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Asterisk on Steroids: The Orgasmatron Installer, Part III

Happy Cinco de Mayo! And you can celebrate the event by installing two dozen turnkey Asterisk® applications in under 5 minutes! We recently introduced our new Orgasmatron Installer for PBX in a Flash. And today the saga continues with Part III in our series. Faxing and email work out of the box. More than a dozen extensions and a number of hosting provider trunks are preconfigured. Delivery of CallerID names with numbers is available from a half dozen providers of your choice. ODBC database connectivity is now painless. And the Flite text-to-speech engine is preconfigured with Cepstral TTS only a few keystrokes away. Also included are FreePBX 2.5, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Here's the complete list of what 5 minutes of your time brings to your Asterisk server platform:

In Part II of this series, we walked you through securing your system and configuring a few of the major applications: AsteriDex, CallerID Superfecta, CallWho, Cepstral, and Emailing with SendMail. Today, we'll tackle nine more applications in the list.

Fax Module with nvFax. The NVfax module provides basic incoming and outgoing fax functionality for your PBX in a Flash system. It's not perfect because faxing with VoIP providers is hit and miss at best! As installed, inbound faxing works after a simple configuration. Here are the three steps:

#1. Log into your server as root and edit fax-process.pl in the /var/lib/asterisk/bin folder. Change the following default parameter to make it your default MAILTO email address:

my $to = "JoeSchmoe\@gmail.com";

NOTE: Always edit system files like this: nano -w filename

#2. Using a web browser, log into FreePBX and choose Admin, Setup, General Settings. In the Fax Machine section of the form, choose system as the extension for receiving faxes, enter the destination email address for incoming faxes, and enter an email from address for outbound faxes.

#3. While still in FreePBX, you need to define how you want faxes processed when they are received from outside your PBX. Choose Admin, Setup, Inbound Routes. For each incoming route on your PBX where you want to enable receipt of faxes, click on that incoming route definition. In the Fax Handling section of the form, choose system as the fax extension, enter the fax email destination address, choose nvfax as the fax detection type, and use 5 as the fax detection delay setting. Save your settings for each inbound destination and then reload your dialplan.

You can test it by plugging a real fax machine into a VoIP phone adapter such as the Linksys SPA-2102 and assigning the ATA an extension number on your PBX. Using the fax machine, simply send a fax to extension 329 (F-A-X). It should arrive as a PDF in your email inbox within a couple minutes.

Once you get fax delivery of faxes from inside your PBX working reliably, then you're ready to graduate to the Big League and get faxing from outside your PBX working. This is 99% dependent upon the quality of inbound calls from your DID provider. If your DID provider doesn't support ULAW, give up or switch providers. We have successfully tested inbound faxing with TelaSIP, Teliax, voip.ms, and Future-Nine. With Teliax and Future-Nine, you will need to add the following settings to your Incoming Trunk Configuration in FreePBX:

t38pt_rtp=no
t38pt_tcp=no
t38pt_udptl=no

For additional tips and tricks, read our Best of Nerd Vittles article on faxing.

FONmail for Asterisk. FONmail is one of several applications that works in conjunction with AsteriDex. It lets you pick up a telephone connected to your Asterisk system, dial 6245 (M-A-I-L), and dictate a message for email delivery to someone in your AsteriDex database. You'll be prompted for the phone number of your recipient, or you can look up a person using the first three letters of their name in the AsteriDex database. Once you record your message and choose the recipient, the dictated message is emailed to the recipient using the email address you've entered for that person in AsteriDex.

For FONmail to work, you obviously have to add entries into AsteriDex (with email addresses) for the recipients you intend to select, and you need to populate the new dialcodes for AsteriDex by following the instructions in Part II of this tutorial. The final piece is specifying your return email address for the outbound emails. Set your return email address by editing the $email entry at the top of nv-mailit.php. The file is stored in /var/lib/asterisk/agi-bin.

FreePBX Backups. A disaster recovery plan is a critical component with any computer system, and PBX in a Flash is no different. You need to have a plan for recovering from a disaster whether that disaster is an Act of God, or man-made, or the result of a hardware failure. Our recommended strategy goes like this. Make weekly full disk backups with Mondo to at least a pair of USB flash drives. Replace the drive each week and take the other drive off site. In addition, make daily or weekly FreePBX backups and copy them to a safe place. Amazon S3 offers a convenient, inexpensive off-site storage facility for FreePBX backups. FreePBX backups let you restore FreePBX components to a machine state at the time the backup was made. Here's how to set up FreePBX automatic backups. Be sure you clean out old backups from time to time as they take up disk space. The backups are stored in folders under /var/lib/asterisk/backups based upon the name you assign to your backup schedule.

Here's how to set one up to make a backup on demand:

1. Open FreePBX with your web browser.
2. Choose Admin, Tools, Backup and Restore, Add Backup.
3. Give the backup schedule a name, e.g. RightNow.
4. Change all Radio buttons to Yes to backup everything.
5. Backup schedule: Run Backup Now.
6. Click Submit Changes button to kick off the backup.

Here's how to set one up to make a weekly backup every Sunday night:

1. Open FreePBX with your web browser.
2. Choose Admin, Tools, Backup and Restore, Add Backup.
3. Give the backup schedule a name, e.g. Daily.
4. Change all Radio buttons to Yes to backup everything.
5. Backup schedule: Run Backup Weekly (on Sunday).
6. Click Submit Changes to save new backup schedule.

Gizmo5 FreePBX Module. One of the VoIP providers that provides enormous flexibility in getting the most out of your new system is Gizmo5. For very little money and virtually no configuration hassles, Gizmo5 can't be beat. One of the slick functions that Gizmo5 provides is the ability to make 5-minute phone calls to any Skype user at no cost. For $20 a year, you can make as many 2-hour Skype calls as you like to your ten best friends. For more details, see our article. The Orgasmatron installer puts everything in place for you to set up a Gizmo account quickly from within the FreePBX interface. Just choose Admin, Setup, Gizmo5 Integration. Just follow the prompts to create your new account and make an initial deposit.

Installing the Hamachi VPN. Once you've run the Orgasmatron Installer, you have the option of installing the Hamachi virtual private network (VPN) which supports the interconnection of 16 computers at no cost. Simply run the install-hamachi.x script which you'll find in your /root/nv folder. For complete configuration instructions, read the install-hamachi.pdf file and hamachi.faq, both of which are also in the same directory.

Interconnecting Asterisk Servers with IAX. If you don't plan to interconnect your Asterisk server with one or more other Asterisk servers, then delete the Remote-Host outbound route in FreePBX and then delete the remote-peer trunk. If you plan to use the ODBC demo examples on extensions 222 and 223, you at least will need to change the Dial Pattern for the Remote-Host outbound route by deleting the 2XX entry as explained elsewhere in this article. What this provided was a simple way to interconnect extensions in the 200-299 range of numbers on a remote PBX.

If you do plan to interconnect Asterisk servers, then change this 2XX Dial Pattern to match the extension numbers on your remote PBX. For example, if the remote Asterisk server uses extensions in the 7000-7999 range of numbers, you'd want to include a 7XXX entry in your Remote-Host Dial Pattern.

To enable, interconnection of your new server to another Asterisk server, edit the remote-peer trunk and insert the actual IP address of your remote host. Also change the secret in the Peer and User sections to a very secure entry and use the same secret entry in your remote host trunk setup.

On the remote server, create a new IAX trunk with settings like the following using your correct secret and the IP address of your new server that was built with the Orgasmatron Installer:

MeetMe Conferences On the Fly. If you're accustomed to spending hundreds of dollars to schedule and run phone conferences with dozens of people, those days are officially over with PBX in a Flash. You now can purchase a phone number in 2600+ rate centers in the United States with support for 20 simultaneous calls for under $9 a month. Once you have purchased your DIDforSale DID and configured the new trunk on your server, simply point the inbound route for that trunk to Misc Destination: MeetMe CONF.

To set up a conference at any time, pick up any phone on your PBX and dial 2663 (C-O-N-F). When prompted for the conference number, make one up, e.g. 30303. When prompted for a conference PIN, make one up, e.g. 1234. Now notify all conference participants to dial the Conference DID (or 2663 for internal users) and to use 30303# for the conference number and 1234# for the PIN. When everyone hangs up, the conference ends. Simple as that!

ODBC Database Connectivity. All of the necessary components to support ODBC database integration with Asterisk have been installed for versions of the Orgasmatron Installer after May 1. Also included are two sample dialplan components that demonstrate how to build ODBC applications. These two samples are explained in the Nerd Vittles ODBC article. The extensions used by these two samples are 222 and 223. If you used an older version of the Orgasmatron Installer, you'll have to manually add ODBC support and the sample extensions conflict with the default routing rules for interconnecting your server to another Asterisk server. So you have two options. Either change the Dial Pattern for interconnecting to the remote server by deleting the 2XX entry or modify the extension numbers for the ODBC demos in /etc/asterisk/odbc.conf. Once you have addressed this inconsistency, you can activate the ODBC demo applications by inserting the following line in the [from-internal-custom] context of extensions_custom.conf in /etc/asterisk: #include odbc.conf

Then reload your Asterisk dialplan: asterisk -rx "dialplan reload"

Reminders by Phone and by Web. The latest version of the Best of Nerd Vittles Telephone Reminders 4.0 application is included in the Orgasmatron Installer. You can schedule reminders by telephone by dialing 1-2-3 from a phone connected to your Asterisk PBX. The default password is 12345678. To keep strangers from using your reminder system, you need to change this password. Edit extensions_custom.conf in /etc/asterisk and search for the 123 extension. Change the password entry in the Authenticate entry and reload your dialplan as shown above.

You also can schedule reminders using a web browser. There's an option in FreePBX: Admin, Tools, Reminders. You also can access the reminders application separate and apart from FreePBX using the IP address of your Asterisk server: http://ipaddress/reminders.

The CallerID number for the application, the TTS engine, and your email address all can be adjusted to meet your needs. See the Best of Nerd Vittles article for details on making these changes.

Continue reading Part IV (Monday, May 25).


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FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

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Some Recent Nerd Vittles Articles of Interest...

Asterisk on Steroids: The Orgasmatron Installer, Part II

In our last column, we introduced you to the new Orgasmatron Installer for PBX in a Flash. After a one-week break to prepare for our visit to the Atlanta Asterisk® Users Group 3d Annual InstallFest, we're back in the saddle today to flesh out the new baby.

For those that are new to all of this, let's briefly review what the Orgasmatron Installer has added to your Lean, Mean Asterisk Machine. Faxing and email now work out of the box. More than a dozen extensions and a number of hosting provider trunks are preconfigured as well. Delivery of CallerID names with numbers is now available from a half dozen providers of your choice. And, of course, the Flite text-to-speech engine is preconfigured with Cepstral TTS only a few keystrokes away. Also included are FreePBX 2.5, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. And here's the complete list with all of your new Nerd Vittles applications:

Security First! Because your phone bill matters, today we begin with security. The design of virtually all of the open source Asterisk PBX aggregations is to leave SIP and IAX ports on your new server exposed to the Internet. This is done to facilitate communications with your hosting providers as well as telephone extensions which may be connected to your server from the other side of the globe. The wrinkle with this design is that, if a bad guy can guess an extension number on your system and its password, they get a free ticket to do whatever could be done from that extension on your PBX. In the case of one unlucky company, this resulted in a phone bill of over $100,000. For details, read our Primer on Asterisk Security. So... Security Matters!

Anyone obviously can download PBX in a Flash and the Orgasmatron Installer. Thus, you need to assume that everyone on the planet knows your default passwords. We walked you through changing some of the important ones with the passwd-master script last week. Use it regularly. Now let's turn our attention to your extensions and trunk passwords.

Extension Security. There now are a couple of ways to secure your extensions from the bad guys. First, you need to establish very secure passwords for your extensions and voicemail boxes. Second, you need to specify the IP addresses that are authorized to access every extension on your PBX. And third, remember do repeat this drill every time you add a new extension to your system.

To change an extension password, open FreePBX using a web browser pointed to the IP address of your server: http://ipaddress/admin/. On PBX in a Flash systems, you'll be prompted for a username (maint) and whatever password you set when you ran passwd-master. Now click the Setup tab and then the Extensions option. You'll see the list of configured extensions on your PBX in the right column. Click on each of those extensions, and you'll see a form like this:



The password for this extension is stored in a field called secret. Make up a very secure password for every extension on your PBX. You will embed this password in the telephone connected to this extension. There's no other place you'll need it so a long and complex numeric password is essential.

The authorized IP addresses for this extension are stored in a field called permit. The way this works is that you first specify which IP addresses should be denied access (the deny field), and then you poke a little hole in the dike, if you're smart, to permit only one or a few IP addresses to connect to the extension. Leave the deny entry as it is. The default permit entry 0.0.0.0/0.0.0.0 opens the floodgates. It means any IP address can log into this extension. To restrict extension access to IP addresses on a private LAN of 192.168.1, the entry would look like this: 192.168.1.0/255.255.255.0. To further restrict extension access to a specific IP address (recommended!), the entry would look like this: 192.168.1.44/255.255.255.255. Use a permit entry that makes you sleep well at night. After all, it's your phone bill.

The third entry you'll want to change is further down the same data entry form, and that's the Voicemail Password field. This entry determines who can actually retrieve voicemails left for this extension. Set it accordingly.

Once you've made the three changes above, save your entries by clicking the Submit button at the bottom of the form. Repeat the drill for every extension, and then click the orange Apply Configuration Changes tab at the top of the screen and then Continue with Reload to reload your Asterisk dialplan.

Trunk Security. Securing the trunks on your PBX is equally important to securing extensions. Keep in mind that, with your trunk credentials, anyone can set up your trunk on their PBX to make calls on your nickel! Unlike the extensions, there are no working usernames and passwords in the default trunks with one exception. If you plan to use the providers we've preconfigured, simply insert your own username, fromuser, and secret settings in the fields provided, and you'll be making calls in a matter of seconds. The process is similar to the one we used for extensions. Choose Setup, Trunks and then click on each trunk and make your entries. Submit your entries and then reload the dialplan when you're finished.

In the case of the remote-peer trunk, this trunk is designed to make it extremely easy to interconnect Asterisk servers for interoffice communications. But it also means that a bad guy can easily interconnect with your server and start dialing. If you don't plan to connect to another Asterisk server, delete this trunk! If you do plan to connect to another Asterisk server, change the trunk secret and IP address of the host to which you are connecting. Do NOT leave the default secret in either the outgoing or incoming settings! Also change the password for the outbound route: Remote-Host. You may want to ultimately remove this password if you actually start interconnecting servers. Otherwise, users will have to enter this password whenever they may a call to an extension on the interconnected Asterisk server.

