IncrediblePBX – Page 51

Paradise Lost… and Found Again: Incredible PBX Returns

Monday, December 6, 2010 / 23 Comments on Paradise Lost… and Found Again: Incredible PBX Returns

For the tenth time in as many months, Google is up to its old tricks again with Google Voice access. Beginning December 1, Gtalk outbound calling requires a web connection with all the cookie mess that’s made Python a household word. And we’re back to the 2-call mumbo jumbo that we started with in the original Incredible PBX. There’s some good news though. First, the return call can use Gtalk as the transport mechanism rather than having to acquire yet another phone number from yet another provider. So the return call connection is virtually instantaneous. Our special thanks to Lost Trunk on the PIAF Forums for his discovery and the pioneering work he did to get everything working again. And second, Asterisk® 1.8 now provides transparent call bridging so the days of jumping in and out of the Asterisk Parking Lot to complete an outbound call are over. For the non-techies, it just works! So just follow the installation procedure in the original article and you’ll be good to go. No need to keep reading unless you enjoy getting down in the weeds.

We’re getting smarter with every Google Voice iteration. In fact, we kinda saw this one coming before Incredible PBX 1.8 was released. All of the Python plumbing already was built into Incredible PBX 1.8 just in case we needed to return to the old fashioned way of doing things. So it was a relatively easy fix to adjust the Asterisk dialplan and FreePBX® contexts to support the new requirements.

There are a few more Christmas bonuses today as well. First, you now can add as many Google Voice accounts to your server as you’d like. So Tom, Dick, and Mary can all have their own Google Voice number if desired. Second, for those that like to experiment and aren’t comfortable with Python scripts, there now are bash and AGI scripts to make Google Voice calls as well. We haven’t played with these, but here are links to the gvout bash script docs and AGI script documentation (such as they are) if you’d like to experiment. Just don’t ask us any questions. But, by all means, share your discoveries!

Can An Existing System Be Upgraded? Let’s begin with the question you’re all wondering about. If you have an existing Incredible PBX 1.8 system, can it be retrofitted so that outbound calling once again works? And the answer is YES! Here’s how. You’ll need to make some changes both in your dialplan and using the FreePBX GUI. Just follow along below. It’s not hard.

Dialplan Adjustments. For the dialplan modifications, we’ve written a script which makes it easy. Log into your server and enter the following commands. When prompted, enter the Google Voice email address and password that you already are using with Incredible PBX 1.8. Then enter the Google Voice email address for call notifications, and you’re all set. If you’ve forgotten which Google Voice email address and password you’re using, then issue the following command: cat /etc/asterisk/jabber.conf.

NOTE: This script assumes you have made no additions to the end of extensions_custom.conf. If you have made changes, then you’ll need to put your changes back into extensions_custom.conf once it has been updated. The original version of the file has been renamed to extensions_custom.conf.last.

cd /root
wget http://incrediblepbx.com/fixit.sh
chmod +x fixit.sh
./fixit.sh

If you happen to screw up and enter your credentials wrong, it won’t hurt to run the script again. Just save a copy of extensions_custom.conf.last with a different file name so you don’t lose the contents of your original file in the event you made changes to it.

FreePBX Changes. Now for the manual stuff. Using your browser, log into FreePBX at http://ipaddress/admin using maint as the username and the password you set up with passwd-master. It’s easier to explain the process using an example so let’s assume the Google Voice account you’re using with Incredible PBX is johndoe@gmail.com. We’ll refer to this as your Google Voice Email. We’ll also be using your Google Voice Name which would be johndoe without the @gmail.com. Finally, you’ll need your 10-digit Google Voice DID (aka phone number) with a +1 prefix: +18431234567. You’re not going to need your Google Voice notification email address. We’ve already taken care of that in the extensions_custom.conf dialplan code.

Step #1: Make yourself a Cheat Sheet with these 3 pieces of information for your actual account:

Google Voice Email: johndoe@gmail.com
Google Voice Name: johndoe
Google Voice DID: +18431234567

Step #2: Create a new Custom Trunk in FreePBX: Setup, Trunks, Add Custom Trunk:

Trunk Name: gvoice-johndoe
Custom Dial String: local/$OUTNUM$@gvoice-johndoe

NOTE: If we don’t show an entry on the FreePBX form, leave the default. After each step, save your entries and reload your dialplan when prompted.

