Turbocharging Your Asterisk@Home PBX

This is the fourth in our series of articles on the Asterisk® PBX. You'll be much the wiser and less frustrated reading this if you begin with Part I. Then read Part II. And then read Part III. Then return here.

30 minutes and 2 cents. That's how long it took to configure a backup VoIP provider and the total cost of testing seven outbound calls in the U.S. while configuring last week's CallMe application to use the backup provider. Our choice was Voxee.com, and nobody could make it much easier or charge much less. Outgoing calls within the U.S. or to Canada, London, Paris, and most of Germany are about a penny a minute, and U.S. calls are billed in 6 second increments. China costs 2¢ and most of Mexico is under a nickel. Any place with an island in the name ... well, it's almost cheaper to go there. But that's true with all providers. It costs $5 through PayPal to seed your Voxee account with call money. After that, it's pay as you go. There are no setup fees. In fact, if you opted for the BroadVoice BYOD-Lite plan which gives you free incoming calls and 100 minutes of outgoing calls a month, then Voxee is the perfect complement. You'll pay a penny a minute for most calls with increment rounding in the U.S. while BroadVoice charges 4¢ a minute in the U.S. with full minute rounding. The other good news is that Voxee supports Asterisk's native protocol IAX2 which makes configuring telephones at remote sites a breeze. Unlike SIP, there are no NAT headaches with IAX2. All you need is a phone that talks IAX. For more information, here's a great article. There's even an adapter to plug in POTS phones using IAX. In short, if you want to skip some SIP headaches, IAX is worth a careful look at both the telephone instrument and the service provider levels.

Adding a Voxee Trunk. To add a Voxee trunk using Asterisk@Home, run AMP, choose Setup->Trunks->Add IAX2 Trunk. Maximum channels only matters if you want to restrict how many simultaneous outgoing calls through Voxee can be made. Otherwise, skip down to the middle of the form and under Outgoing Settings, name your trunk voxee. For the Peer Details, insert the following using your username and password assigned when you registered for an account:

type=friend
host=66.246.246.52
username=some number assigned by Voxee goes here
secret=the password you chose at Voxee goes here


Now drop down to the Registration field and plug in the following: YourAcctNoHere:YourVoxeePasswordHere@66.246.246.52. Save your settings and click the red bar.

Adding a Voxee Outbound Dialing Route. The final step is to add an outbound dialing route for your Voxee calls. The easiest way to set this up is to use a dialing prefix for Voxee calls, e.g. 9. So click Outbound Routing within AMP Setup. In the Add Route screen, name your new route OutVoxee. Whether to have a route password is, of course, up to you. For the dial strings, we recommend the following which lets you dial U.S. and foreign Voxee calls by dialing 9, 1, area code and number or 9, 011, country code, and number. The 9's will be stripped off before the dial string is sent to Asterisk to place the call. That's what the "9|" syntax means.

9|011.
9|1NXXNXXXXXX

Now click on the Trunk Sequence pull-down and choose voxee for your outbound trunk. Click the Submit Changes button and then the red bar to update Asterisk. The only gotcha here is to be sure you dial a 9 and a 1 plus area code and number for U.S. calls. BroadVoice doesn't require a 1, but Voxee does. If you want to use the callme application we built last week, the correct syntax to have Asterisk place the call through Voxee is http://asterisk.dyndns.org/callme.php?number=iax2/voxee/16781234567 where asterisk.dyndns.org is the fully qualified domain name for your Asterisk server and 6781234567 is the phone number to be sent dialtone. For early readers of last week's column, please note that a code change was made on Saturday to avoid a potential security problem. It's explained in the comments section of last week's article if you want more details. Otherwise, just download the callme.php file again and replace your old version.