To interconnect your server to another server, you would simply add a new trunk called main-peer on the other server that looks like this (using your new password and correct IP address):


Configuring AsteriDex. AsteriDex is plug-and-play for most users. However, as configured, your AsteriDex web site is reachable from the Internet if you have mapped port 80 on your hardware-based firewall to your PBX in a Flash server or if you don't have a hardware-based firewall and your server is directly exposed to the Internet. If you don't mind people seeing your contact list or making prank calls that ring your extensions, this may be okay. If it's of concern to you, the easiest security precaution is to rename the asteridex4 directory to an obscure name that only you know, e.g. bahbah143. Here are the commands to issue after logging into your server as root. By using all of these commands, AsteriDex still will be accessible through FreePBX and the PBX in a Flash GUI:

cd /var/www/html
mv asteridex4 bahbah143
sed -i 's|asteridex4|bahbah143|' admin/modules/asteridex/page.asteridex.php
sed -i 's|asteridex4|bahbah143|' welcome/.htindex.cfg

The other adjustment you may need to make to AsteriDex is to configure who can access the Admin tab to add, modify, and delete entries in your database. As configured, the Admin tab is available to any computer with an IP address that begins with 192.168. This may not match your private subnet, and not all 192.168 IP address are non-routable. So you may wish to tighten this restriction to match your internal subnet. In the /var/www/html/asteridex4 folder (or whatever name you've chosen above), you'll find a configuration file: config.inc.php. Simply edit this file and change the $local_net entry. You also can set the long distance prefix ($LDprefix), your CallerID number ($CallerID), and the default extension to ring for click-to-dial from the web interface ($INtrunk and $defaultExt). The extension to dial can now be set from the web interface as well. Unless you really know what you're doing, leave everything else the way it is.

CallerID Superfecta. Most hosting providers deliver CallerID numbers as part of your payment for using their DIDs. Almost none deliver CallerID names without an additional charge. CallerID Superfecta is designed to fill that gap... for free. A number of us have worked on this project for years. And it now has been integrated directly into FreePBX. There are two steps to getting everything working properly on your new PBX. First, you need to identify which CallerID lookup sources you wish to use on your system. Then, you need to specify CallerID Superfecta as the lookup source on each Inbound Route where you want CallerID names looked up for incoming calls.

Open FreePBX with your web browser and navigate to Setup, CID Superfecta. You'll get a form that looks like this:


With the exception of AsteriDex and SugarCRM lookups which are almost instantaneous, keep in mind that each lookup takes a little time and slows down receipt of your inbound call. So long as you have a good Internet connection, you shouldn't have a problem using all of the sources. The way the CallerID Superfecta works is that, once it gets a name match in any of the sources beginning with AsteriDex and SugarCRM, it ends the lookups and provides the CallerID name it found to Asterisk for display on the extensions which are ringing in the designated inbound route. Filling out the form is self-explanatory for the most part. Tick off the lookup sources you wish to use. If you plan to use whocalled.us, you'll need to sign up for an account and provide your credentials before the lookup will work. With SugarCRM, fill in the blanks to match your implementation of SugarCRM. Click the SAVE button when you have CallerID Superfecta configured to meet your needs.

The final step in implementing CallerID Superfecta is to designate it as the CallerID Lookup Source for your Inbound Routes. Click on Setup, Inbound Routes and a list of your existing routes will be displayed in the right column. As installed, there will only be one: Any DID / Any CID. Click on this entry to display the form. Scroll down to the CallerID Lookup Source dropdown box and choose CallerID Superfecta. You'd do the same with any other inbound route you create down the road. Click the Submit button and reload your dialplan to enable CallerID Superfecta. Now sit back and wait on your first call.

CallWho for Asterisk. CallWho for Asterisk is a little script we put together to make it easy to look up and dial the numbers of people in your AsteriDex database. When you dial 4-1-2, you'll be prompted to enter the first three letters of the name of the person you wish to call. Once you key in the three letters, CallWho for Asterisk will look up every matching entry in your AsteriDex database and read you the list of matches. For example, if you had Joe Schmo and Joe The Plumber in your database, CallWho would say something like this:

Press 1 for Joe Schmo.
Press 2 for Joe The Plumber.

When you press 2, CallWho will place a call to Joe The Plumber. Not sure why you'd ever want to do that, but now you understand the way it works.

Before CallWho for Asterisk will work at all, you need to run the script which associates three letter codes with every entry in your AsteriDex database. And, whenever you add new entries to your database, you need to run it again. Using a web browser, here's the program to run. Be sure to use the correct IP address for your Asterisk server and your newly designated AsteriDex location instead of asteridex4:

http://192.168.0.44/asteridex4/dialcode.php

Cepstral TTS for Asterisk. PBX in a Flash is delivered with the Flite text-to-speech engine already enabled. But, unless you like the voices of Lurch and Fred Munster, you may wish to cough up a little cash and install Cepstral on your server. Cepstral now has a synthesized voice of Allison which exactly matches all of the other voice prompts in Asterisk. I'm embarrassed to report that we can't seem to get the correct installation script deposited in our Orgasmatron builds... ever! So, if you want to use Cepstral, here are the steps to download the real, working installation script and to install Cepstral:

cd /root/nv
rm install-cepstral
wget http://pbxinaflash.net/source/cepstral/install-cepstral
chmod +x install-cepstral
./install-cepstral

Once the 65MB download completes, you'll be prompted to agree to the license. You do this by pressing the Enter key to scroll down the license agreement. When you reach 100%, type yes to continue with the install. Press Enter to accept /opt/swift as the install directory. Very important: Type y to create the directory. The default is No which will mess up the installation. Now type yes to complete the install. Once the install completes, you can purchase a license for the Allison voice at this link. Under Voices, choose Language: US English, Voice: Allison-8kHz, and Platform: Linux. For non-commercial use, the $30 voice registration is all you need. For commercial use, you also need to acquire Concurrency Licenses which authorize a certain number of simultaneous voice ports on your system for Cepstral voices. These run $50 per port in 2-port multiples and are in addition to the $30 Allison voice license. For Nerd Vittles readers, you can save 15% on your purchase by sending an email to sales at cepstral.com explaining how you plan to use Cepstral and requesting the discount code.

We'll have an in-depth article on Cepstral in coming weeks. For those that want a head start, each of the Nerd Vittles text-to-speech applications typically includes dialplan code and one or more PHP/AGI scripts. The dialplan code can be found in /etc/asterisk/extensions_custom.conf. When you scroll through the dialplan code you will see entries like the following for each of the TTS applications:

exten => 611,5,Flite("Enter a 3 character airport code.")
;exten => 611,5,Swift("Enter a 3 character airport code.")
exten => 611,6,Read(APCODE,beep,3)
exten => 611,7,Flite("Please hold a moment.")
;exten => 611,7,Swift("Please hold a moment.")

The semicolon at the beginning of a line tells Asterisk this is a comment and to ignore it. To change the voice from the Munsters to Allison, just comment out the Flite lines and uncomment the Swift lines by deleting the leading semicolons. When you're finished making the changes, save the file and then reload your dialplan: asterisk -rx "dialplan reload". So, in the example above, the code would now look like this:

;exten => 611,5,Flite("Enter a 3 character airport code.")
exten => 611,5,Swift("Enter a 3 character airport code.")
exten => 611,6,Read(APCODE,beep,3)
;exten => 611,7,Flite("Please hold a moment.")
exten => 611,7,Swift("Please hold a moment.")

You also need to modify the PHP/AGI scripts that go with each application. All of these files are stored in /var/lib/asterisk/agi-bin. Typically the filenames begin with nv- and end in .php:
-rwxrwxr-x 1 asterisk asterisk 6835 Sep 16 2008 nv-callwho.php
-rwxrwxr-x 1 asterisk asterisk 201 Jul 12 2006 nv-config-555.php
-rwxrwxr-x 1 asterisk asterisk 201 Apr 2 13:08 nv-config.php
-rwxrwxr-x 1 asterisk asterisk 14329 Feb 10 2008 nv-mailcall.php
-rwxrwxr-x 1 asterisk asterisk 6072 Sep 24 2008 nv-mailit.php
-rwxrwxr-x 1 asterisk asterisk 10490 Apr 20 10:34 nv-news.php
-rwxrwxr-x 1 asterisk asterisk 6545 Apr 12 15:10 nv-today.php
-rwxrwxr-x 1 asterisk asterisk 21537 Apr 2 13:07 nv-weather.php
-rwxrwxr-x 1 asterisk asterisk 12043 Apr 2 13:07 nv-weather-world.php
-rwxrwxr-x 1 asterisk asterisk 22243 Apr 2 13:07 nv-weather-zip.php

In each of these scripts, you'll find a variable near the top that controls the TTS engine: $ttspick = 0 ;

To use Cepstral as the TTS engine instead of Flite, just change the $ttspick value from 0 to 1 and save the file.

Email That Works With SendMail. It's always been a knuckle drill to get your new server to reliably send outbound emails. Assuming your Internet service provider doesn't block downstream mail servers, the Orgasmatron Installer will get this working reliably. You can test it out by logging into your server as root and issuing the following command using your real email address. If you get the email, you can move on.
echo "test" | mail -s testmessage yourname@gmail.com

If you didn't get the email, you probably have a provider such as Comcast that blocks port 25 in many areas of the country. The easiest way to solve this is to set up a free Gmail account and use Gmail to deliver outbound messages from your server. This message thread on the PBX in a Flash Forum will walk you through the setup process. There's also a Comcast solution if you'd prefer not to use Gmail.

Stay Tuned. Your eyes are probably glazing over about now. I know mine are. So we'll quit here for today. In our next episode, we'll tackle the rest of the goodies that make up the Orgasmatron Installer. Enjoy!

Continue reading Part III.

Continue reading Part IV (Monday, May 25).


Tip of the Week. Ever wanted a 20-seat conference bridge for under $9 a month with a local phone number in any of 2600+ rate centers all over United States? You can add load balancing and automatic failover for an extra $1 per month. After you use the Orgasmatron Installer, just set up a conference extension in FreePBX and then head over to the PBX in a Flash Forum to read all about the latest rage in DID providers.


whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 


Some Recent Nerd Vittles Articles of Interest...

Asterisk on Steroids: Introducing the Orgasmatron Installer

If an Asterisk® distribution with every bell and whistle on the planet is at the top of your Wish List, then the new Orgasmatron Installer may just be your cup of tea. Let’s face it. The Asterisk learning curve is horrendous. As some of you know, we have built some custom PBX in a Flash systems for the Dell, Everex, and Atom platforms. These builds differ from the PBX in a Flash base install in that they were turnkey PBXs with dozens and dozens of custom applications, extensions, and trunks already preconfigured. While you still needed to change some passwords and plug in some phones, the Orgasmatron builds reduce the Asterisk learning curve to almost zero. Out of the box, email works. Faxing works. ENUM works. Interconnecting Asterisk servers for free calling works. And extensions for 15 phones already are in place. Plug in your Vitelity credentials, and you can place calls to any phone in the world using your new VoIP PBX in a couple of minutes. That’s the good news.

The problem with these builds lies in their basic architecture. To date, all of them were really Mondo backups. And once you strayed from the platform on which the original system was built, your odds of getting a successful restore went down the toilet quickly. Well, that was then. And this is now!

Today we introduce an installation script for PBX in a Flash that lets you build a PBX in a Flash base system, run the Orgasmatron Installer script, and boom! Within a few minutes, you’ve got an Asterisk-based Orgasmatron server on the computer platform of your choice regardless of processor, disk controller, disk drive, network card, and video adapter. And it works equally well in a virtual environment using an open source platform such as the fantastic and free Proxmox Virtual Environment.

Update: Be sure to check out the latest Orgasmatron V Installer at this link.

For those that are wondering what’s included in this new Orgasmatron build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with Asterisk 1.4 or 1.6, FreePBX 2.5, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin:

Getting Started. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some precautions to protect your phone bill. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you’re still not asleep, there’s loads of additional documentation on the PBX in a Flash documentation web site.

Installation. Here’s a quick tutorial to get you started. First, install the 32-bit version of PBX in a Flash with Asterisk 1.4. Boot your system from the installation CD and type ksalt to begin. When your machine reboots, remove the CD and choose option A to load the most stable payload. When the install completes, reboot your system once again and login as root with the password you chose when you built your system. Now issue the following commands to bring your system current and protect your system passwords: update-scripts, update-fixes, passwd-master. You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.

Now you’re ready to run the Orgasmatron Installer. While still logged into your new server as root, issue the following commands:

cd /root
wget http://pbxinaflash.net/orgasmatron/orgasmatron.x
chmod +x orgasmatron.x
./orgasmatron.x
reboot

Stick around while the install script is running. Parts of it are interactive. For now, choose the Flite option when you’re prompted for text-to-speech preferences. That way you’ll have a working system when you’re finished. Once the installer script is finished, type status and write down the IP address of your server. You’ll need it in the next step to log into FreePBX.

Using a web browser, open FreePBX on your new server with a command like this (substituting the IP address you wrote down above). When prompted for your account name, type maint and use the password you assigned when running passwd-master above:

http://192.168.0.123/admin/

You’re NOT done yet!

These next four steps are important. They get all of the FreePBX modules installed and then restore the FreePBX backup set that’s at the heart of the Orgasmatron build. Just follow along here, and don’t skip any steps. It’s easy.

1. Choose Module Admin, Check for Updates online, Upgrade All, Process, Confirm, Return, Apply Config Changes, Continue.

2. Choose Module Admin, Check for Updates online, Download All, Process, Confirm, Return, Apply Config Changes, Continue.

3. Repeat the above #2 commands a second time.

4. Click on the Tools tab and choose Backup & Restore, Restore, RightNow, and select the .tar.gz file that is displayed. Then choose Restore Entire Backup Set, OK, Apply Config Changes, and Continue.

Securing Your System. You’re almost done. We always like to reboot the server just to make sure nothing got lost in the shuffle. When the reboot is finished, log into FreePBX with a browser again. Before you do anything else, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit after changing secret and Voicemail Password. Repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.

Now let’s change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue. Whew! Your system now is relatively secure. Follow the steps in the tutorials we recommended, and you’re ready to experiment. Plug in a SIP phone or softphone and configure it using one of the available extensions together with the secret for that extension.

Finally, be sure to change the credentials on all of your trunks to match those assigned by your providers. And, in the case of the remote-peer trunk, change the secret and IP address to match the identity on your host Asterisk server. If you don’t have another Asterisk server, change the password anyway so no one can break into your system. Better yet, just delete the trunk unless you plan to use it down the road. We’ll have more to say about this next week. For now, just make up your own, secure password to protect this trunk from outside access by unwanted visitors.

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then next week, we’ll cover in detail how to customize every application that’s been loaded. For openers, we recommend you set up an account with Vitelity using our special link below. This gives your PBX a way to communicate with every telephone in the world, and it also gets you a real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

The VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. After you’ve connected a phone to your new system, begin your adventure by dialing these 10 numbers:

  • D-E-M-O – Check out the Nerd Vittles Orgasmatron Demo
  • Z-I-P – Enter a five digit zip code for any U.S. weather report
  • 6-1-1 – Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 – Get the latest news and sports headlines from Yahoo News
  • T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
  • F-A-X – Send a fax to an email address of your choice
  • 4-1-2 – 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L – Record a message and deliver it to any email address
  • C-O-N-F – Set up a MeetMe Conference on the fly
  • 1-2-3 – Schedule a regular or recurring phone reminder
  • Dial *68 – Schedule a hotel-style wakeup call on any extension

Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. Then log into your server as root and look through the scripts added in the /root/nv folder. You’ll find all sorts of goodies to keep you busy. Enjoy!

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV (Monday, May 25).


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 


Some Recent Nerd Vittles Articles of Interest…

Another Dell with Asterisk, Dude: Introducing the Orgasmatron III for Dell’s New PowerEdge T100

Dell finally kissed its SC440 server goodbye last week so we've been scrambling for a replacement VoIP candidate for Asterisk® that has performance sufficient to serve as a 50 to 100-user small business PBX without breaking the bank. It turns out Dell's PowerEdge T100 introduced in September is strikingly similar to the SC440 both in performance, design, and even price, but it scales a bit better. If it walks like a duck, quacks like a duck, and is priced like chicken feed, that's good enough for us.