Be very careful with these entries. A single typo means nothing will work. Don’t post a comment saying nothing works, or you’ll get a cryptic response that says, "Check for typos!" 🙄

Step #3: Create a new Custom Destination in FreePBX: Tools, Custom Destinations, Add Custom Destination:

Custom Destination: googlein,johndoe@gmail.com,1
Description: GV In – johndoe

Step #4: Create a new Inbound Route in FreePBX: Setup, Inbound Routes, Add Incoming Route:

DID Number: gvoice-johndoe
CallerID Number: +18431234567
CID Lookup Source: Caller ID Superfecta
Custom Destination: GV In – johndoe

Step #5: Modify the Default Outbound Route in FreePBX: Setup, Outbound Routes, Default. Delete the ENUM Trunk Sequence by clicking on the Trashcan icon beside it. Then change the remaining Trunk Sequence entry to the following: gvoice-johndoe.

Step #6: Modify the GoogleVoice Outbound Route in FreePBX: Setup, Outbound Routes, GoogleVoice. Change the existing Trunk Sequence entry to: gvoice-johndoe.

Step #7: Modify the TollFree Outbound Route in FreePBX: Setup, Outbound Routes, TollFree. Change the existing gvoice Trunk Sequence entry to: gvoice-johndoe.

Step #8: Reboot your server for good measure.

UPDATE: If you still have trouble completing outbound calls, here’s a temporary fix to address what appears to be a bug in Asterisk 1.8.0. Log into your server as root and issue the following commands once or twice. Then all will be well… at least until you reboot. 😉

amportal restart
/root/jab

UPDATE #2: Kudos to Doktur on the PIAF Forums for actually finding a permanent outbound calling solution. It turns out that, unlike Asterisk 1.6, Asterisk 1.8 now requires that the Python gvoice script be run with root privileges. This now has been fixed in the base install so you won’t have to worry about it. However, to update an existing system, here’s a link to the commands to fix it once and for all.

Adding Multiple Google Voice Accounts. Once you’re sure everything is back to normal, and you can successfully make outbound calls and receive inbound calls from your Google Voice number, then you’re ready to add support for additional Google Voice accounts. Set up your Google Voice second account using the steps outlined in the Incredible PBX 1.8 article. Then do the following:

Step #1: Repeat the first seven FreePBX steps shown above using the Google Voice credentials for your second account instead of your first. In steps 4, 5, and 6, simply add your second account as an additional Trunk Sequence. This will provide a second outbound call path when the first one is busy.

Step #2: Log into your server as root and edit /etc/asterisk/jabber.conf. Copy the entire [asterisk] context and name it [asterisk2]. Then replace the username and secret in [asterisk2] with your new Google Voice name and password for your second account.

Step #3: Edit /etc/asterisk/extensions_custom.conf. Find the [googlein] context and make a duplicate copy of the top section of code which includes all lines beginning with exten => johndoe@gmail.com. In the duplicate section, replace johndoe@gmail.com with your second account Google Voice Email, replace both gv_dialout_johndoe entries using your second account Google Voice Name, and change the ALERTNAME email address in the second line, if desired.

Step #4: While still in extensions_custom.conf, make a duplicate of the [gvoice-johndoe] context which should be at the end of the file. Copy the duplicate just below it and name it using your second account Google Voice Name, e.g. [gvoice-marydoe]. Enter your Google Voice credentials for the second account in the first and eighth lines of the new context. Remember to modify the email address twice in the first line! In the third, fifth, and seventh lines, change gv_dialout_johndoe to reflect your second account Google Voice Name. Save your changes.

Step #5: Reboot your server. Enjoy!

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

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…

Avoiding a $100,000 Phone Bill: VoIP WhiteList for IPtables

Friday, December 3, 2010 / 1 Comment on Avoiding a $100,000 Phone Bill: VoIP WhiteList for IPtables

It’s been almost a year since we last wrestled with VoIP security for Asterisk®. With Christmas just around the corner, it seemed like a fitting time for a report card. Suffice it to say, the bad guys have not stood still. Attacks have become much more frequent and more sophisticated as VoIP systems have proliferated. A year ago we saw brute force attacks with thousands of password attempts on VoIP servers. These attacks could easily be detected by Fail2Ban. What we are seeing today are one and two hit drive-bys that usually are initiated from Windows zombies or hosted accounts established with stolen credit cards. These VoIP attacks fly under the radar unless you review your logs every day. Have the creeps gotten more patient? No, just smarter. They now understand the VoIP security model that has been deployed on systems like PBX in a Flash, and they simply work around it. Two hits per server, and they’re off to the next IP address only to return in a few hours to try two more. Are these attempts successful? Well, here’s the latest recipient of a $100,000 phone bill so the answer would appear to be affirmative.