Adding a Voicemail Address. We all are accustomed to having email addresses. So what's next: Voicemail Addresses, of course. Yes, with most SIP phones, you now can make calls to addresses that look just like your existing email address: homer@thesimpsons.com. If you've followed our tutorials thus far and have set up a fully-qualified domain name with dyndns.org or if you have your own domain pointing to your Asterisk server, then it's a two-minute operation to add voicemail addressing. Using AMP, click on the Maintenance tab and open extensions_custom.conf. Now add the following lines to the [from-internal-custom] context of the file substituting your real name for homer and adding an actual extension number on your Asterisk PBX for 1000:

exten => homer,1,dial(SIP/1000,20,m)
exten => home,2,VoiceMail(u1000@default)

Once you save your changes and restart Asterisk, anyone can call you from any IP telephone or softphone by "dialing" sip:homer@asterisk.dyndns.org where homer is the name you plugged in to extensions_custom.conf and asterisk.dyndns.org is the fully-qualified domain name pointing to your Asterisk server. You can add as many additional accounts as desired. It's equally simple to match address names to the locations of IP phone extensions in your home or office: kitchen, office, playroom, pool, or whatever else you might need. And your friends can sign up for a SIPphone account or a Free World Dialup (FWD) account to make the calls ... which are free! Many phones don't even require the sip: prefix just as most web browsers no longer force you to type http://. For those that want to make your new voicemail address match your email address, here's a link that will tell you what's necessary to make it happen with your ISP.

Sipura SPA-3000. There is no finer piece of telephony equipment you can purchase than the SPA-3000. As we mentioned last week (but it's worth repeating), the SPA-3000 does three things and does them all well. First, if you want to connect your existing home or office Ma Bell phone line to your Asterisk server so that incoming calls to your regular phone line can be managed with Asterisk just like your VoIP line, then the SPA-3000 is the way to go. It provides the best voice quality period. Second, if you want the flexibility and redundancy of having a Ma Bell line to place outgoing calls (especially for 911 service), then the SPA-3000 is just the ticket. You even get failover protection when your Internet service croaks. And finally, if you want to connect a 5.8 GHz wireless phone set to your Asterisk PBX for use throughout your home or office, the SPA-3000 is a must-have. With some careful shopping, you can purchase an "unlocked" SPA-3000 for just under $100. Unlocked means you can access the administrator setup features of the unit. You need this capability to use the device with an Asterisk PBX.

Buying the SPA-3000 it turns out is the easy part. Getting it configured properly to work with Asterisk can be a nightmare, and we're not quite sure why. Actually, we are pretty sure why. Sipura is one of those fine companies that thinks only dealers can read so you'll have a hell of a time finding even a user's guide for the product. For those of you that can read, here it is. Another part of the problem is that Asterisk is an incredibly complex toolbox which can be set up in many, many different ways. So providing HOW-TO instructions to configure an SPA-3000 is a daunting task. We're assuming you are using Asterisk@Home and that you've configured Asterisk@Home according to our tutorials. If not, you may be in for a rough ride with the SPA-3000, but we'll try to provide some explanations as we go so that you can get back on track if your configuration differs from ours. We're also going to stick with Plain English rather than telephony jargon so, if you're a telephone geek or a purist, don't go postal. Just bear with us.

Overview. There are four parts to the SPA-3000 configuration drill today. First, we'll set up the SPA-3000 hardware device and get everything plugged in correctly. Next we'll set up some Asterisk extensions to support the SPA-3000. Then we'll use a web browser to configure the SPA-3000 device to work with your Asterisk@Home server. And finally we'll add some bells and whistles to Asterisk to show off a little bit. When we're finished, your Asterisk server should be able to answer calls from both BroadVoice and your home/office Ma Bell phone line. And, you should be able to place calls through BroadVoice or your Ma Bell phone line using the phone or wireless phone set connected to the SPA-3000. As we did with the Asterisk server, we're assuming you have placed the SPA-3000 behind a rock-solid firewall. Otherwise, your phone bill may include thousands of dollars of calls to the Queen Mary that you didn't make.

SPA-3000 Hardware Interfaces. So let's begin with the hardware basics. The SPA-3000 is designed to plug in to four different things:

Once you get all the wires connected, plug in the SPA-3000. Or, if you plugged it in before connecting it to your LAN, unplug it, count to 10, and plug it back in. Once all the lights stop blinking, pick up the telephone you connected to the Phone jack. You should hear a dial tone. If not, go back to square one. Otherwise, press **** which will access the Sipura configuration menu. Once Egor stops talking, press 110# to retrieve the IP address assigned to your unit. Write it down and hang up the phone. Rather than hard-code an IP address into the SPA-3000, our preference is to tell your router to reserve the IP address already assigned so that it is reassigned to the SPA-3000 whenever you turn it off and back on. We did the same thing with your Asterisk box in Part II so reread the IP Configuration of Asterisk section of that tutorial if you need a refresher.