In early December, we got our first new T100: a Dual Core Intel® Pentium®E2180, 2.0GHz processor with 1MB Cache, an 800MHz FSB, two 80GB 7.2K RPM Serial ATA 3Gbps 3.5-in Cabled Hard Drives connected to the onboard SATA controller, 512MB of 667MHz DDR2 RAM, a DVD-ROM Drive, and an On-Board Single Gigabit Network Adapter for $299. Sound familiar? It should. The T100 special pricing was virtually identical to the $299 special on the SC440 except Dell now has thrown in a DVD-ROM drive in lieu of the SC440's CD-ROM drive. For $19 more, you can bring the system up to 2GB of RAM which is an excellent idea. If you missed out, don't fret. There will be another deal in a week or two. Even the regular pricing on this unit with a Celeron 1.8GHz processor, 2 gigs of RAM, and two 80GB drives is only $339. And international pricing is equally competitive. We haven't yet seen the $199 single-drive U.S. price that appeared regularly with the SC440, but it shouldn't be too long given the current economy.

As for scaling, if you're interested in a growth path, you'll love the T100 compared to the SC440. It supports numerous processors up to the Quad Core Xeon 2.83GHz with 2x6M Cache and 1333MHz FSB as well as two one-terabyte SATA drives (just don't buy them from Dell 😯 ). And, unlike the SC440, the T100 accepts up to 8GB of RAM. So the remaining question: "Will the SC440 Orgasmatron II build work with the T100?" And the answer is "sort of." But have no fear, we've put Humpty back together again and have added even more bells and whistles to the new Orgasmatron III custom-designed for the T100 today. It now includes your own, free and private Hamachi VPN cloud for up to 16 computers.

To get email alerts when the T100 again goes on sale, go to techbargains.com. Then click on Send Email Deal Alert and fill out the form entering T100 as your search term. Be sure to confirm the alert by replying to the email.

If you want a cash rebate on your Dell purchase, use our eBates link to Dell or click on the coupon image in the right column of this article. It takes less than 30 seconds to sign up, and you get $5 (and so do we!) plus you receive 2% cash back on your Dell small business purchases which can be deposited directly into your PayPal account.

We expect these units will follow in the footsteps of their SC440 cousin and go on sale roughly every two weeks... so be ready! The T100 also is good news for our international friends because Dell now markets this machine virtually everywhere in the world at very competitive prices. It's selling for 40% off in the U.K. and 299€ in many European countries as we speak.

For long-time readers, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i. But both of our previously anointed small business/home servers on which to run a production Asterisk system for 50-100 employees, the Everex gPC2 (aka "The WalMart Special") and the Dell SC440, are no more. So this build brings us current with Dell's very latest offering in the low-cost, high-performance server category and builds on the SC440 tradition of providing a quantum leap in performance and reliability compared with traditional home PCs. The ISO images you'll be downloading were captured as a backup on the flash drive of our new T100 lab machine. You can expect at least twice the performance on the PowerEdge T100 compared to the WalMart Special. Today's Orgasmatron III Build provides a preconfigured T100 installation on a 2-disk ISO image backup of the whole system using Mondo. And, NO, it won't work with any other hardware! Once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own T100. Wait to install any add-on cards until after you complete the Orgasmatron install. You must have a T100 configured as above, or this Mondo restore may not work. So accept no substitutes, or you may end up with an Electronic Brick instead of an Orgasmatron.

We've preconfigured some extensions on your new system as well as outbound and incoming trunks from some terrific providers including our second homegrown entry for VoIP terminations. Joe Roper and his business partner in Spain now offer a terrific IAX VoIP termination service. You can choose penny a minute service in the U.S. and most of Canada, or you can opt for premium VoIP service at about 2¢ a minute in the U.S. International rates also are VERY reasonable! You literally can sign up for service, plug in your phones, and have a system in full operation in under an hour.

If you've missed our previous Orgasmatron articles, suffice it to say this is the Ultimate Kitchen Sink for Asterisk. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... just 15 minutes! In fact, it will take less time to create your new system than it will take you to finish reading this article. Please do BOTH! The Orgasmatron III includes PBX in a Flash 1.3 in all its glory including Asterisk 1.4.21.2 running under CentOS 5.2 with a version of Zaptel that actually works with legacy cards, plus the newly released FreePBX 2.5, a full-function fax server, a full-disk backup and restore solution (that actually works!), the latest Hamachi VPN software, every imaginable Nerd Vittles text-to-speech application for Asterisk, and so much more. Complete documentation for the TTS apps is available here.

  • Inbound and Outbound VoIP Faxing Using nvFax... finally!
  • FONmail for Asterisk to send voice messages to any email address on the planet
  • AsteriDex RoboDialer and Telephone Directory
  • Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
  • NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
  • Weather Reports by Airport Code (TTS)
  • Weather Reports by ZIP Code (TTS)
  • Worldwide Weather Forecasts (TTS)
  • xTide for Asterisk (TTS)
  • MailCall for Asterisk: Get Your Email By Telephone (TTS)
  • TeleYapper 4.0 Message Broadcasting System
  • CallWho for Phone Lookup and Dialing of Entries in the AsteriDex Database (TTS)
  • TFTP Server with preconfigured setups for 10 Aastra 57i SIP telephones

In addition, you get dozens of preconfigured telephony applications and functions that would take even an expert the better part of a year or two to build independently. And, unlike all of the other distributions, we build Asterisk from source so it's simple to modify and upgrade whenever you feel the need. Here's a short list of what you have to look forward to:

  • Stealth AutoAttendant with Welcome and Application IVRs
  • Key Telephone Support Using Park and Parking Lot
  • Intercom/Paging Support
  • Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
  • DISA
  • Blacklisting with Web and Telephony Interfaces
  • CallerID Name Lookups from Numerous Providers
  • Weekly Automated System Backups to a Flash Drive
  • One Touch Day/Night Service
  • Music on Hold
  • Voicemail with Email Delivery of Messages and Pager Notification
  • Voicemail Blasting
  • Cell Phone Direct Dial
  • Call Forward: All, Busy, No Answer
  • Call Waiting
  • Call Pickup
  • Zap Barge
  • Call Transfer: Attended and Blind
  • Dictation Service with Email Delivery
  • Do Not Disturb
  • Gabcast
  • Phonebook Dial by Name
  • Speed Dial
  • Flite Text to Speech (TTS)
  • Windows Networking with SAMBA
  • Linux Firewall and Fail2Ban with SSH, HTTP, and SIP/IAX login protection
  • PBX in a Flash Software Update Service To Keep Your System Current
  • One-Click Cepstral TTS Install with Allison... Just Type install-cepstral

Prerequisites. As mentioned, you'll need a T100 configured with the specs outlined above including the 2GB RAM upgrade. We also recommend an 8GB USB flash drive on which to store automatic weekly backups of your new system. Just plug it into your new machine, and follow the simple steps below to activate Mondo. Every Sunday night, you'll get a new backup in ISO format on your flash drive. If something goes wrong on your system, copy the ISOs to CDs and reboot with Disk 1. It doesn't get any easier than that. And you can always check on the latest backup by issuing the command: usbcheck

Pay to Play. Greed has finally set in at Nerd Vittles. After all, Christmas is just around the corner! The download of this two-disk ISO image will set you back a whopping $10. In addition to covering the bandwidth and storage costs for the builds themselves, it also seems only fair that those using the builds help cover the hardware costs associated with these technology refreshes. When you compare our pricing to the Lime Green PBX offering from Dell... well, you don't really wanna know! There's one other little difference. Once you download our image from DreamHost, you are more than welcome to pass it along to as many of your friends and business acquaintances as you like. You can even do it electronically through the DreamHost Files Forever program. And, if you're inclined to host this image for your fellow man at no cost, be our guest... and thank you!

Bottom line: With a little patience waiting on Dell's next special, for about $300 and some lunch money, you'll have the slickest, newest, fastest, most reliable PBX and fax machine on the planet with rock-solid weekly backups and, of course, the availability of our one-of-a-kind PBX in a Flash Software Update Service! In fact, this may very well be The PerfectPBX™ even if we do say so.

Getting Started. Once you have your T100 in hand, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. You'll need a USB keyboard for typing temporarily. We also strongly recommend that you always keep your system running behind a NAT-based firewall/router. We strongly recommend the dirt-cheap dLink WBR-2310 WiFi router which handles NAT issues with VoIP masterfully. Don't redirect any ports to the machine and don't turn the PC on just yet.

Download the two ISO images for the T100 from here. Unzip the file and create two CDs from the ISO images. If you don't know how to create a CD from an ISO image, read that section from our previous article. In fact, read the whole article. It'll help you immensely down the road.

Once you've created your two CDs, turn on the T100 and quickly insert Disk 1 into the DVD drive and close the drive. When prompted, press F11 to choose the boot device and select the DVD-ROM drive. You'll note that the default T100 setup now apparently looks for a network boot device so you'll need to do a little BIOS reconfiguring, but you can do that at your convenience. F2 gets you into the T100 BIOS setup. Then choose Integrated Devices and, using the space bar, change Embedded Gb NIC from Enabled with PXE to simply Enabled. Press the escape key twice and then choose Save and Exit.

For now, choose the DVD-ROM drive as the boot device and proceed with the Mondo restore. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names and you really do have a T100, ignore them. Otherwise, halt the install by pressing CTL-ALT-DEL and remove the CD. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 10 minutes. You'll be prompted to insert Disk 2 and press Enter to finish the install. When the second CD finishes, eject it and wait for the prompt. Then type "exit" and press Enter. Your T100 will reboot, and you're ready to go.

After the reboot finishes, type root at the login prompt for your username and password for your password. The IP address assigned by your DHCP server should appear on the status screen. Write it down. If there is no IP address, your machine does not have network connectivity or access to a DHCP server with an available IP address. Correct the problem and reboot.

Securing Passwords. We're going to change five passwords now. For the time being (until you've done some reading), think up one really difficult password (that you won't forget) and use it for all five passwords. At the root@pbx:~ $ command prompt, type the following commands and type in your new password when prompted. Don't forget your password or you'll get to put in your two CDs and start over.

passwd
passwd-maint
passwd-wwwadmin
passwd-meetme
/usr/libexec/webmin/changepass.pl /etc/webmin root yournewpasswordhere

Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click the Admin tab, the password is password. Then choose the FreePBX Administration button. Log in as maint with your new maint password. Before you do anything else, change ALL of the 10 extension passwords to something very secure... as if your phone bill depended upon it! Click Setup, Extensions and then choose each extension, modify BOTH the device secret and Voicemail Password, and click Submit. When you finish all the extensions, then reload the dialplan to save your changes. Finally, change your DISA password to something very, very secure: Setup, DISA, DISAmain, PIN. Reload your dialplan once again to save your changes.

Regardless of what you may read elsewhere, the Orgasmatron III has all the very latest security patches as of today. If you want more security, take our advice and add a hardware-based firewall/router between your Internet connection and your new Orgasmatron III and don't expose port 80 (the web interface) to the Internet!

Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set. Either way, you need a permanent IP address for your machine when all is said and done. Once you have a permanent IP address, hop on over to dyndns.org and sign up for your own fully-qualified domain name (FQDN), e.g. mypbx.dyndns.org. You're going to need it for a whole host of things with your new PBX, and dyndns.org is about the easiest way to do it. Once you have your FQDN and DynDNS username and password, log in as root and edit: /etc/ddclient/ddclient.conf. Search (Ctl-W) for ***. Fill in your username and password and uncomment those two lines. Then search for *** again, uncomment the next three lines and fill in your fully-qualified domain name. Save the file and service ddclient restart. To make sure everything worked, issue the following command: ddclient -force. Assuming there are no errors, issue the following command to start ddclient each time your server reboots: /sbin/chkconfig --add ddclient. Now the IP address of your Asterisk server will always resolve to your FQDN from DynDNS. And anyone can call you via SIP for free using the following SIP URI: mothership@yourFQDN.dyndns.org. You can take this a step further and sign up for a free incoming phone number at ipkall.com. For your account type, choose SIP. For your SIP phone number, enter: mothership. For your SIP proxy, enter the fully-qualified domain name (FQDN) for your server, e.g. mypbx.dyndns.org. Choose a password and enter your real email address, and they will beam you a Washington state phone number within a day or so. You can't beat the price!

Getting Phones to Work Reliably. If you or the the person at the other end of your calls only hears half the conversation or if your calls get abruptly disconnected after a few minutes, it's probably because you forgot to add IP addresses to tell SIP how to communicate with your Asterisk server sitting behind a firewall. Edit /etc/asterisk/sip_custom.conf and add an entry for your external IP address and also for your local (internal) subnet where Asterisk resides. Then restart Asterisk: amportal restart.

externip=68.28.142.83
localnet=192.168.0.0/255.255.255.0

If you have a dynamic IP address and you set up ddclient above with your fully-qualified domain name, we've created a little script to keep these entries up to date automatically. Just edit the following file:

/var/lib/asterisk/agi-bin/ip.sh

Fill in the correct entries for your fqdn and localnet. Then uncomment the last line in /etc/crontab which runs ip.sh once every 5 minutes.

Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone (highly recommended). Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack of cigs) known as an SPA-2102. It's under $70. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the device into your LAN, and then plug your phone instrument into the SPA-2102. Note that this adapter supports two-line cordless phones! Your router will hand out a private IP address for the SPA-2102 to talk on your network. You'll need the IP address of the SPA-2102 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The device will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.

Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab and then repeat this drill for the Line2 tab if you want to connect the device to two extensions on your Asterisk system. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter your actual password for this extension in the Password field, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Now repeat the drill for Line2 using extension 702. Pick up a phone and dial 1234# to test out BOTH extensions.

Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with FreePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best phone out there is the Aastra 57i for under $200. Another $100 buys you the Aastra 57i CT with a cordless DECT phone.


Configuring Aastra 57i SIP Phones. Your new system comes preconfigured to automatically configure up to 15 Aastra 57i phones. Plug each phone into your network and wait for it to boot. Once it boots, press the Option button, then Phone Status (3), then IP & MAC Address (1). Write down each phone's IP address and MAC address. Then press Done to exit from the menus.

Next, we need to tell your phone to use your new Asterisk server as the TFTP server to obtain its setup. Press the Option button again, then Admin Menu (5). Type 22222 for the admin password and press Enter. Then choose Config Server (1), then TFTP Settings (2), then Primary TFTP (1), enter the IP address of your new server, and press Done a half dozen times.

Log back into your server as root. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone AND rename each file (filenames are 701.cfg to 715.cfg) to the MAC address of each phone.cfg. Do NOT use hyphens or colons in the MAC address. Edit each of the .cfg files and replace the SIP line1 password with the new password you created for the extension using FreePBX. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!

sed -i 's|192.168.0.0|192.168.0.123|g' /tftpboot/aastra.cfg

Now restart each phone by pressing the Option button and then Restart Phone (6) and then the Restart button. Once the phone reboots, you can make a test call by dialing 1-2-3-4. You can get the latest news by dialing 5-1-1. Or get a weather forecast by airport code (6-1-1) or zip code (Z-I-P).

A Word About Ports. For the techies out there that want to configure remote telephones or link to a server in another town, you'll need to know the ports to remap to your new server from your firewall. Here's a list of the ports available and used by PBX in a Flash. We don't recommend exposing UDP 5038 which is used to communicate with Asterisk via the Asterisk Manager.