We continue to wrestle with new security approaches to better protect Asterisk VoIP systems, and we’ve stumbled upon another golden arrow for your security quiver. Our Incredible PBX platform continues to offer the very best security solution because it is designed to sit safely behind a hardware-based firewall with virtually no exposure to the Internet. But such deployments assume that both your server and your phones are all safely ensconced behind a hardware-based firewall. If it turns out that you want to deploy a SIP phone for use by grandma or you’ve decided you’d like to try hosted PBX service from a provider such as rentpbx.com,¹ then there either need to be holes opened in the firewall or there is no hardware firewall protection in the case of hosted service.

Over the past few weeks, we’ve explored a number of new security approaches to better protect your Asterisk server. These include The SunshineNetworks Knock as well as VoIP Black Lists and VoIP White Lists. If you’re technically savvy, you’ll want to carefully consider "The Knock" for all of your SIP phones exposed to the Internet.

We spent a good bit of time considering various VoIP BlackList solutions. As the name implies, a list of the bad guys’ IP addresses is fed into IPtables which then blocks access to your server from these addresses. Sounds good, right? One approach with a BlackList is to block all IP addresses from "problem countries." The methodology to implement this solution can be found in this thread on the PIAF Forums. The problem, of course, is identifying the "problem countries." Another option was to implement an IPtables Blacklist based upon the work of the VoIP Blacklist Project. Perhaps ironically, the VoIP Blacklist Project actually blocks the IP addresses of both Nerd Vittles and PBX in a Flash, and emails requesting removal of our IP address were ignored. To save time, the VoIP Blacklist Project employs CIDR Masks which can blacklist hundreds of thousands of IP addresses in one fell swoop. Problem is that a lot of innocent people get caught in the net, and there’s no easy way out without maintaining the blacklist yourself. The final dagger in the black list approach is zombies. Insecure Windows machines have been compromised by the droves worldwide and particularly in the United States. So identifying all of these now-malicious systems is not unlike playing Whack-a-Mole. When you block one of them, six more pop up. So, after giving it the good old college try, our view of VoIP Blacklists should be obvious. No, thanks. There are very real risks that the bad guys can and have poisoned existing blacklists with safe IP addresses, and the number of Windows zombies grows geometrically making it all but impossible to have or maintain a blacklist that affords any real protection.

These results with black lists led us to the conclusion that the only real security mechanism that could protect many VoIP servers today was a VoIP WhiteList for IPtables. As the name implies, we want to identify the IP addresses of every SIP and IAX trunk and extension on your server and then feed those addresses into IPtables so that the only access to VoIP resources on your server is from these addresses. Today’s VoIP WhiteList for IPtables consists of two bash scripts: one queries the MySQL database in which FreePBX stores all of the trunk and extension information for your server and the other populates IPtables with the results of the queries. We would hasten to add that a similar white list is equally important for SSH access to your server although we think it is better to implement an SSH WhiteList on your hardware-based firewall. In this way, you can adjust the SSH white list via web browser while traveling without locking yourself out of your Asterisk server.

Prerequisites. To use today’s VoIP WhiteList for IPtables, you’ll need either a current version of PBX in a Flash or Incredible PBX. Other aggregations will also work provided your system is FreePBX-based (version 2.6 or later), has IPtables already installed and functioning properly, and has an /etc/sysconfig/iptables configuration file that closely matches the stock PBX in a Flash design. We’ll leave it to you to make that call after reviewing the scripts.

VoIP WhiteList Design. We’ve designed the VoIP WhiteList for IPtables to be modular. There’s a firewall-whitelist-gen.sh script which extracts from MySQL the list of IP addresses used by your trunks and extensions. This text-based list is stored in /etc/firewall.whitelist. You can manually add and delete entries from the list once it is populated.You also can rerun the script at any time to generate a fresh catalog of WhiteList IP addresses based upon your current trunk and extension settings. This script also enables access to your server from the public IP address of your server as well as all non-routable IP addresses. Finally, it modifies /etc/sudoers slightly so that Travelin’ Man can be used to add dynamic IP addresses on the fly. We’ll cover that below.