Configuring Asterisk to Support the SPA-3000. Before configuring the SPA-3000, let's turn our attention to Asterisk for a bit. You'll need three extensions to support the SPA-3000: one for incoming calls, one for outgoing calls, and one to handle the telephone instrument(s) you plugged into the Phone jack above. From your web browser, access AMP->Setup by going to the IP address of your Asterisk box. When prompted, type in maint for your username and whatever password you assigned to AMP previously. The trick to adding these three extensions is that you have to do everything twice because you can't enter all of the data for each extension in the original form. Aside from that, it's pretty straight-forward. You can obviously use any extension numbers you like, but using ours may make it simpler when we start configuring the SPA-3000. So here goes.

SPA-3000 Incoming Extension. Click Extensions and fill out the Add an Extension form as follows:

Click the Add Extension button to save your work and then the red bar to restart Asterisk. In the right column, click on PSTN Incoming - Extension 99 that you just added. Make sure your form looks like the following. Then save your changes and click the red bar to restart Asterisk.

Now click Submit Changes and then the red bar to update Asterisk. One down. Two to go.

SPA-3000 Outgoing Extension. Click Extensions and fill out the Add an Extension form as follows:

Click the Add Extension button to save your work and then the red bar to restart Asterisk. In the right column, click on PSTN - Extension 199 that you just added. Make sure your form looks like the following. Then save your changes and click the red bar to restart Asterisk.

Now click Submit Changes and then the red bar to update Asterisk. Two down. One to go.

SPA-3000 Phone Extension. Click Extensions and fill out the Add an Extension form as follows:

Click the Add Extension button to save your work and then the red bar to restart Asterisk. In the right column, click on Wireless - Extension 204 that you just added. Make sure your form looks like the following. Then save your changes and click the red bar to restart Asterisk.

Click Submit Changes and then the red bar to update Asterisk. Done. Now we're finally ready to configure the SPA-3000.

Configuring Your SPA-3000 for Asterisk. Before we get down to the nitty gritty, let's chat about what you'll have in place when we get finished. Our design plan is to support incoming calls from both your BroadVoice VoIP number and your POTS (aka Ma Bell or PSTN) number that you've had forever. For outgoing calls in the U.S., dialing a 7-digit number will place the call through your PSTN line. Dialing a 10-digit number will also place the call through your PSTN with BroadVoice as a backup. Dialing 1 and then area code and phone number will place the call through BroadVoice. Dialing 9 and then the area code and phone number will place the call through Voxee. Dialing 911 will place an emergency call through your PSTN (i.e. local) phone number. Dialing 011 calls for other countries will go out through Voxee. If you need another configuration, post your question using language similar to what we've outlined, and we'll respond to some of them in a future column. Don't ask questions about other VoIP providers. That's what the VoIP forums are for, and we've previously pointed you to several good ones. So here we go.

First, a reminder: MAKE SURE YOUR SPA-3000 IS BEHIND A FIREWALL! Now let's reset the SPA-3000 to its factory defaults just to be sure we're all reading from the same sheet of music. With a phone connected to the Phone jack, lift the receiver and dial **** to access the Sipura Configuration Menu. Now press 73738#. When prompted, press 1 to confirm your request. The SPA-3000 will reboot. Then, using a web browser, access the IP address of your SPA-3000 that you wrote down above. When the main screen displays, click Admin Login in the upper right corner. Then click Advanced. If you're prompted for a password, you have a locked unit, and all bets are off. Contact your vendor and either get the passwords or send the unit back. You won't have this problem with Voxilla. After clicking on the Advanced option, you'll have access to all the settings in the SPA-3000.

Regional Tab Settings. You'll notice there are nine tabs across the top of the SPA-3000 Admin Config page. Start by clicking on the Regional tab and set your Time Zone. This matters because POTS phones connected to the Phone jack get their time from the SPA-3000. The Time Zone pull-down is in the Miscellaneous section of the form toward the bottom of the screen. The SPA-3000 doesn't know about Daylight Savings so pick your time zone accordingly. Save your change by clicking the Submit All Changes button. The SPA-3000 will reboot.