TCP 80 - HTTP (needed to access the web sites on your server from the Net)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access... not recommended!!!)
UDP 10000-62000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX connection between Asterisk servers)

Setting Up Trunks for Outgoing and Incoming Calls. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. And you'll need a phone number (DIDs) so that folks can call you. Unlike the Ma Bell world, you need not rely upon the same provider for both. And nothing prevents you from having multiple outbound and incoming trunks to your new PBX. At a minimum, however, you do need one outbound trunk and one inbound phone number unless you're merely planning to talk to other extensions set up on your system. We've actually put all the hooks in place to make it easy for you to interconnect to other Asterisk servers, but we'll save that for another day. For today, we want to get you a functioning system so that you can place outbound calls to anywhere in the world and can receive incoming calls from anywhere in the world.

For outbound calling, we recommend you establish accounts with several providers. We've included the necessary setups for Joe Roper's new service for PBX in a Flash as well as Vitelity and AOL. To register for the service, just visit the web site and register. To sign up to the service in the USA and be charged in US Dollars, please sign up here. To sign up for the European Service and be charged in Euros, sign up here.

In addition to being one of the least expensive providers, there's also the premium service option. You can prefix any number with 000 to try it out. Give it a try. We think you'll be pleased with the service AND the pricing. DIDs for inbound service are not yet available, but Vitelity has lots of them, and there's a link below to get you started.

Vitelity: One of the Best Providers on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus reasonable entry level pricing plus high quality calls, then Vitelity is a winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. For PBX in a Flash users, sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price (except with us) and the call quality is excellent as well. We've tried just about everybody.

To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes. The same setup drill will get you going the the PIAF VoIP service as well.

To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good SIP service can be, pick up a phone on your system and dial D-E-M-O. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications.

An Alternate Outbound Calling Solution. As we said, it costs you almost nothing to add an alternate outbound calling solution to your new system. As luck would have it, adding a third outbound calling provider is now a breeze because AOL just entered the SIP terminations market with a product called AIM Call Out. We wrote about it recently, and you can read the article here. All you need is an AOL or AIM account name and $5 to get you started. The system you've just installed is preconfigured to use AIM Call Out. All you have to do is plug in your username and password, and you can immediately make calls to anywhere in the United States for under 2¢ per minute. Adding international calling is as easy as inserting the correct dial string. If you never use it, it doesn't cost you a dime. So $5 is mighty cheap insurance in our book.

First things first. Sign up for the service at this link. Your username will look something like this: johndoe@aim.com. You also will be assigned a password. Using your web browser, open FreePBX by pointing to the IP address of your new server and choosing Administration, then FreePBX. Type in admin as your username and the password you assigned to your system. From the main FreePBX menu, choose Setup, Trunks, and click on SIP/AIM in the far right column. Scroll down to the Peer Details section of the form and replace yourAIMpassword with your new password. Then replace yourAIMaccountname with your actual AIM account name. Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Setting Up an Alternate DID for Incoming Calls. You also may want to consider a second phone number where people can call you. For example, if Grandma and Grandpa happen to be in another state and still have an old fashioned telephone, you might consider adding an additional DID to your system in their area code. They then can make a local call to reach you by dialing the local DID. On the les.net pay-as-you-go plan, it costs less than a dollar a month plus a penny a minute for the calls. Money well spent if we do say so... and you'll sleep better.

If this setup looks a bit complicated, don't be intimidated. Remember, we're connecting your PBX to the rest of the world so people can call you! With les.net, you have a choice of rate plans for most DIDs. You either can pay $3.99 a month for unlimited inbound calls with two concurrent channels or 99¢ per month and 1.1¢ per minute with four concurrent channels. Just visit their site and click Signup to register. Once you are registered, click Login and then Order DIDs. Pick a phone number. Then click Peers/Trunks and Create New Peer. Write down the Peer Name as you will need it in a minute to set up your connection. Choose SIP for Peer Technology, RFC2833 for DTMF Mode, G.711 for Codecs, Registration for Peer Type, enter the public IP address of your server for Peer Address, make up a secure password and write it down also, specify an Outbound CallerID for your calls, and check the 10-digit dialing box. Leave voicemail unchecked since you'll handle this on your end. Save your changes.

Now choose Your DIDs and click on the one you just ordered. We now need to tie the phone number to the Peer setup you just created above. Click on the DID and select the Route to Peer which you just created. Check the Send DID Prefix box and leave everything else blank. Click Save Changes and you're finished at the les.net end. Now let's set up your inbound DID trunk in Asterisk using FreePBX.

Log into FreePBX using a web browser. Click Setup, Trunks and then Add SIP Trunk. Fill in the CallerID and then drop down to the Outgoing Settings section of the form. For Trunk Name, use the Peer Name that you created above and wrote down. It ought to look something like this: 1092832198. For Peer Details, enter the following using the Peer Name and Password you assigned at les.net:

canreinvite=no
context=from-trunk
fromuser=1092832198
host=did.voip.les.net
insecure=port,invite
nat=yes
secret=yourpassword
type=peer
username=1092832198

For Incoming Settings, use from-pstn for the User Context and enter the following User Details:

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=port,invite
nat=yes
type=user

For the registration string, enter a string like the following using your Peer Name and Password:

1092832198:yourpassword@did.voip.les.net/1092832198

Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Choosing a VoIP Provider That Supports Faxing. We've included a reliable fax solution in this build. You can review the details in this Nerd Vittles article. To test your machine, you can connect a real fax machine to one of the extensions using an SPA-2102. Then send a fax to extension 329 (F-A-X). But first you must configure your email address in two places using FreePBX: Setup, General Settings, Email address to have faxes emailed to AND Setup, Inbound Routes, any DID / any CID, fax Email. Once you've saved your settings, send the fax and see if it's delivered to your email address. If it works reliably, then the fax and email applications on your machine are configured correctly. Unfortunately, that's only half the battle. To receive faxes from outside your system, you'll also need a DID from a provider that supports faxing. And then it's still only about a 90% proposition... on a good day. We've tested this with many, many VoIP providers. Some work. Many don't. Some, such as Vitelity, offer a faxing service for a fee. Guess what? Their regular VoIP setup doesn't support faxing. Our old friends at Telasip.com still support faxing. We've also had good luck with Future-Nine and Teliax. You can read our fax dissertation here for more details. With the exception of the trunk setup covered in the article, all of the remaining setup steps already have been completed on your new server!

Interconnecting Two Asterisk Servers. We've preconfigured this build to support an IAX interconnect to a second PBX in a Flash system. The trunk setup for the second machine to match the setup on this build can be printed out. The filename is /root/MainPeerTrunkSetup.gif.

Choosing a Preferred Provider. Finally, you'll need to decide whether to use PIAF-USA or AOL or Vitelity as your primary terminations provider. HINT: Joe's new service is the cheapest! So we've set things up this way. This is handled in FreePBX in the Outbound Routes tab under the Default entry. You can adjust easily these in any way you like by adding trunks or moving entries up and down the list to change their priority. Just be sure to leave ENUM at the top of the list since ENUM calls are always free. If a free call isn't possible, your server will automatically drop down to the next trunk in the priority list. Don't add Vitelity to the list unless you have actually created a Vitelity account since they handle unsuccessful connections in a non-standard way which will cause FreePBX not to drop down to the next trunk to attempt a connection.

Activating the Stealth AutoAttendant for Inbound Calls. By default, all incoming calls are routed to the Day/Night Code 1 context which allows you to toggle calls between a Day setting and a Night setting by pressing *281. The Day setting for Code 1 is set to our Stealth Autoattendant which plays a brief greeting during which you can choose other options or direct dial extensions on your system before the call is passed to Ring Group 700. To change the options, edit MainIVR.

Activating Mondo Backups. We would be remiss if we didn't mention what a fantastic open source product Mondo Rescue is. It's the sole reason that today's build was possible. Our special thanks go to the development team: Bruno Cornec, Andree Leidenfrost, and Hugo Rabson. It is the first (and only) backup software for Linux builds that actually works reliably. The best way to prove that for yourself is to download the Orgasmatron III and try it for yourself. It has much more flexibility than what you will experience, but that would take another dozen pages to explain. We'll save that for another day. In the meantime, if you'd like more information, visit the Mondo Rescue web site.

WARNINGS: If you update the version of Mondo shipped with this distribution to the current version using either yum or a standalone RPM, you will break your backup system. The advantage of the newer version is that it can create bootable flash drives with your backup image. The disadvantage is that the restore process croaks and locks up your machine. So don't update for the time being. We'll let you know when it's safe to upgrade.

Particularly if you have more than one drive in your system, be aware that the device name for your USB flash drive may differ from the setting of /dev/sdb1 that is preconfigured in this backup. This depends upon the number of internal hard disks and the Dude that built your Dell.

To safely activate backups on a stock T100 configured as we've outlined above, here are the mandatory steps:

1. Format every USB stick you plan to use for backups. Insert the USB flash drive into the right USB slot on the front of your Dell T100. Log into your server as root and type: /root/usbformat.sh. Your USB flash drive is now formatted. Repeat the process for any additional USB flash drives. WARNING: Do not use this script if you have added additional drives on your system as it may inadvertently reformat the wrong drive! The script assumes you have one or two internal SATA drives and one USB stick inserted in the right USB slot on the front of your Dell T100.

2. Assign the proper device name to Mondo and activate it: With a formatted USB flash drive in place, log into your server as root and type: /root/usbdevice.sh. You're all set. A backup will be made each Sunday night. If no flash drive is present, the backup will be saved in /etc/usbmondo.

3. Run a test backup: With a USB flash drive in place, log in as root, and type: /etc/cron.weekly/disk-backup.cron. To be sure it worked, see #4.

4. Check the contents of your USB stick regularly! Plug it into the front right USB port, log in as root, and type usbcheck. It's a good practice to check this on Mondays to be sure you got a fresh backup on Sunday night!!

Other Backup Options. Of course, there are some other backup options. FreePBX is preconfigured to make an automatic backup of your FreePBX data once a week. This is controlled by the settings in Tools, Backup and Restore, WeeklyBackup. It currently is set to make a backup every Wednesday morning. You also may want to consider off-site backups. Amazon's S3 service is preconfigured including all necessary software and scripts. All you need is an account and password. For detailed instructions, see this Nerd Vittles' article.

Installing Cepstral on Your New Server. If you want real text-to-speech with Allison's familiar voice, then you'll need to buy Cepstral. It's dirt cheap for single, non-commercial use. To install it, run install-cepstral from the command prompt while logged in as root. At one point you'll be asked whether to create a missing directory for the Cepstral installation. Be sure to type y at the prompt rather than just pressing the Enter key. Instructions for registering your copy of Cepstral are displayed when the install completes. For complete documentation, read our previous tutorial.

Creating Your Own Hamachi VPN Network. We've saved the best for last today. This latest Orgasmatron III build includes the Hamachi VPN network software. All you have to do is initialize it. Once configured, you can add as many as 16 computers (including Windows, Mac, and Linux machines) to your own private virtual private network. Communications between all of your systems then will be encrypted by simply connecting to the other systems using their VPN network addresses (5.x.x.x). For complete setup instructions, take a look at our VPN in a Flash knol on Google. The entire setup takes less than 5 minutes.

News Flash: As we put this article to bed last night, we tried one final experiment. We took the bootable USB flash drive from our VPN in a Flash build for the Aspire One NetBook that was featured last week and plugged it into the Dell T100. Guess what, Dude? Twelve minutes later we had a perfect clone of the Aspire One build on our new Dell T100. So, if you're looking for a state-of-the-art operating system with a fantastic GUI interface to pair up with Asterisk and PBX in a Flash, we may have another surprise for you to ring in the new year with your new T100. And it should work splendidly on the older SC440 as well as other machines with any industry-standard SATA drive. For 2009, PBX in a Flash perhaps should be renamed PBX on a Flash. Imagine carrying a full-featured, preconfigured PBX around on your keychain. Now that should impress even your nerdiest friends. There still are a few kinks with the latest version of Mondo which have forced us to build our own custom patches to get a successful restore, but we're oh so close... Stay tuned!


Special Thanks. As another year comes to a close, we want to take a moment to thank all of you for reading Nerd Vittles. About 50,000 folks from 137 countries around the globe read Nerd Vittles every week. The Nerd Vittles Official Flag above shows all of your home towns. Incidentally, the countries are ordered by the number of actual visitors from each country.

Where To Go From Here. We've covered a good bit of territory today. When you're ready, move on to the second part of this article at the link below. In the meantime, you have a new phone system that works. And there are a number of PDF documents in the /root folder on your new system which are worth a read. Better yet, you can browse through all of the documentation which is available for PBX in a Flash by going here. You also can dial D-E-M-O on your new system and see just how powerful direct SIP connections can be to other Asterisk hosts (in this case, ours!)... at no cost. Finally, you can log into your server and type help-pbx for access to a treasure trove of additional features. Enjoy and have a Merry Christmas!

Continue reading Part II...


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 


Some Recent Nerd Vittles Articles of Interest...

It’s a Dell With Asterisk, Dude: Introducing the Orgasmatron II for the Dirt-Cheap Dell SC440

About 20 years ago, we began our migration from proprietary DEC VAX minicomputers by acquiring two of the first Dell servers ever manufactured. They were serial numbers 6 and 7. And we were just about as excited about the transition as the folks on Sesame Street. The machines were quickly named Bert and Ernie, and there was even a scrolling LCD display on the units showing the machine names. Considering that this all occurred inside a federal courthouse, it was revolutionary at the time. Michael Dell has gotten a little richer since that initial $10,000 investment (a bargain at the time!), and I'd have to say Dell servers have improved a good bit as well. So we finally bit the bullet last week and bought two of Dell's SC440 servers when they went on sale for a whopping $199 each. For this price, you got Dual Core Intel® Pentium®E2180, 2.0GHz processors with 1MB Cache, an 800MHz FSB, an 80GB 7.2K RPM Serial ATA 3Gbps 3.5-in Cabled Hard Drive connected to the onboard SATA controller, 512MB of 667MHz DDR2 RAM, a 48X CD-ROM Drive, and an On-Board Single Gigabit Network Adapter. For $19 more, you got 2GB of RAM. We hope some of you also took advantage of the offer because today we're releasing our plug-and-play Osgasmatron II for the SC440. Nov. 13 NEWS FLASH: Dell once again is offering the SC440 with Dual-Core processor but it now includes a second hard disk for $299 until November 19. It's still a great deal on a top-notch server. To get email alerts when the SC440 again goes on sale, go to techbargains.com and search for SC440. Then click on Send Email Deal Alert. Be sure to confirm the alert by replying to the email. These units have gone on sale roughly every two weeks since early September.

If you've missed the last two week's articles, the Orgasmatron II is the Ultimate Kitchen Sink for Asterisk®. It includes PBX in a Flash 1.3 in all its glory plus the newly released FreePBX 2.5 and so much more. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... 15 minutes!

If you've been following along with our articles, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i. But our previously anointed perfect small business/home computer on which to run a production Asterisk server for about 50 employees, the Everex gPC2 (aka "The WalMart Special"), is no more. So this build moves to a different platform and a very different performance level. You'll see about twice the performance on the SC440 compared to the WalMart Special. Today's build provides a preconfigured SC440 installation on a 2-disk ISO image backup of the whole system using Mondo. And, NO, it won't work with any other hardware! Once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own SC440. Wait to install any add-on cards until after you complete the Orgasmatron install. You must have an SC440 configured as above, or this Mondo restore will not work. So accept no substitutes, or you may end up with an Electronic Brick instead of an Orgasmatron II.