The second script is firewall-whitelist.sh, and it is used to actually implement your new VoIP WhiteList in IPtables. The changes take effect immediately. It also can be run again to update these entries if you manually add or delete IP addresses in /etc/firewall.whitelist. This script always creates a backup copy of your previous /etc/sysconfig/iptables file and names it iptables.timestamp where the timestamp is the date and time of your last update, e.g. iptables.12012010-083841 was created on Dec. 1, 2010 at 08:38:41. If you should ever shoot yourself in the foot, simply copy one of the iptables backup files to /etc/sysconfig/iptables and then restart IPtables: service iptables restart.

WARNINGS: In order to implement the WhiteList, the script removes the existing IPtables entries which permit SIP and IAX access from anywhere using UDP ports 4569 and 5000 to 5082. If you have edited these entries in any way, you’ll need to remove them and restart IPtables before running firewall-whitelist.sh. Otherwise, your more general firewall entries will leave your system vulnerable to access from IP addresses not in your VoIP WhiteList.

If your system is running on a hosted server, you’ll need to make a couple of additions to /etc/sysconfig/iptables and restart IPtables (service iptables restart) before running firewall-whitelist.sh, or you may lock yourself out of your own server. Be sure to add the public IP address of your server, and also add the IP address from which you are making changes to your server. Each entry should look like the following example using your actual IP addresses. And the entries should be added above the COMMIT line in the same section of the iptables file as the existing UDP 10000:20000 ACCEPT entry:

-A INPUT -s 222.222.222.222 -j ACCEPT

Installing the VoIP WhiteList for IPtables. Installation is easy. Just log into your server as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/firewall-whitelist.tar.gz
tar zxvf firewall-whitelist.tar.gz
./firewall-whitelist-gen.sh
./firewall-whitelist.sh

If you installed one of the beta versions of the VoIP WhiteList from the PIAF Forums, then you’ll need to do a little housecleaning before actually running either of the scripts. Just edit /etc/sysconfig/iptables and clean out all of the entries that contain 5000:5082 as well as any entries nearby that include the non-routable IP addresses, e.g. 192.168.0.0. Finally, if there are entries beginning with -A WHITELIST, delete those as well. Then restart IPtables: service iptables restart. Thank you for your testing and feedback!

Deploying Remote SIP Phones. What remains is some method for connecting remote SIP phones with dynamic IP addresses. Our Travelin’ Man application was specifically designed to provide this support although the initial version only opened the necessary IP address for Asterisk access. The latest release also provides the necessary IPtables support. You have two options: either remove the old version and supporting directories under /var/www/travelman or edit the index.php file in each subdirectory you’ve created and make the change shown in this post on the PIAF Forums. Enjoy!

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

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…

We gratefully acknowledge the contributions of rentpbx.com to the PBX in a Flash Development Team. In addition to hosted accounts to test PBX in a Flash in the hosted environment, rentpbx.com also has contributed technical assistance particularly as it relates to our Google Voice-Asterisk integration efforts. [↩]

The Incredible PBX: Safely Interconnecting Asterisk Servers

Monday, November 15, 2010 / 5 Comments on The Incredible PBX: Safely Interconnecting Asterisk Servers

WOW! What a couple of weeks it has been. The response to Incredible PBX for Asterisk® 1.8 has been, well, incredible. Just last week, SlickDeals and FatWallet introduced over 50,000 bargain hunters to the beauties of Asterisk and Google Voice using Incredible PBX. They joined our regular 50,000 weekly visitors in discovering what may be the best VoIP calling platform on the planet, free or otherwise.

But we’ve also heard from long-time users of PBX in a Flash: "How can we take advantage of this new Google Voice technology without breaking our existing server?" Well, starting today, it’s easy! We’re going to show you how to interconnect as many Asterisk servers as you like using a simple FreePBX tweak to make free calls using your Incredible PBX. To begin, just set up a second server or virtual machine running Incredible PBX 1.8. Then we’ll walk you through interconnecting it with any other Asterisk server that’s running FreePBX. It really is a 5 minute project… once you’ve finished reading this article.

