AJitterBufferLength		10
	G729TxPacketLength			20
	G729JitterBufferLength		10

	              DiffServMode			OFF
	DefaultDiffServParam	192
	RTPDiffServParam		160

	VlanMode				Disable
	VlanID				1
	PcVlanID				2

	TftpAddress				10.1.1.10

	TimeZone				-6
	EnableDST				YES
	EnableSNTP				YES
	SntpServerIP			10.1.1.10
	SntpRetrySec			1800

	              PreferredCodec			g711u,g711a,g729
	Language				English

	              StunServerAddr			0.0.0.0
	StunServerPort			3478

	DirectIpDialing		No

	AdminPassword		1234/5678

Use unique configurations on each IP phone, and while you’re at it, do some firmware revision control, too.

There are two files for each Uniden IP phone on the TFTP server: one that’s shared by all of the phones on the network (unidencom.txt, described in “Configure Multiple IP Phones at One Time” [Hack #68] ) and one that’s exclusive to each phone on the network. These exclusive, phone-specific config files, whose filenames contain the names of their corresponding phone’s MAC hardware address, control the firmware and hotkey setup of that particular IP phone. I’ll step you through a sample Uniden phone-specific config file, as it might appear on your TFTP server:

	AutoFirmwareUpdate		YES
FirmwareFileName			uip200_455enc.pac
FirmwareVersion				BS4.55

Enabling AutoFirmwareUpdate with a YES will cause the phone to attempt a firmware patch automatically when it boots. It will try to grab (and install) the firmware package specified by FirmwareFileName from the TFTP server. The desired firmware version is specified by FirmwareVersion, and the phone will grab the firmware file you specify only if the version is different from the version currently running on the phone.

	MyLcdDisplay			Maddie's Phone
	MyDialNumber			1138
	DisplayName				Madelyn
	UserNameForProxy		1138
	PasswordForProxy		uniden
	UserNameForRegistrar	1138
	PasswordForRegistrar	uniden

MyLcdDisplay determines what greeting to display on the phone when it is waiting to call or be called, and MyDialNumber determines what number to display. DisplayName attempts to set the caller ID name to be used on outgoing calls, if the softPBX supports this. UserNameForProxy, PasswordForProxy, UserNameForRegistrar, and PasswordForRegistrar establish the login credentials to be used when the phone logs into the SIP servers that handle its calls (proxies and registrars are often hosted on the same server, so the credentials are often identical).

	ProgrammableKey1		OneTouchDial
	ProgrammableKey2		TwoTouchDial
	ProgrammableKey3		CallForward
	ProgrammableKey4		DoNotDisturb
	ProgrammableKey5		VMA
	ProgrammableKey6		Mute

The ProgrammableKey1 through ProgrammableKey8 settings allow you to assign functions to the UIP200’s hotkey. Here’s what the possible values do:

	OneTouchKey1		18005551212
	OneTouchKey2		411
	TwoTouchDigit0		3000
	TwoTouchDigit1		3001
	TwoTouchDigit2		3002

You use the OneTouchKey1 through OneTouchKey4 settings to supply the phone numbers that are used with up to four ProgrammableKey settings, so you can set up to four of the UIP200’s eight hotkeys to be one-touch dialing keys. The TwoTouchDigit0 through TwoTouchDigit9 keys, on the other hand, are used to set up two-touch dialing (first the hotkey, and then a number key on the dial pad). Values supplied here become the phone numbers that are called whenever a two-touch dial occurs.

	VmaDirectCallNo		8080
	VmwiLampIndicator	Enable

VmaDirectCallNo tells the phone what number to call when the VMA hotkey is pressed. VmwiLampIndicator, when Enabled, permits the phone to light its message-waiting indicator light. It’s probably not a good idea to disable this one.

Once you’ve got this file set up the way you like, save it in the format uniden <MAC>.txt, where <MAC> is the Ethernet hardware address of the phone it applies to. Then, reboot the phone!

Using an X10 phone controller, you can turn your lights on and off from the comfort of your IP phone.

X10 home-control interface equipment has been a favorite pastime of geeks for decades. Since the early days of 8-bit hobby computers, you’ve been able to automate your home using your keyboard (and later, your mouse). X10 interface controllers connect to lights and other appliances in your house, and your computer can send serial commands to the controllers to turn them on and off and adjust voltage like a dimmer. Some X10 interfaces even offer telephone-based user interfaces, letting you control them by calling them with your phone.