We've preconfigured some extensions on your new system as well as outbound and incoming trunks from some terrific providers including our second homegrown entry for VoIP terminations. Joe Roper and his business partner in Spain now offer a terrific IAX VoIP termination service. You can choose penny a minute service in the U.S. and most of Canada, or you can opt for premium VoIP service at about 2¢ a minute in the U.S. International rates also are VERY reasonable! You literally can sign up for service, plug in your phones, and have a system in full operation in under an hour.

So... what do you get with this preconfigured build? In addition to all of the goodness of a stock PBX in a Flash 1.3 build including Asterisk 1.4.21.2 running under CentOS 5.2 with a version of Zaptel that actually works with legacy cards. You also get the brand new FreePBX 2.5 as well as the latest versions of Apache, MySQL, PHP, and SendMail. And you get a Baker's Dozen preconfigured Nerd Vittles applications. Complete documentation is available here.

  • Inbound and Outbound VoIP Faxing Using nvFax... finally!
  • FONmail for Asterisk to send voice messages to any email address on the planet
  • AsteriDex RoboDialer and Telephone Directory
  • Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
  • NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
  • Weather Reports by Airport Code (TTS)
  • Weather Reports by ZIP Code (TTS)
  • Worldwide Weather Forecasts (TTS)
  • xTide for Asterisk (TTS)
  • MailCall for Asterisk: Get Your Email By Telephone (TTS)
  • TeleYapper 4.0 Message Broadcasting System
  • CallWho for Phone Lookup and Dialing of Entries in the AsteriDex Database (TTS)
  • TFTP Server with preconfigured setups for 10 Aastra 57i SIP telephones

In addition, you get dozens of preconfigured telephony applications and functions that would take even an expert the better part of a year or two to build independently. And, unlike all of the other distributions, we build Asterisk from source so it's simple to modify and upgrade whenever you feel the need. Here's a short list of what you have to look forward to:

  • Stealth AutoAttendant with Welcome and Application IVRs
  • Key Telephone Support Using Park and Parking Lot
  • Intercom/Paging Support
  • Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
  • DISA
  • Blacklisting with Web and Telephony Interfaces
  • CallerID Name Lookups from Numerous Providers
  • Weekly Automated System Backups to a Flash Drive
  • One Touch Day/Night Service
  • Music on Hold
  • Voicemail with Email Delivery of Messages and Pager Notification
  • Voicemail Blasting
  • Cell Phone Direct Dial
  • Call Forward: All, Busy, No Answer
  • Call Waiting
  • Call Pickup
  • Zap Barge
  • Call Transfer: Attended and Blind
  • Dictation Service with Email Delivery
  • Do Not Disturb
  • Gabcast
  • Phonebook Dial by Name
  • Speed Dial
  • Flite Text to Speech (TTS)
  • Windows Networking with SAMBA
  • Linux Firewall and Fail2Ban with SSH, HTTP, and SIP/IAX login protection
  • PBX in a Flash Software Update Service To Keep Your System Current
  • One-Click Cepstral TTS Install with Allison... Just Type install-cepstral

Prerequisites. As mentioned, you'll need an SC440 configured with the specs outlined above including the 2GB RAM upgrade. We also recommend a 4GB USB flash drive on which to store automatic weekly backups of your new system. Just plug it into your new machine, log in as root, and type: /root/usbformat.sh. That's it! Every Sunday night, you'll get a new backup in ISO format on your flash drive. If something goes wrong on your system, copy the ISOs to CDs and reboot with Disk 1. It doesn't get any easier than that. And you can always check on the latest backup by issuing the command: /root/usbcheck.sh

Finally, you'll need to cough up a whopping $5 to download the two-disk ISO image for this build. And, yes, we eat our own dog food. The ISO images you'll be downloading were captured as a backup on the flash drive of one of our SC440 lab machines. We got 'em yesterday! If you use this special build, it seemed only fair that you cover the cost of the bandwidth to download it. As most of you know, we don't have the luxury of freeloading off SourceForge for our downloads. And we didn't want to impose upon our existing bandwidth providers to bring you this custom image. The good news is that, once you download the image from DreamHost, you are more than welcome to pass it along to one or more of your friends or business acquaintances at no charge. You can even do it electronically through the DreamHost Files Forever program. And, if you'd like to host this image for your fellow man at no cost, be our guest... and thank you! Bottom line: For under $250, you'll have the slickest, fastest, most reliable PBX and fax machines on the planet with rock-solid weekly backups and, of course, access to the one-of-a-kind PBX in a Flash Software Update Service!

Getting Started. Once you have your SC440 in hand, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. You'll need a USB keyboard for typing temporarily. We also strongly recommend that you always keep your system running behind a NAT-based firewall/router. We strongly recommend the dirt-cheap dLink WBR-2310 WiFi router which handles NAT issues with VoIP masterfully. Don't redirect any ports to the machine and don't turn the PC on just yet.

Download the two ISO images for the SC440 from here. If you don't know how to create a CD from an ISO image, read that section from our previous article. In fact, read the whole article. It'll help you immensely down the road. Once you have the two CDs in hand, turn on the SC440 and quickly insert Disk 1 into the CD drive and close the drive. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names, halt the install by pressing CTL-ALT-DEL and remove the CD. Otherwise, ignore the errors. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 10 minutes. After fileset #87 is restored, you'll be prompted to insert Disk 2 and press Enter to finish the install. When the second CD finishes, eject it and wait for the prompt. Then type "exit" and press Enter. Your SC440 will reboot, and you're ready to go.

After the reboot finishes, type root at the login prompt for your username and password for your password. The IP address assigned by your DHCP server should appear near the top of the screen. Write it down. If there is no IP address, your machine does not have network connectivity or access to a DHCP server with an available IP address. Correct the problem and reboot. You can safely ignore the warning that Fail2Ban is OFFLINE. We've updated the Fail2Ban software to protect Asterisk SIP and IAX connections, and our status program isn't up to date as this article goes to press. update-fixes will get you a new version from our Software Update Service shortly.

Securing Passwords. We're going to change five passwords now. For the time being (until you've done some reading), think up one really difficult password (that you won't forget) and use it for all five passwords. At the root@pbx:~ $ command prompt, type the following commands and type in your new password when prompted. Don't forget your password or you'll get to put in your two CDs and start over.

passwd
passwd-maint
passwd-wwwadmin
passwd-meetme
/usr/libexec/webmin/changepass.pl /etc/webmin root yournewpasswordhere

Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click the Admin tab and then choose the FreePBX Administration botton. Log in as maint with your new maint password. Before you do anything else, change ALL of the 10 extension passwords to something secure... as if your phone bill depended upon it! Click Setup, Extensions and then choose each extension, modify BOTH the device secret and Voicemail Password, and click Submit. When you finish all the extensions, then reload the dialplan to save your changes. Finally, change your DISA password to something very, very secure: Setup, DISA, DISAmain, PIN. Reload your dialplan once again to save your changes.

Regardless of what you may read elsewhere, the Orgasmatron II has all the very latest security patches as of today. If you want more security, take our advice and add a hardware-based firewall/router between your Internet connection and your new Orgasmatron II and don't expose port 80 (the web interface) to the Internet!

Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set. Either way, you need a permanent IP address for your machine when all is said and done. Once you have a permanent IP address, hop on over to dyndns.org and sign up for your own fully-qualified domain name (FQDN), e.g. mypbx.dyndns.org. You're going to need it for a whole host of things with your new PBX, and dyndns.org is about the easiest way to do it. Once you have your FQDN and DynDNS username and password, log in as root and edit: /etc/ddclient/ddclient.conf. Search (Ctl-W) for ***. Fill in your username and password and uncomment those two lines. Then search for *** again, uncomment the next three lines and fill in your fully-qualified domain name. Save the file and service ddclient restart. To make sure everything worked, issue the following command: ddclient -force. Assuming there are no errors, issue the following command to start ddclient each time your server reboots: /sbin/chkconfig --add ddclient. Now the IP address of your Asterisk server will always resolve to your FQDN from DynDNS. And anyone can call you via SIP for free using the following SIP URI: mothership@yourFQDN.dyndns.org. You can take this a step further and sign up for a free incoming phone number at ipkall.com. For your account type, choose SIP. For your SIP phone number, enter: mothership. For your SIP proxy, enter the fully-qualified domain name (FQDN) for your server, e.g. mypbx.dyndns.org. Choose a password and enter your real email address, and they will beam you a Washington state phone number within a day or so. You can't beat the price!

Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone (highly recommended). Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack of cigs) known as an SPA-2102. It's under $70. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the device into your LAN, and then plug your phone instrument into the SPA-2102. Note that this adapter supports two-line cordless phones! Your router will hand out a private IP address for the SPA-2102 to talk on your network. You'll need the IP address of the SPA-2102 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The device will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.

Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab and then repeat this drill for the Line2 tab if you want to connect the device to two extensions on your Asterisk system. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter your actual password for this extension in the Password field, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Now repeat the drill for Line2 using extension 702. Pick up a phone and dial 1234# to test out BOTH extensions.

Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with FreePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best phone out there is the Aastra 57i for under $200. Another $100 buys you the Aastra 57i CT with a cordless DECT phone.

Configuring Aastra 57i SIP Phones. Your new system comes preconfigured to automatically configure up to 15 Aastra 57i phones. Plug each phone into your network and wait for it to boot. Once it boots, press the Option button, then Phone Status (3), then IP & MAC Address (1). Write down each phone's IP address and MAC address. Then press Done to exit from the menus.

Next, we need to tell your phone to use your new Asterisk server as the TFTP server to obtain its setup. Press the Option button again, then Admin Menu (5). Type 22222 for the admin password and press Enter. Then choose Config Server (1), then TFTP Settings (2), then Primary TFTP (1), enter the IP address of your new server, and press Done a half dozen times.

Log back into your server as root. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone AND rename each file (filenames are 701.cfg to 715.cfg) to the MAC address of each phone.cfg. Do NOT use hyphens in the MAC address. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!

sed -i 's|192.168.0.0|192.168.0.123|g' /tftpboot/aastra.cfg

Now restart each phone by pressing the Option button and then Restart Phone (6) and then the Restart button. Once the phone reboots, you can make a test call by dialing 1-2-3-4. You can get the latest news by dialing 5-1-1. Or get a weather forecast by airport code (6-1-1) or zip code (Z-I-P).

A Word About Ports. For the techies out there that want to configure remote telephones or link to a server in another town, you'll need to know the ports to remap to your new server from your firewall. Here's a list of the ports available and used by PBX in a Flash. We don't recommend exposing UDP 5038 which is used to communicate with Asterisk via the Asterisk Manager.

TCP 80 - HTTP (needed to access the web sites on your server from the Internet... not recommended!!!)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access... not recommended!!!)
UDP 10000-62000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX communications typically between Asterisk servers)

Setting Up Trunks for Outgoing and Incoming Calls. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. And you'll need a phone number (DIDs) so that folks can call you. Unlike the Ma Bell world, you need not rely upon the same provider for both. And nothing prevents you from having multiple outbound and incoming trunks to your new PBX. At a minimum, however, you do need one outbound trunk and one inbound phone number unless you're merely planning to talk to other extensions set up on your system. We've actually put all the hooks in place to make it easy for you to interconnect to other Asterisk servers, but we'll save that for another day. For today, we want to get you a functioning system so that you can place outbound calls to anywhere in the world and can receive incoming calls from anywhere in the world.

For outbound calling, we recommend you establish accounts with several providers. We've included the necessary setups for Joe Roper's new service for PBX in a Flash as well as Vitelity and AOL. To register for the service, just visit the web site and register. To sign up to the service in the USA and be charged in US Dollars, please sign up here. To sign up for the European Service and be charged in Euros, sign up here.

In addition to being one of the least expensive providers, there's also the premium service option. You can prefix any number with 000 to try it out. Give it a try. We think you'll be pleased with the service AND the pricing. DIDs for inbound service are not yet available, but Vitelity has lots of them, and there's a link below to get you started.

Vitelity: One of the Best Providers on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus reasonable entry level pricing plus high quality calls, then Vitelity is a winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. For PBX in a Flash users, sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price (except with us) and the call quality is excellent as well. We've tried just about everybody.

To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes. The same setup drill will get you going the the PIAF VoIP service as well.

To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good SIP service can be, pick up a phone on your system and dial D-E-M-O. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications.

An Alternate Outbound Calling Solution. As we said, it costs you almost nothing to add an alternate outbound calling solution to your new system. As luck would have it, adding a third outbound calling provider is now a breeze because AOL just entered the SIP terminations market with a product called AIM Call Out. We wrote about it recently, and you can read the article here. All you need is an AOL or AIM account name and $5 to get you started. The system you've just installed is preconfigured to use AIM Call Out. All you have to do is plug in your username and password, and you can immediately make calls to anywhere in the United States for under 2¢ per minute. Adding international calling is as easy as inserting the correct dial string. If you never use it, it doesn't cost you a dime. So $5 is mighty cheap insurance in our book.

First things first. Sign up for the service at this link. Your username will look something like this: johndoe@aim.com. You also will be assigned a password. Using your web browser, open FreePBX by pointing to the IP address of your new server and choosing Administration, then FreePBX. Type in admin as your username and the password you assigned to your system. From the main FreePBX menu, choose Setup, Trunks, and click on SIP/AIM in the far right column. Scroll down to the Peer Details section of the form and replace yourAIMpassword with your new password. Then replace yourAIMaccountname with your actual AIM account name. Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Setting Up an Alternate DID for Incoming Calls. You also may want to consider a second phone number where people can call you. For example, if Grandma and Grandpa happen to be in another state and still have an old fashioned telephone, you might consider adding an additional DID to your system in their area code. They then can make a local call to reach you by dialing the local DID. On the les.net pay-as-you-go plan, it costs less than a dollar a month plus a penny a minute for the calls. Money well spent if we do say so... and you'll sleep better.

If this setup looks a bit complicated, don't be intimidated. Remember, we're connecting your PBX to the rest of the world so people can call you! With les.net, you have a choice of rate plans for most DIDs. You either can pay $3.99 a month for unlimited inbound calls with two concurrent channels or 99¢ per month and 1.1¢ per minute with four concurrent channels. Just visit their site and click Signup to register. Once you are registered, click Login and then Order DIDs. Pick a phone number. Then click Peers/Trunks and Create New Peer. Write down the Peer Name as you will need it in a minute to set up your connection. Choose SIP for Peer Technology, RFC2833 for DTMF Mode, G.711 for Codecs, Registration for Peer Type, enter the public IP address of your server for Peer Address, make up a secure password and write it down also, specify an Outbound CallerID for your calls, and check the 10-digit dialing box. Leave voicemail unchecked since you'll handle this on your end. Save your changes.

Now choose Your DIDs and click on the one you just ordered. We now need to tie the phone number to the Peer setup you just created above. Click on the DID and select the Route to Peer which you just created. Check the Send DID Prefix box and leave everything else blank. Click Save Changes and you're finished at the les.net end. Now let's set up your inbound DID trunk in Asterisk using FreePBX.