Line 1 Tab Settings. In the Proxy Registration section, enter the IP address of your Asterisk box in the Proxy field, set Register to Yes, set Register Expires to 60, and Use Outbound Proxy should be No. Leave all the other settings as you find them. In the Subscriber Information section,

Set Auto PSTN Fallback to Yes. Finally, one could write a book on Dial Plan settings, and someone probably should. But it won't be me. While I'm by no means an expert, I'm learning fast. So here's a Dial Plan configuration to get things working, and we'll revisit it in a future column to add additional features and safeguards. For now, plug the following into the Dial Plan field at the bottom of the form after erasing what's already there. Then Submit All Changes:

(#xx|< :*>*xxxS0|*xx|[3469]11|0|00|1[2-9]xx[2-9]xxxxxxS0|[2-9]xxxxx.|[2-9]xx[2-9]xxxxxxS0|xxxxxxxxxx.)

PSTN Line Tab Settings. Things get a little tricky in this form so type carefully, or you'll have a real mess when you try to receive calls. In SIP Settings, set the SIP Port to 5061. In the Proxy Registration section, enter the IP address of your Asterisk box in the Proxy field, set Register to Yes, set Register Expires to 60, and Use Outbound Proxy should be No. Leave all the other settings as you find them. In the Subscriber Information section,

should be entered for Dial Plan 8 in the Dial Plans section. That's a less than symbol, then S, then zero, then a colon, 99, and a greater than symbol. In the VOIP-To-PSTN Gateway Setup, set Gateway Enable to Yes and Caller Auth Method to None. In the PSTN-To-VoIP-Setup, set Gateway Enable to Yes, PSTN Ring Thru Line 1 to Yes, PSTN CID for VoIP CID to Yes, PSTN Caller Default DP to 8, and PSTN CID Number Prefix to 00 (that's zero-zero).

In the FXO Timer Values section, set VoIP Answer Delay to 1, PSTN Ring Thru Delay to 3, and PSTN Ring Thru CWT Delay to 3. Click Submit All Changes button to save your work thus far.

User1 Settings. In the Selective Call Forward Settings section, set Cfwd Sel1 Caller to 00* (zero-zero-asterisk). Set Cfwd Sel1 Dest to 99. And, at the bottom of the form, set VMWI Ring Policy to New VM Arrives. If you forget to change this last one, your phone will ring once a minute, all night long whenever anyone leaves you a voicemail message in the middle of the night. Click the Submit All Changes button, and we're finished configuring the SPA-3000 ... at least for today.

Tweaking the Asterisk@Home Dial Plan to Support the SPA-3000. Our final drill for today is to tell Asterisk about the SPA-3000, add a few custom routines to our Extensions_Custom config file, and test everything to make sure it's working. Whew! No one said this would be easy, did they?

Let's first add the PSTN trunk. Call up AMP->Setup->Add SIP Trunk. For the Outbound Caller ID, enter your home phone number with area code. For maximum channels, type 1. Skip the Outgoing Dial Rules section as well as the Incoming and Registration sections. In the Outgoing Settings, let's name the trunk pstn. For PEER Details, enter the following but not our comments on the right. Then click the Submit Changes button and then the red bar.


auth=md5
context=from-internal
dtmfmode=rfc2833
fromuser=asterisk < -- don't change this host=192.168.0.115 <-- plug in the internal IP address of your SPA-3000 here insecure=very port=5061 <-- important! secret=123456 <-- use your password for extension 204 here type=peer username=asterisk <-- don't change this

Next we need to reconfigure our Outbound Routing a bit. Let's add a new Outbound Route and name it OutPSTN. For the Dial Rules, enter the following:


911
NXXNXXXXXX
NXXXXX.


For the Trunk Sequence, choose SIP/pstn and then add SIP/bv. Be sure they appear in this order or press one of the blue arrow keys to reorder them correctly. Now click Submit Changes and then click the red bar. When the list of trunks reappears, make sure they are in the following order (top to bottom): OutVoxee, OutPSTN, Outside. If not, click on the arrows to reorder them, submit changes, and click the red bar.

Modifying Extensions_Custom to Handle Incoming PSTN Calls. Click on AMP->Maintenance->Config Edit and then choose the extensions_custom.conf file. When the editor opens, move down several lines to an opening in the [from-internal-custom] context. Insert the following code. WARNING: If you cut and paste code from these articles and the code contains quotation marks (such as below), be sure to replace the WordPress-inserted, front and back quotes with normal quotation marks, or you’ll send Asterisk into the ozone.