There are a few ways to integrate X10 controls with an Asterisk phone system. The integration method depends on the type of X10 controller purchased. For this hack, I chose the X10 TR16A phone controller that operates by DTMF digits. Ordinarily, you would hook a phone line to it and then call that phone line with a standard phone to operate the TR16A. But, with Asterisk, you can connect directly to the TR16A as if you yourself are the phone company. Then, controlling the TR16A is as simple as a dial-plan hack in Asterisk.

The Asterisk system I used to connect to the TR16A contains a single Wild-card TDM400P using two foreign exchange station (FXS) modules and one FXO module. The system has one analog phone and one SIP-enabled Polycom IP500 phone. The SIP phone is the one I used as my “remote control”—the phone from which I sent my commands to the X10 controller.

I performed the following steps to integrate the TR16A with the Asterisk system.

For the TR16A controller:

For the Asterisk system:

Here’s the bit from /etc/asterisk/extensions.conf that you would use, assuming the FXS port is Zap/1:

	exten => 100,1,Dial(Zap/1)

Here’s the bit from /etc/asterisk/sip.conf:

	[200]
	username=200
	secret=200
	type=friend
	dtmfmode=inband
	disallow=all
	allow=ulaw

Verifying if the setup is correct requires a few test calls. Use the Polycom IP phone to call the controller. Go off-hook on the IP phone and dial the extension number assigned to the controller (100 in this example). The line will ring for about 15 seconds before the controller will answer. When the controller answers, you will hear three beeps. Enter your PIN code, and then a second set of three beeps will confirm that you entered the proper PIN code. After the second set of three beeps, enter the module number you want to control, followed by * or # to turn the module on or off, respectively.

I mentioned that there are a few ways to integrate X10 controls with Asterisk. Using the TR16A, I demonstrated how you could connect to an Asterisk phone system using an FXS port and an IP phone to operate the X10 controller. X10 has another controller known as the CM15A. The CM15A connects to your local PC using a USB cable, and it boasts a free SDK that you can use to write custom scripts. Using the Asterisk system’s IVR function, you can program Asterisk to run a script. In turn, that script can control the X10 modules, allowing for a pure software-based solution—that is, no FXS interfacing necessary.

	exten => 100,1,Dial(Zap/1,30,D(1212))

Or “How Grandma Mabel Learned to Love Voice over IP.”

What do a 19th-century Missouri undertaker and SIP have in common? The common thread is their ability to connect two parties without a third party intervening. That is, they both serve an intermediary role between caller and callee, signaling the call and providing a pathway for its sound signals.

In 1891, a Missouri undertaker, Almon B. Strowger, was granted a patent for an electromechanical device called a stepper switch. The stepper switch allowed the calling party to control whom he would connect to without the need for an operator. Today, that control mechanism might be analogized to a phone number. But the important point is that Strowger’s invention made it possible to connect phone calls without a telephone operator’s intervention.

SIP promotes the ability to call a party without the need for a SIP gateway or gatekeeper, as long as you know the recipient’s sip@ address, called a SIP Uniform Resource Indicator (URI). To pay homage to the pioneering Mr. Strowger, we have included this hack to connect a rotary phone, such as a 1920 Western Electric candlestick phone (Figure 5-5), to a VoIP network.

First, let me give you a quick overview of how a rotary phone works. The rotary phone provides signaling to the central office by establishing a flow of current and then interrupting the flow momentarily to signal a pulse. The number of pulses produced in a given time period represents the number being dialed. Five pulses represent the number 5.

Adding a rotary phone to a VoIP network is a relatively painless process, as long as you can build an intermediary gateway that understands pulse signaling and SIP, and can signal both legs of the call. What can be painful is finding a VoIP device that supports rotary or “pulse” dialing. That’s why you might be better off building this gateway yourself.

Do Pulse with an IAXy

The Digium IAXy FXS gateway supports pulse dialing, providing a simple and complete solution. In fact, the IAXy is a complete gateway in and of itself: it has an analog phone port on one side and an Ethernet RJ45 port on the other, complete with a little Inter-Asterisk Exchange (IAX) user agent built in.

Figure 5-5. A 1920s vintage Western Electric candlestick phone

The Digium IAXy supports pulse dialing out of the box. Configure /etc/asterisk/iax.conf with some entries that will allow the IAXy to be used as a peer:

	[rotary]
	type=peer
	username=asterisk
	secret=supersecret

In this case, I used rotary as the IAX peer name. All you need to do, once this config is done, is register the IAXy with your Asterisk server as the IAX user asterisk. The rotary phone connects to the IAXy. Now you can use the rotary phone as you would any valid endpoint in your dial plan by routing calls to IAX/rotary using a Dial command.

	pulsedial=yes