Log into FreePBX using a web browser. Click Setup, Trunks and then Add SIP Trunk. Fill in the CallerID and then drop down to the Outgoing Settings section of the form. For Trunk Name, use the Peer Name that you created above and wrote down. It ought to look something like this: 1092832198. For Peer Details, enter the following using the Peer Name and Password you assigned at les.net:

canreinvite=no
context=from-trunk
fromuser=1092832198
host=did.voip.les.net
insecure=port,invite
nat=yes
secret=yourpassword
type=peer
username=1092832198

For Incoming Settings, use from-pstn for the User Context and enter the following User Details:

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=port,invite
nat=yes
type=user

For the registration string, enter a string like the following using your Peer Name and Password:

1092832198:yourpassword@did.voip.les.net/1092832198

Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Choosing a VoIP Provider That Supports Faxing. We've included a reliable fax solution in this build, and we'll cover all the details soon. We do want to give you a head start if you plan to use your new machine to handle inbound faxes. To test your machine, you can connect a real fax machine to one of the lines on an SPA-2102. Then send a fax to extension 329 (F-A-X). But first you must configure your email address in two places using FreePBX: Setup, General Settings, Email address to have faxes emailed to AND Setup, Inbound Routes, any DID / any CID, fax Email. Once you've saved your settings, send the fax and see if it's delivered to your email address. If it works reliably, then the fax and email applications on your machine are configured correctly. Unfortunately, that's only half the battle. To receive faxes from outside your system, you'll also need a DID from a provider that supports faxing. And then it's still only about a 90% proposition... on a good day. We've tested this with many, many VoIP providers. Some work. Many don't. Some, such as Vitelity, offer a faxing service for a fee. Guess what? Their regular VoIP setup doesn't support faxing. Our old friends at Telasip.com still support faxing. We've also had good luck with Future-Nine and Teliax. You can read the beginnings of our fax dissertation here for more details. With the exception of the trunk setup covered in the article, all of the remaining setup steps already have been completed on your new server!

Interconnecting Two Asterisk Servers. We've preconfigured this build to support an IAX interconnect to a second PBX in a Flash system. The trunk setup for the second machine to match the setup on this build can be printed out. The filename is /root/MainPeerTrunkSetup.gif.

Choosing a Preferred Provider. Finally, you'll need to decide whether to use PIAF-USA or AOL or Vitelity as your primary terminations provider. HINT: We're the cheapest! So we've set things up this way. This is handled in FreePBX in the Outbound Routes tab under the Default entry. You can adjust easily these in any way you like by adding trunks or moving entries up and down the list to change their priority. Just be sure to leave ENUM at the top of the list since ENUM calls are always free. If a free call isn't possible, your server will automatically drop down to the next trunk in the priority list. Don't add Vitelity to the list unless you have actually created a Vitelity account since they handle unsuccessful connections in a non-standard way which will cause FreePBX not to drop down to the next trunk to attempt a connection.

Activating the Stealth AutoAttendant for Inbound Calls. By default, all incoming calls are routed to the Day/Night Code 1 context which allows you to toggle calls between a Day setting and a Night setting by pressing *281. For builds before Rev. D, the Day setting for Code 1 is set to Ring Group 700 which rings all of the extensions on your system. If you'd prefer our Stealth Autoattendant which plays a brief greeting during which you can choose other options or direct dial extensions on your system before the call is passed to Ring Group 700, then edit Day/Night Code 1 and set the Day option to IVR: MainIVR.

A Word About Mondo Rescue. We would be remiss if we didn't mention what a fantastic open source product Mondo Rescue is. It's the sole reason that today's build was possible. Our special thanks go to the development team: Bruno Cornec, Andree Leidenfrost, and Hugo Rabson. It is the first (and only) backup software for Linux builds that actually works reliably. The best way to prove that for yourself is to download this build and try it for yourself on your Dell, dude. It has much more flexibility than what you will experience, but that would take another dozen pages to explain. We'll save that for another day. In the meantime, if you'd like more information, visit the Mondo Rescue web site.

What you need to know today is that the device name for your USB flash drive may differ from the setting of /dev/sdb1 that is preconfigured depending upon the Dude that built your Dell. If you have the Rev. D build (shown at the bottom of the DreamHost download site), simply log into your server as root and type: /root/usbdevice.sh. You're all set. With prior builds, to find out the identity of your USB stick, plug it into one of the front USB ports, log in as root, and type dmesg. Included in the output will be a section that looks something like this:

USB Mass Storage support registered.
Vendor: VBTM Model: Store 'n' Go Rev: 5.00
Type: Direct-Access ANSI SCSI revision: 00
SCSI device sdb: 2013184 512-byte hdwr sectors (1031 MB)
sdb: Write Protect is off
sdb: Mode Sense: 23 00 00 00
sdb: assuming drive cache: write through
SCSI device sdb: 2013184 512-byte hdwr sectors (1031 MB)
sdb: Write Protect is off
sdb: Mode Sense: 23 00 00 00
sdb: assuming drive cache: write through
sdb: sdb4

If the entry in bold above does not say "sdb1" then you have a little work to do. First, edit /root/usbcheck.sh and change sdb1 on the mount line to the sdb# entry shown above in bold. Save your change: Ctrl-X, Y, then Enter. Now edit /root/usbformat.sh and make the same change in the fdisk AND mkdosfs lines of the script. Save your changes. Finally edit /etc/asterisk/disk-backup.conf. Press Ctl-W and search for sdb1. Change the entry to the device name in bold above. Save your change. Now restart Asterisk with the command: amportal restart. Finally, format your new flash drive and you're ready to go: /root/usbformat.sh. Be sure to check your flash drive periodically to make certain you're getting backups: /root/usbcheck.sh.

Installing Cepstral on Your New Server. If you want real text-to-speech with Allison's familiar voice, then you'll need to buy Cepstral. It's dirt cheap for single, non-commercial use. To install it, there's still a problem with the script on your new machine unfortunately. Something has happened to Darren Sessions' archives, but luckily we still have backups. This has been fixed in Rev. D. Otherwise, to point to the uncorrupted version of the software, log in to your server as root and issue the following two commands:

sed -i 's|www.darrensessions.com/pub|pbxinaflash.net/source|' /root/install-cepstral
sed -i 's|www.darrensessions.com/pub|pbxinaflash.net/source|' /usr/local/sbin/install-cepstral

Then run install-cepstral from the command prompt. At one point you'll be asked whether to create a missing directory for the Cepstral installation. Be sure to type y at the prompt rather than just pressing the Enter key. Instructions for registering your copy of Cepstral are displayed when the install completes. For complete documentation, read our previous tutorial.

Addendum: Enabling Parking Lot. As configured, FreePBX gets confused by the 700 ring group thinking it is your default parking lot. To fix the problem, simply enable the Parking Lot feature from the FreePBX Setup tab. Click on the Enable checkbox, leave the default 70 extension to place calls in the parking lot, and choose a default location to which to send orphaned parking lot calls. Then everything works normally.

Where To Go From Here. Well, we've covered a good bit of territory today. When you're ready, move on to the second part of this article at the link below. In the meantime, you have a new phone system that works. And there are a number of PDF documents in the /root folder on your new system which are worth a read. Better yet, you can browse through all of the documentation which is available for PBX in a Flash by going here. You also can dial D-E-M-O on your new system and see just how powerful direct SIP connections can be to other Asterisk hosts (in this case, ours!)... at no cost. Finally, you can log into your server and type help-pbx for access to a treasure trove of additional features. Enjoy!

Continue reading Part II...


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 


Some Recent Nerd Vittles Articles of Interest...

Orgasmatron II for Asterisk: A Turnkey PBX Install in Under 15 Minutes, Part II

We began our 15-minute adventure with a turnkey install of Asterisk onto either a $199 Everex gPC2 or a Dell SC440 or T100 using a fully-customized version of PBX in a Flash. If you haven't yet read the first article, start there. In Part II, we want to cover what components are included and walk you through using most of them. When we're finished, you'll have a good idea why PBX in a Flash is not only different but also a quantum leap forward in the turnkey IP telephony marketplace. We'll also cover the new fax addition to this build as well as adding RAID 1 redundant drive support to your new gPC2 server (not the Dell) for about $40.

Putting Your Backup System Into Operation. Hopefully, you heeded our recommendation and purchased a $20 4GB USB flash drive to store backups of your new PBX in a Flash system. Cheap insurance! Now let's put it into production. In the /root folder of your new system, you'll find a PDF with complete documentation for the new Mondo Rescue backup system. If you flip to Appendix A, it will walk you through formatting your new flash drive for use with the backup software. If you'd prefer the easy way, log into your new server as root and type: /root/usbformat.sh. That's it. Your flash drive is now ready to make automatic backups of your entire system every Sunday night. Let's kick off one backup just to be sure everything is working. Log into your server as root and type /etc/cron.weekly/disk-backup.cron. Now go have a cup of coffee. When the command prompt returns in about 30 minutes, type /root/usbcheck.sh to get a listing of the files on your USB flash drive. Now you can sit back and relax knowing that every Sunday night a new full system backup will be loaded onto your flash drive. Should something go horribly wrong with your main drive down the road, it's a simple matter to burn CDs of the ISO backup images and reload everything, the same process you used to build your new system in the first place. Remember, we provided you a Mondo Rescue backup to build your system from ours so you know it works. For us at least, having automatic backups of your data is a critical component in any computer system, particularly your entire telephone system. While Asterisk® aggregations are a dime a dozen these days, no one else has implemented any system backup solution except PBX in a Flash.

Text-to-Speech on Steroids. The next thing you need to do is install Cepstral with Allison on your system. This gives something close to perfect text-to-speech capability for your entire phone system for under $25. And, yes, you can try it out first without spending a dime. Log into your server as root and edit /root/install-cepstral. Delete the current contents and substitute the following code:1

#!/bin/bash
cd /root
wget http://downloads.cepstral.com/cepstral/i386-linux/ ↵
Cepstral_Allison-8kHz_i386-linux_5.1.0.tar.gz
tar -zxvf Cepstral*
cd Cepstral_Allison-8kHz_i386-linux_5.1.0
./install.sh
echo /opt/swift/lib > /etc/ld.so.conf.d/cepstral.conf
ldconfig
cd /usr/src
wget http://pbxinaflash.net/source/app_swift/app_swift-1.4.2.tar.gz
tar -zxvf app_swift-1.4*
rm *.gz
cd app_swift-1.4.2
make
make install
cp swift.conf.sample /etc/asterisk/swift.conf
chown asterisk:asterisk /etc/asterisk/swift.conf
ln -s /opt/swift/bin/swift /usr/bin/swift
sed -i 's|David-8kHz|Allison-8kHz|' /etc/asterisk/swift.conf
amportal restart
asterisk -rx "core show application swift"
echo "Installation completed. "
echo "To purchase a license, go here:"
echo "https://www.cepstral.com/cgi-bin/store/home"
echo "Choose US English, Allison-8kHz, Linux."
echo "To register your installed copy of Cepstral, type: swift --reg-voice"

Now save the script and then run it: /root/install-cepstral. Accept the defaults except create the missing directory when prompted. You're done. That was hard wasn't it. We'll test it out in a few minutes.

PiaF Software Update Service. The PBX in a Flash Software Update Service continues to be a free option on all PBX in a Flash systems until November so, by all means, use it to keep your system current, bug-free, and secure. Log into your server as root and type update-scripts. Once the new scripts are loaded onto your system, type update-fixes. Yes, you can build an Asterisk system from many other ISO distributions. But you won't find another one that can keep your system current and secure without starting all over with a new ISO install. And when you want the latest and greatest version of Asterisk without missing a beat, that's easy, too. Just type update-source and have another cup of coffee while your system is upgraded. And don't forget to run update-fixes one more time to clean up any mess created by the upgrade. NOTE: There's no need to run update-source after installing the Orgasmatron II build. All of the updates already are included in the ISO image you downloaded.

Help at Your Fingertips. And, what if you forget all of these commands down the road and you're too lazy to pull out the documentation? Not to worry! Log into your server as root and type help-pbx.

What's Next? Now that you have a stable, secure, and up-to-date server, let's have some fun. We've loaded and preconfigured most of the Nerd Vittles applications in this build so all you have to do is learn the numbers to dial to use most of the applications. Here's a quick thumbnail sketch for each of the applications:

  • The Ultimate VoIP Fax Machine
    Orgasmatron II now incorporates the original nv-Fax application for sending and receiving faxes using your new Asterisk system. Every incoming call is screened for a fax tone. If it's detected, the fax is received, converted to a PDF document, and emailed to the email address you set up in Part I of this article. You also can convert any document to a fax by simply faxing it to the F-A-X extension on your system. And, when you need to send a fax, just save the document in /tmp with a PDF file extension and the number to which the fax should be routed. Then pick up any phone on your system and dial F-A-X-I-T. Specify the matching destination phone number, and your fax will be on its way. For complete documentation, click on the link above.
  • AsteriDex RoboDialer and Telephone Directory
    This app gets you a phonebook, a web-based dialer using a browser or your cellphone, and a CallerID lookup source when used in conjunction with Ultimate CNAM. To add and update entries or lookup numbers, point your web browser to the IP address of your server: http://ipaddress/asteridex4/. For cell phone access, point the web browser on your cellphone to the public IP address or fully-qualified domain name of your server: http://publicIPaddress/cellphone/. You now can import all of your Microsoft Outlook contacts as well. Just click on the link above for complete documentation and security suggestions.
  • Telephone Reminders 4.0 with Support for Recurring Reminders and Web-based TTS Reminder Messages
    This app lets you schedule reminders for future events by telephone (dial 1-2-3) or with a web browser (http://ipaddress/reminders/). When the appointed date and time arrives, Asterisk swings into action and places a call to the number you designate to deliver a customized reminder message. Recurring reminders (daily, weekday, weekly, monthly, and annual) also are supported. And the text-to-speech web interface lets you schedule and deliver reminders using either Flite or Cepstral-generated messages with any web browser. For more info, click on the link above.
  • NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS) - Dial 5-1-1
  • Weather Reports by Airport Code (TTS) - Dial 6-1-1
  • Weather Reports by ZIP Code (TTS) - Dial Z-I-P
  • Worldwide Weather Forecasts (TTS) - Dial 6-1-2
  • MailCall for Asterisk: Get Your Email By Telephone (TTS)
    This app reads your emails to you over the telephone. Some setup is required to plug in information about your email account. Once configured, dial 5-5-5 to retrieve your messages. Click on the link above for setup instructions.
  • TeleYapper 4.0 Message Broadcasting System - Dial M-S-G (licensed for non-commercial use only!)
  • CallWho for TTS Retrieval and Dialing of Entries in the AsteriDex Database (TTS)
    After entering contacts in AsteriDex, run http://serverIPaddress/asteridex4/dialcode.php to populate the dialcodes. Then dial 4-1-2 and enter the first three letters of anyone in your AsteriDex database to place a call.
  • TFTP Server with preconfigured setups for 15 Aastra 57i SIP telephones - See setup instructions in last week's article

Of course, there are literally hundreds of things you can do with your PBX in addition to running the Nerd Vittles applications. Here's a short list of some of our favorites with some tips to get you started. The best source of information for more detail is our original article on PBX in a Flash 1.3 and the PBX in a Flash Forum.