;next extension (99) is to handle incoming PSTN calls
exten => 99,1,GotoIf($["${CALLERIDNUM:0:2}" = "00"]?2:3)
exten => 99,2,SetCIDNum(${CALLERIDNUM:2})
exten => 99,3,SetMusicOnHold(default)
exten => 99,4,Answer
exten => 99,5,Wait(1)
exten => 99,6,Background(custom/welcome)
exten => 99,7,DigitTimeout,2
exten => 99,8,ResponseTimeout,2
exten => t,1,Answer
exten => t,2,Wait(1)
exten => t,3,Background(pls-hold-while-try)
exten => t,4,Dial(SIP/204&SIP/200,20,m)
exten => t,5,VoiceMail(204@default)
exten => t,6,Hangup
exten => i,1,Answer
exten => i,2,Wait(1)
exten => i,3,Playback(wrong-try-again-smarty)
exten => i,4,Goto(99,5)


The only line above that you'll need to modify is t,4 where you'll need to specify the numbers of each extension you want to ring when a call comes into Asterisk from your PSTN (home phone) line. If you have more than one extension, separate them with an ampersand and use the SIP/extensionnumber syntax for each extension. Now add the following code snippet at the bottom of the file after adding include => custom-recordme-code near the top of the file in the [from-internal-custom] context :

[custom-recordme-code]
exten => 456,1,Playback(custom/record-msg)
exten => 456,2,Wait(2)
exten => 456,3,Record(/tmp/asterisk-recording:gsm)
exten => 456,4,Wait(2)
exten => 456,5,Playback(/tmp/asterisk-recording)
exten => 456,6,Wait(2)
exten => 456,7,Hangup


Recording Voice Prompts with Asterisk. Save your changes by clicking the Update button and then restart Asterisk as previously described. Now pick up a phone and dial 456#. When you hear a beep, say "Please say your voice prompt at the tone. Press the pound key when you are finished." Then press the # key, listen to your recording, and hang up. If you want to revise it, just repeat the steps. Now go to the Asterisk console and log in as root. Change to the tmp directory: cd /tmp. Rename the custom recording we just made: mv asterisk-recording.gsm record-msg.gsm. Now move the file to its permanent home: mv record-msg.gsm /var/lib/asterisk/sounds/custom. Now let's record one more. Dial 456# again. At the tone, say "Hi, you've reached the Rockefeller's. Someone will be right with you." Now press # to save your recording. Now go back to the Asterisk console. Go to the /tmp directory. Rename the recording: mv asterisk-recording.gsm welcome.gsm. Then move the file to its permanent home: mv welcome.gsm /var/lib/asterisk/sounds/custom.

Why are we doing all of this? Well, there really is a reason. Now when someone (like you) calls that knows what's going on, you can dial any extension on your Asterisk system while this mundane greeting plays. You can even call up the weather.

At this point, everything should be working. You should be able to dial in using either your home phone number or your BroadVoice number. And you should be able to dial out using the dialing rules we outlined when we began. Finally, a word of caution. Being human, we sometimes forget a step in a process as complex as this was. We've actually built our new system at the same time we wrote this so everything oughta work fine. Ours does. If not, post a comment (don't email me with technical problems!), and we'll have a look. For those that are curious, we purchased an Ice Cube HU61 from Now Micro in St. Paul, Minnesota. For just over $500, it includes an AMD Athlon 64 XP3000 processor, 512MB of RAM, a 250GB Seagate 7200 ATA IDE drive (don't buy SATA drives as they won't work with this version of Linux and Asterisk), a 52X CD ROM drive, and a 3 year parts and labor warranty. Remember, you don't need an operating system for this machine. It's on the Asterisk@Home installation CD. The Ice Cube has the capacity to handle about 1000 phone extensions and roughly 200 simultaneous SIP calls so it'll do just fine for home use unless we have more than a few Strom Thurmond "moments." And, on the weekends, you can disconnect your phones and take your ass-kickin' Ice Cube to the finest LAN parties in town. See you next week.

There are numerous additional articles in this series now. You can read all of them by clicking here.