  • Stealth AutoAttendant with Welcome and Application IVRs
    Whenever an incoming call comes into your PBX, a generic greeting will play. If no button is pressed on the caller's phone, the call then will be routed to a ring group (700) for all of the extensions set up in that ring group. If no one answers, the call will be sent to the voicemail box for extension 701. While the greeting message is playing, the caller can press a digit on their phone to activate a hidden option in the Main IVR. As delivered, the only one that works is 0. This presents the caller with a list of Nerd Vittles apps from which to choose. You can add other options by modifying the Main IVR settings in FreePBX. To try out the Main IVR from any extension on your system, dial 7-7-7.
  • Key Telephone Support Using Park and Parking Lot
    Most PBXs do not support shared line appearances like the old key telephones from Ma Bell. With these phones you could answer a call, place it on hold, and then someone else could pick up the call by pressing the blinking light on their phone. Our Aastra phone setup does much the same thing except, instead of placing a call on hold, you press the Park button. The parked extension number then will be read to you by Allison (starting with 71). Anyone else on your system can retrieve the parked call by pressing the ParkLot button on their Aastra phone and selecting the call to be retrieved by CallerID. Or, if the recipient knows the parking lot extension (e.g. 71), the recipient can pick up any phone and dial that extension number to retrieve the call.
  • Intercom/Paging Support
    The Aastra phone setup for PBX in a Flash fully supports intercom calls and paging by pressing the ICom button on the phone. For more information, click Setup, Paging and Intercom from within the FreePBX web interface.
  • Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
    Your system is preconfigured to support a USB Bluetooth dongle. No additional software installation is required. When properly configured, this lets you automatically forward your calls to your cellphone just by leaving your home or office with your Bluetooth-enabled cellphone. When you return, your calls will magically begin ringing on your local extension again. Click the link above for setup instructions.
  • DISA
    Direct Inward System Access lets you call into your PBX and get dialtone to make an outbound call. To use it, you typically would add it as a hidden option on your IVR with a very secure password. We have preconfigured DISA support on your server. Just be sure you change the password to something very secure before activating it. To change the password, click Setup, DISA, DISAmain in FreePBX. Then save your changes and reload the Asterisk dialplan.
  • Blacklisting with Web and Telephony Interfaces
    To block future calls from the last person who called you, dial *32. To block calls from a specific phone number, dial *30. To remove a number from the blacklist, dial *31. You also can use FreePBX to blacklist certain numbers. Just click Setup, Blacklist to access the web interface.
  • CallerID Name Lookups from 8 Providers
    Most telephony providers reliably pass CallerID numbers but discard CallerID name info. With Ultimate CNAM which is preinstalled on your system, you can look up CallerID names from up to 8 different directory providers. To activate it, use FreePBX and click Setup, Inbound Routes, DefaultIncoming. Scroll down to CID Lookup Source and choose Ultimate CNAM from the dropdown box. Save your changes and reload the dialplan. For complete documentation, consult cnam_user_guide.pdf in the /root folder on your server. To choose the providers to use for the lookups, log into your server as root and type: cnam-config.pl
  • Weekly Automated System Backups to a Flash Drive
    See the first section of today's article for the one-minute setup instructions.
  • One Touch Day/Night Service
    With our Aastra phone setup, there is a DayNite button that toggles your system between Day and Night operation. As configured, the Night option transfers all calls to voicemail for extension 701. The Day option routes all incoming calls through the Main IVR which routes calls to the 700 Ring Group on timeout. To activate Night service from an Aastra phone, just press the DayNite button. To deactivate Night service, just press the button again. You also can dial *28 from any phone on your system to toggle Day/Night mode.
  • Music on Hold
    Royalty-free music on hold is provided as part of the basic Asterisk install. Additional music can be added through the Music on Hold option in FreePBX. WAV files must be PCM Encoded, 16 Bits, at 8000Hz. See this thread for assistance. For other royalty-free and free music on hold, start here, choose Creative Commons for the License Type, and then click Go.
  • Voicemail with Email Delivery of Messages and Pager Notification
    All of these settings are performed within FreePBX for each extension. Choose Setup, Extensions, and pick one of the extensions you already have created. Make certain that Voicemail Status is enabled. Then enter a valid email address and pager address. To include the voicemail as an attachment in the delivered email message, set Email Attachment to Yes. To include the CallerID in the voicemail message, set Play CID to Yes. To include the date and time of the call, set Play Envelope to Yes. To delete the voicemail message from the system after emailing it, set Delete Vmail to Yes. Don't ever do this until you're sure it's working reliably! If you want the option of calling back the caller when you retrieve your voicemail message by phone, set VMoptions to callback=from-internal. Submit your changes and reload the Asterisk dialplan to put the modifications into effect. If the emails are not delivered, then it may be because your ISP is blocking downstream SMTP traffic. To reconfigure your server to use gMail or Comcast as your SMTP host, click on one of the links.
  • Voicemail Blasting
    This feature allows you to record a message and distribute it via voicemail to one or more extensions without actually calling the users. We've already configured extension 500 to send voicemail blasts to extensions 701 and 702. You can adjust the destinations in FreePBX by choosing Setup, Voicemail Blasts, Vmail (500). You also can add additional extensions to handle voicemail blasts to a different group of phones.
  • Cell Phone Direct Dial
    There are two ways to make a cellphone an integral part of your PBX. The first involves setting up a specific extension for each cellphone and forwarding incoming calls to that extension to your cellphone. First, create an extension 501 on your system if it doesn't already exist. Once the extension is created, simply log into your server as root and issue a command like this where 6781234567 is your actual cellphone number:

    asterisk -rx "database put CF 501 6781234567"

    When callers dial 501 on your system, your cellphone will automatically ring. Another option is to use FreePBX's Follow Me function under Setup. With this option, you can specify multiple destinations for incoming calls to a specific extension. Point to the Ring Strategy option and review the available choices. Choose the one that best meets your needs. Then enter the numbers to be called. Numbers outside your PBX should be in the format 6781234567# and must match your outbound dialing rules. You also can choose the time to attempt the call and what to do if no one answers. Very slick!
  • Call Forward: All, Busy, No Answer
    While you can certainly use FreePBX's Follow Me functionality to accomplish any flavor of call forwarding, you also can dial codes from any extensions to activate call forwarding. To activate Call Forwarding All, dial *72; for Call Forwarding Busy, dial *90; for Call Forwarding No Answer, dial *52. To deactivate Call Forwarding All, dial *73; for Call Forwarding Busy, dial *91; for Call Forwarding No Answer, dial *53. To deactivate Call Forwarding All from a different extension, dial *74; for Call Forwarding Busy, dial *92. You also can activate and deactivate Call Forwarding from any Aastra phone using our default setup.
  • Call Waiting
    To activate Call Waiting from any extension (which is the default), dial *70. To deactivate Call Waiting, dial *71.
  • Call Pickup
    To pickup a call ringing on another extension, dial **.
  • Zap Barge
    To barge into an existing call, dial 888.
  • Call Transfer: Attended and Blind
    For attended call transfers where you can remain on the line until the other party answers, dial *2. For unattended call transfers, dial ## and then the number to which the call should be transferred.
  • Dictation Service with Email Delivery
    Before using FreePBX's dictation service, you must activate Dictation Services for the specific extension to be used. Using FreePBX, go to Setup, Extensions, and click on the desired extension. Scroll down to the Dictation Services section of the form and enter your email address, the format of the sound files to be used, and change Dictation Service to Enabled. Save your settings and reload the dialplan. Then you can dictate your message by dialing *34. Once you finish your dictation, you can email it to your email address for this extension by dialing *35.
  • Do Not Disturb
    To activate Do Not Disturb on any extension, dial *78. To deactivate Do Not Disturb, dial *79. There's also a button to accomplish the same thing with our Aastra phone setup.
  • Phonebook Dial by Name - Dial 4-1-1
  • VoiceMail Options
    To retrieve your voicemail from any phone, dial *97. To retrieve voicemail for a different extension, dial *98 or *98701 where 701 is the extension desired. To leave a voicemail message for any extension with voicemail enabled, dial *701 where 701 is the extension desired.
  • Speed Dial
    To set up a user speed dial entry, dial *75. To call any previously established speed dial entry, dial *0 plus the speed dial number. To create or modify speed dial entries in FreePBX, click Tools, Asterisk Phonebook. You also can import entries from a CSV-formatted file.
  • Flite and Cepstral Text to Speech (TTS)
    Flite TTS is installed by default with all PBX in a Flash systems using Asterisk 1.4 or 1.6. Cepstral can be installed using the directions below with Asterisk 1.4. To use Flite with Egor in your dialplan, here's the syntax:

    exten => 444,5,Flite("Hello World.")

    To use Cepstral with Allison in your dialplan, use this syntax:

    exten => 444,5,Swift("Hello World.")
  • One-Click (almost) Cepstral TTS Install with Allison
    After logging in as root, type install-cepstral to install Cepstral. Accept all the defaults except create the missing directory when prompted by the install script to do so. For detailed instructions on reconfiguring Nerd Vittles apps to use Cepstral instead of Flite, see this article. No software needs to be reinstalled. Simply change the dialplan and PHP app settings to use Cepstral as explained in the article. For more background on Cepstral, read this article. To register your newly installed Allison voice, go to this link. Be sure you select U.S. English language, Allison-8kHz voice, and Linux platform before you check out, or it's money down the drain. Write down the name, company (optional), and key that is issued once you fill in the blanks. Then log into your server as root, and type swift --reg-voice. Fill in the blanks with the information you wrote down above, and you're all set.
  • Windows Networking with SAMBA
    Windows Networking with SAMBA is disabled by default in this special build. The default workgroup is "workgroup." To change the workgroup, log into your server as root and edit /etc/samba/smb.conf. To start SAMBA, type service smb start. You then can connect to your server from any computer that supports Windows networking using root as your username and whatever root password you created. For more setup tips and to configure SAMBA for automatic startup on boot, click on the link above.
  • Linux Firewall
    The IPtables firewall is enabled by default in all PBX in a Flash systems. For this build, we have disabled SAMBA access to your server. To enable it, log into your server as root, and edit /etc/sysconfig/iptables by adding the following three lines just above the COMMIT line at the end of the file:

    -A INPUT -p udp -m udp --dport 137:138 -j ACCEPT
    -A INPUT -p tcp -m tcp --dport 139 -j ACCEPT
    -A INPUT -p tcp -m tcp --dport 445 -j ACCEPT

    Then issue the following command to restart SAMBA:

    service iptables restart
  • WebMin
    WebMin is often described as the Swiss Army Knife of Linux. It provides a terrific web interface to Linux.everything. It is enabled by default in this install. To access it using a web browser, go to http://serverIPaddress:9001/ and login as root with the password you set up above for WebMin access. For complete documentation, go here.
  • PBX in a Flash Software Update Service To Keep Your System Current
    To load current fixes for this build of PBX in a Flash, log into your server as root and type the following commands:

    update-scripts
    update-fixes

More Good News with the Everex gPC2. From the "Learn Something New Every Day Department," this newsflash. The Everex gPC2 has built in hardware SATA RAID 1 support that actually works. What you'll need to get this going is a second 80GB hard disk to match the one delivered in your original box. Total cost: about $40. If one disk fails, the other kicks in automatically. Here's a link to purchase your drive. And here's the link that'll tell you how to get everything set up. Before you begin, make certain that you have a current ISO backup on your flash drive so that you can restore your system once the RAID setup is up and running. See the top of this article for the backup and testing procedure.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

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The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 


Some Recent Nerd Vittles Articles of Interest...

  1. Join the following line whenever you encounter this character: ↵ []

The Asterisk Mother Lode: Introducing the Orgasmatron II for the $199 Everex gPC2

Well, okay. Today's creation still doesn't quite measure up to the legendary Orgasmatron... but, we're getting closer. It's been several months since we released our first Orgasmatron for Asterisk®. Much has changed both in Asterisk and in the hardware and software environment since then. So today, to celebrate the release of PBX in a Flash 1.3 and FreePBX 2.5, we're taking another stab at building the Ultimate Kitchen Sink. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... 15 minutes! There's now a custom build for the Dell SC440 as well. Here's the link.

Our approach today is refined a bit since the last time around. The processing overhead of CentOS 5.2 continues to make VMware problematic. Luckily, the price of hardware continues its downward spiral. So today we're comfortable recommending the best phone, the best value PC, and our own new entry in the VoIP provider sweepstakes. But, you'd better hurry, there's only one retailer still carrying the Everex Green PC: good old WalMart. And now you can even get free shipping of the unit to the WalMart store of your choice.

If you've been following along with our articles, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i, and we've also identified a perfect small business/home computer on which to run a production Asterisk server for about 50 employees, the Everex gPC2 (aka "The WalMart Special"). So this build provides a preconfigured gPC2 installation on a 2-disk ISO image backup of the whole system using Mondo. And, NO, it won't work with any other hardware! Once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own Everex gPC2. But you must have a gPC2 so accept no substitutes, or you may end up with an Electronic Brick instead of an Orgasmatron II. Once again for the reading impaired, the $199 gPC2 systems are only available from WalMart.

We've preconfigured some extensions on your new system as well as outbound and incoming trunks from some terrific providers including our own new entry for VoIP terminations. Ours is dirt cheap, of course, at just over a penny a minute in the U.S. and about half that in many parts of Canada. You literally can sign up for service, plug in your phones, and have a system in full operation in under an hour.

So... what do you get with this preconfigured build? In addition to all of the goodness of a stock PBX in a Flash 1.3 build including Asterisk 1.4.21.2 running under CentOS 5.2, you also get the brand new FreePBX 2.5 as well as the latest versions of Apache, MySQL, PHP, and SendMail. And you get a Baker's Dozen preconfigured Nerd Vittles applications. Complete documentation is available here.

  • Inbound and Outbound VoIP Faxing Using nvFax... finally!
  • FONmail for Asterisk to send voice messages to any email address on the planet
  • AsteriDex RoboDialer and Telephone Directory
  • Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
  • NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
  • Weather Reports by Airport Code (TTS)
  • Weather Reports by ZIP Code (TTS)
  • Worldwide Weather Forecasts (TTS)
  • xTide for Asterisk (TTS)
  • MailCall for Asterisk: Get Your Email By Telephone (TTS)
  • TeleYapper 4.0 Message Broadcasting System
  • CallWho for Phone Lookup and Dialing of Entries in the AsteriDex Database (TTS)
  • TFTP Server with preconfigured setups for 15 Aastra 57i SIP telephones

In addition, you get dozens of preconfigured telephony applications and functions that would take even an expert the better part of a year or two to build independently. And, unlike all of the other distributions, we build Asterisk from source so it's simple to modify and upgrade whenever you feel the need. Here's a short list of what you have to look forward to:

  • Stealth AutoAttendant with Welcome and Application IVRs
  • Key Telephone Support Using Park and Parking Lot
  • Intercom/Paging Support
  • Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
  • DISA
  • Blacklisting with Web and Telephony Interfaces
  • CallerID Name Lookups from 8 Providers
  • Weekly Automated System Backups to a Flash Drive
  • One Touch Day/Night Service
  • Music on Hold
  • Voicemail with Email Delivery of Messages and Pager Notification
  • Voicemail Blasting
  • Cell Phone Direct Dial
  • Call Forward: All, Busy, No Answer
  • Call Waiting
  • Call Pickup
  • Zap Barge
  • Call Transfer: Attended and Blind
  • Dictation Service with Email Delivery
  • Do Not Disturb
  • Gabcast
  • Phonebook Dial by Name
  • Speed Dial
  • Flite Text to Speech (TTS)
  • Windows Networking with SAMBA
  • Linux Firewall
  • PBX in a Flash Software Update Service To Keep Your System Current
  • One-Click Cepstral TTS Install with Allison... Just Type install-cepstral

Prerequisites. As mentioned, you'll need a $199 Everex gPC2 (aka The WalMart Special) to use this build. We also recommend an additional $25 gig of RAM for anything other than home use. We also recommend a 4GB USB flash drive on which to store automatic weekly backups of your new system. Just plug it into your new machine, log in as root, and type: /root/usbformat.sh. That's it! Every Sunday night, you'll get a new backup in ISO format on your flash drive. If something goes wrong on your system, copy the ISOs to CDs and reboot with Disk 1. It doesn't get any easier than that. And you can always check on the latest backup by issuing the command: /root/usbcheck.sh

Finally, you'll need to cough up a whopping $5 to download the two-disk ISO image for this build. And, yes, we eat our own dog food. The ISO images you'll be downloading were captured as a backup on the flash drive of our gPC2 lab machine. If you use this special build, it seemed only fair that you cover the cost of the bandwidth to download it. As most of you know, we don't have the luxury of freeloading off SourceForge for our downloads. And we didn't want to impose upon our existing bandwidth providers to bring you this custom image. The good news is that, once you download the image from DreamHost, you are more than welcome to pass it along to one or more of your friends or business acquaintances at no charge. You can even do it electronically through the DreamHost Files Forever program. And, if you'd like to host this image for your fellow man at no cost, be our guest... and thank you! Bottom line: For about $250, you'll have the slickest, most reliable PBX and fax machine on the planet with rock-solid weekly backups and, of course, access to the one-of-a-kind PBX in a Flash Software Update Service!

Getting Started. Once you have purchased your Everex gPC2, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. Having said that, we strongly recommend that you always keep your system running behind a NAT-based firewall/router. We strongly recommend the dirt-cheap dLink WBR-2310 WiFi router which handles NAT issues with VoIP masterfully. Don't redirect any ports to the machine and don't turn the PC on just yet.

Download the two ISO images for the gPC2 from here. If you don't know how to create a CD from an ISO image, read that section from our previous article. In fact, read the whole article. It'll help you immensely down the road. Once you have the two CDs in hand, turn on the gPC2 and quickly insert Disk 1 into the CD/DVD drive and close the drive. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names, halt the install by pressing CTL-ALT-DEL and remove the CD. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 12 minutes. When prompted, insert Disk 2 and press the Enter key to finish the install. When the CD ejects, remove it and your gPC2 will reboot after you perform the three-finger salute (Ctl-Alt-Del).

After the reboot finishes, type root at the login prompt for your username and password for your password. The IP address assigned by your DHCP server should appear near the top of the screen. Write it down. If there is no IP address, your machine does not have network connectivity or access to a DHCP server with an available IP address. Correct the problem and reboot.

Securing Passwords. We're going to change five passwords now. For the time being (until you've done some reading), think up one really difficult password (that you won't forget) and use it for all five passwords. At the root@pbx:~ $ command prompt, type the following commands and type in your new password when prompted. Don't forget your password or you'll get to put in your two CDs and start over.

passwd
passwd-maint
passwd-wwwadmin
passwd-meetme
/usr/libexec/webmin/changepass.pl /etc/webmin root yournewpasswordhere

Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click the Admin tab and then choose the FreePBX Administration botton. Log in as maint with your new maint password. Before you do anything else, change ALL of the 16 extension passwords to something secure... as if your phone bill depended upon it! Click Setup, Extensions and then choose each extension, modify BOTH the device secret and Voicemail Password, and click Submit. When you finish all the extensions, then reload the dialplan to save your changes. Finally, change your DISA password to something very, very secure: Setup, DISA, DISAmain, PIN. Reload your dialplan once again to save your changes.

Regardless of what you may read elsewhere, the Orgasmatron II has all the very latest security patches as of October 1. If you want more security, take our advice and add a hardware-based firewall/router between your Internet connection and your new Orgasmatron II and don't expose port 80 (the web interface) to the Internet!

Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set. Either way, you need a permanent IP address for your machine when all is said and done. Once you have a permanent IP address, hop on over to dyndns.org and sign up for your own fully-qualified domain name (FQDN), e.g. mypbx.dyndns.org. You're going to need it for a whole host of things with your new PBX, and dyndns.org is about the easiest way to do it. Once you have your FQDN and DynDNS username and password, log in as root and edit: /etc/ddclient/ddclient.conf. Search (Ctl-W) for ***. Fill in your username and password and uncomment those two lines. Then search for *** again, uncomment the next three lines and fill in your fully-qualified domain name. Save the file and service ddclient restart. To make sure everything worked, issue the following command: ddclient -force. Assuming there are no errors, issue the following command to start ddclient each time your server reboots: /sbin/chkconfig --add ddclient. Now the IP address of your Asterisk server will always resolve to your FQDN from DynDNS. And anyone can call you via SIP for free using the following SIP URI: mothership@yourFQDN.dyndns.org. You can take this a step further and sign up for a free incoming phone number at ipkall.com. For your account type, choose SIP. For your SIP phone number, enter: mothership. For your SIP proxy, enter the fully-qualified domain name (FQDN) for your server, e.g. mypbx.dyndns.org. Choose a password and enter your real email address, and they will beam you a Washington state phone number within a day or so. You can't beat the price!

Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone (highly recommended). Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack of cigs) known as an SPA-2102. It's under $70. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the device into your LAN, and then plug your phone instrument into the SPA-2102. Note that this adapter supports two-line cordless phones! Your router will hand out a private IP address for the SPA-2102 to talk on your network. You'll need the IP address of the SPA-2102 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The device will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.

Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab and then repeat this drill for the Line2 tab if you want to connect the device to two extensions on your Asterisk system. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter your actual password for this extension in the Password field, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Now repeat the drill for Line2 using extension 702. Pick up a phone and dial 1234# to test out BOTH extensions.

Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with FreePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best phone out there is the Aastra 57i for under $200. Another $100 buys you the Aastra 57i CT with a cordless DECT phone.

Configuring Aastra 57i SIP Phones. Your new system comes preconfigured to automatically configure up to 15 Aastra 57i phones. Plug each phone into your network and wait for it to boot. Once it boots, press the Option button, then Phone Status (3), then IP & MAC Address (1). Write down each phone's IP address and MAC address. Then press Done to exit from the menus.

Next, we need to tell your phone to use your new Asterisk server as the TFTP server to obtain its setup. Press the Option button again, then Admin Menu (5). Type 22222 for the admin password and press Enter. Then choose Config Server (1), then TFTP Settings (2), then Primary TFTP (1), enter the IP address of your new server, and press Done a half dozen times.

Log back into your server as root. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone AND rename each file (filenames are 701.cfg to 715.cfg) to the MAC address of each phone.cfg. Do NOT use hyphens in the MAC address. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!

sed -i 's|192.168.0.0|192.168.0.123|g' /tftpboot/aastra.cfg

Now restart each phone by pressing the Option button and then Restart Phone (6) and then the Restart button. Once the phone reboots, you can make a test call by dialing 1-2-3-4. You can get the latest news by dialing 5-1-1. Or get a weather forecast by airport code (6-1-1) or zip code (Z-I-P).

A Word About Ports. For the techies out there that want to configure remote telephones or link to a server in another town, you'll need to know the ports to remap to your new server from your firewall. Here's a list of the ports available and used by PBX in a Flash. We don't recommend exposing UDP 5038 which is used to communicate with Asterisk via the Asterisk Manager.

TCP 80 - HTTP (needed to access the web sites on your server from the Internet... not recommended!!!)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access... not recommended!!!)
UDP 10000-62000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX communications typically between Asterisk servers)

Setting Up Trunks for Outgoing and Incoming Calls. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. And you'll need a phone number (DIDs) so that folks can call you. Unlike the Ma Bell world, you need not rely upon the same provider for both. And nothing prevents you from having multiple outbound and incoming trunks to your new PBX. At a minimum, however, you do need one outbound trunk and one inbound phone number unless you're merely planning to talk to other extensions set up on your system. We've actually put all the hooks in place to make it easy for you to interconnect to other Asterisk servers, but we'll save that for another day. For today, we want to get you a functioning system so that you can place outbound calls to anywhere in the world and can receive incoming calls from anywhere in the world.

For outbound calling, we recommend you establish accounts with several providers. We've included the necessary setups for our own service as well as Vitelity and AOL. To register for our service, just dial any 10-digit phone number from a phone on your system before you set up any other trunks. We're one of the least expensive providers, but you know the old saying about that. Give us a try and, if you don't like the call quality, do some more shopping. We think it's pretty good quality actually, but we don't sell DIDs for inbound service... yet.

Vitelity: One of the Best Providers on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus reasonable entry level pricing plus high quality calls, then Vitelity is a winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. For PBX in a Flash users, sign up before October 15, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price (except with us) and the call quality is excellent as well. We've tried just about everybody.

To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes. The same setup drill will get you going the the PIAF VoIP service as well, and you have your choice of the following POPs: Houston, Dallas, LAX, NYC, London, Montreal, and Toronto. The POP addresses are entered in the following format: sip.lax.pbxinaflash.net or sip.london.pbxinaflash.net.

To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good SIP service can be, pick up a phone on your system and dial D-E-M-O. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications.

An Alternate Outbound Calling Solution. As we said, it costs you almost nothing to add an alternate outbound calling solution to your new system. As luck would have it, adding a third outbound calling provider is now a breeze because AOL just entered the SIP terminations market with a product called AIM Call Out. We wrote about it recently, and you can read the article here. All you need is an AOL or AIM account name and $5 to get you started. The system you've just installed is preconfigured to use AIM Call Out. All you have to do is plug in your username and password, and you can immediately make calls to anywhere in the United States for under 2¢ per minute. Adding international calling is as easy as inserting the correct dial string. If you never use it, it doesn't cost you a dime. So $5 is mighty cheap insurance in our book.

First things first. Sign up for the service at this link. Your username will look something like this: johndoe@aim.com. You also will be assigned a password. Using your web browser, open FreePBX by pointing to the IP address of your new server and choosing Administration, then FreePBX. Type in admin as your username and the password you assigned to your system. From the main FreePBX menu, choose Setup, Trunks, and click on SIP/AIM in the far right column. Scroll down to the Peer Details section of the form and replace yourAIMpassword with your new password. Then replace yourAIMaccountname with your actual AIM account name. Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Setting Up an Alternate DID for Incoming Calls. You also may want to consider a second phone number where people can call you. For example, if Grandma and Grandpa happen to be in another state and still have an old fashioned telephone, you might consider adding an additional DID to your system in their area code. They then can make a local call to reach you by dialing the local DID. On the les.net pay-as-you-go plan, it costs less than a dollar a month plus a penny a minute for the calls. Money well spent if we do say so... and you'll sleep better.

If this setup looks a bit complicated, don't be intimidated. Remember, we're connecting your PBX to the rest of the world so people can call you! With les.net, you have a choice of rate plans for most DIDs. You either can pay $3.99 a month for unlimited inbound calls with two concurrent channels or 99¢ per month and 1.1¢ per minute with four concurrent channels. Just visit their site and click Signup to register. Once you are registered, click Login and then Order DIDs. Pick a phone number. Then click Peers/Trunks and Create New Peer. Write down the Peer Name as you will need it in a minute to set up your connection. Choose SIP for Peer Technology, RFC2833 for DTMF Mode, G.711 for Codecs, Registration for Peer Type, enter the public IP address of your server for Peer Address, make up a secure password and write it down also, specify an Outbound CallerID for your calls, and check the 10-digit dialing box. Leave voicemail unchecked since you'll handle this on your end. Save your changes.

Now choose Your DIDs and click on the one you just ordered. We now need to tie the phone number to the Peer setup you just created above. Click on the DID and select the Route to Peer which you just created. Check the Send DID Prefix box and leave everything else blank. Click Save Changes and you're finished at the les.net end. Now let's set up your inbound DID trunk in Asterisk using FreePBX.

Log into FreePBX using a web browser. Click Setup, Trunks and then Add SIP Trunk. Fill in the CallerID and then drop down to the Outgoing Settings section of the form. For Trunk Name, use the Peer Name that you created above and wrote down. It ought to look something like this: 1092832198. For Peer Details, enter the following using the Peer Name and Password you assigned at les.net:

canreinvite=no
context=from-trunk
fromuser=1092832198
host=did.voip.les.net
insecure=port,invite
nat=yes
secret=yourpassword
type=peer
username=1092832198

For Incoming Settings, use from-pstn for the User Context and enter the following User Details:

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=port,invite
nat=yes
type=user

For the registration string, enter a string like the following using your Peer Name and Password:

1092832198:yourpassword@did.voip.les.net/1092832198

Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Choosing a VoIP Provider That Supports Faxing. We've included a reliable fax solution in this build, and we'll cover all the details next week. We do want to give you a head start if you plan to use your new machine to handle inbound faxes. To test your machine, you can connect a real fax machine to one of the lines on an SPA-2102. Then send a fax to extension 329 (F-A-X). But first you must configure your email address in two places using FreePBX: Setup, General Settings, Email address to have faxes emailed to AND Setup, Inbound Routes, any DID / any CID, fax Email. Once you've saved your settings, send the fax and see if it's delivered to your email address. If it works reliably, then the fax and email applications on your machine are configured correctly. Unfortunately, that's only half the battle. To receive faxes from outside your system, you'll also need a DID from a provider that supports faxing. And then it's still only about a 90% proposition... on a good day. We've tested this with many, many VoIP providers. Some work. Many don't. Some, such as Vitelity, offer a faxing service for a fee. Guess what? Their regular VoIP setup doesn't support faxing. Our old friends at Telasip.com still support faxing. We've also had good luck with Future-Nine and Teliax. You can read the beginnings of our fax dissertation here for more details. With the exception of the trunk setup covered in the article, all of the remaining setup steps already have been completed on your new server!

Choosing a Preferred Provider. Finally, you'll need to decide whether to use us or AOL or Vitelity as your primary terminations provider. HINT: We're the cheapest! So we've set things up with us and then AOL. This is handled in FreePBX in the Outbound Routes tab under the Default entry. You can adjust easily these in any way you like by adding trunks or moving entries up and down the list to change their priority. Just be sure to leave ENUM at the top of the list since ENUM calls are always free. If a free call isn't possible, your server will automatically drop down to the next trunk in the priority list. Don't add Vitelity to the list unless you have actually created a Vitelity account since they handle unsuccessful connections in a non-standard way which will cause FreePBX not to drop down to the next trunk to attempt a connection.

A Word About Mondo Rescue. We would be remiss if we didn't mention what a fantastic open source product Mondo Rescue is. It's the sole reason that today's build was possible. Our special thanks go to the development team: Bruno Cornec, Andree Leidenfrost, and Hugo Rabson. It is the first (and only) backup software for Linux builds that actually works reliably. The best way to prove that for yourself is to download this build and try it for yourself on your Everex gPC2. It has much more flexibility than what you will experience, but that would take another dozen pages to explain. We'll save that for another day. In the meantime, if you'd like more information, visit the Mondo Rescue web site.

Where To Go From Here. Well, we've covered a good bit of territory today so we're going to save the really fun stuff for our next installment. In the meantime, you have a new phone system that works. And there are a number of PDF documents in the /root folder on your new system which are worth a read. Better yet, you can browse through all of the documentation which is available for PBX in a Flash by going here. You also can dial D-E-M-O on your new system and see just how powerful direct SIP connections can be to other Asterisk hosts (in this case, ours!)... at no cost. Finally, you can log into your server and type help-pbx for access to a treasure trove of additional features. Enjoy!

Continue reading Part II...


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

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The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 


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