Don’t be intimidated by all of the screen shots shown below. We’re just showing multiple ways of doing the same thing. So you don’t need to use all of them. Once you’ve added one trunk entry on each of your servers and an outbound route on your existing Asterisk server, all of the users on your primary server can instantly begin making free outbound calls through the Google Voice setup on your Incredible PBX. Keep in mind that, at least for now, there is no limit to the number of simultaneous (free) outbound calls you can make within the U.S. and Canada using the Incredible PBX 1.8 platform. And you can interconnect as many Asterisk servers as you like assuming you have the 100kbps VoIP bandwidth to support each simultaneous call.

To get started, follow our last article to get an Incredible PBX 1.8 server set up. As shown in the diagram above, we’re going to assume you’ve got both your new and old Asterisk servers running on the same subnet behind a very secure hardware-based firewall. But this isn’t really required from a technical standpoint. One or more additional servers could be strung all around the globe if that’s your requirement. Or you may wish to take advantage of the incredible deal at RentPBX.com and let them host Incredible PBX 1.8 for you at $15 a month. Just use this special coupon code: BACK10. Then all of your other Asterisk servers can take advantage of today’s free-calling solution. We would hasten to add that, once you’re using the Internet as the transport mechanism for interconnecting servers, we recommend you read and use the secure VPN setup outlined in our VPN in a Flash knol, but the IAX setup outlined below is secure except your voice data is not encrypted. So that’s your call to make.

Today’s Drill. We’re going to show you how to make calls from your existing Asterisk server through The Incredible PBX today. We’ll leave it to you to get things working in the other direction if that is a requirement for your project. First, we’ll create a new trunk on The Incredible PBX, and then we’ll create both a new trunk and a new outbound route on your existing server. We’ll also cover two different interconnection setups. First, we’ll do it using SIP. And then we’ll show you a similar setup using Asterisk’s IAX.

If both servers are sitting on the same private LAN, then the SIP setup is a little easier because the Linux firewall running on Incredible PBX allows SIP traffic to flow freely without any adjustment. It assumes you have added the recommended hardware firewall layer of protection with SIP access to your servers closed off. If one or more of your servers are outside the hardware firewall that is protecting Incredible PBX 1.8, then we recommend the VPN solution referenced above first and the IAX solution outlined here as a second option because the data is unencrypted. Both of these options avoid having to open up any SIP ports on your hardware firewall, and require only a minor adjustment to IPtables, the Linux-based firewall running on The Incredible PBX.

Naming Conventions. To keep things simple, we’re going to refer to the two servers in our example as incredible-pbx and piaf-main where incredible-pbx is your new Incredible PBX 1.8 server that will host the outbound Google Voice calls for users on your piaf-main server. You can obviously adjust these names in any way you like. The only gotcha is that Asterisk attempts to match an incoming call’s username against one of its corresponding trunk names before allowing the call. If there’s no match, the call will fail. So make sure that, if you change the names in the example, do it for both the username and trunk name entries on both servers. Better yet, follow the naming convention in our example, and it just works. 😉

Security Implications. If any of your Asterisk servers allow direct SIP traffic from the Internet, then you need to be extra careful in setting up this interconnectivity since it may allow anyone to attempt to make calls through your Incredible PBX depending upon how your primary server’s dialplan is configured. For example, once a server is interconnected with Incredible PBX, anyone could dial 6789876543@youripaddress and the call might be processed by Google Voice. To avoid this, the simple solution is to password-protect every Outbound Route on your Incredible PBX by adding a Route Password. Or, better yet, don’t expose any of your Asterisk servers to Internet SIP access. Whatever you do, be sure to test making a SIP URI call such as the one shown here once you have all of the pieces in place. Then you’ll know whether you have a security issue or not.

Setting Up Incredible PBX for Interconnecting Servers. Let’s set up a SIP and IAX trunk on your Incredible PBX first. You really don’t need both of these. To repeat, if The Incredible PBX is located on the same private subnet as your other Asterisk server, just use the SIP trunk. If you need access from an Asterisk server outside your private LAN, use the IAX setup. To begin, login to FreePBX using maint and the password you set up with passwd-master. To create a trunk, first choose Setup, Trunks.

To create a SIP trunk, click Add SIP Trunk. For the Trunk Name, enter piaf-main. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.50 with the actual IP address of your piaf-main server. Replace password with a very secure alphanumeric password. Leave the other entries as they are.

To create an IAX trunk, click Add IAX2 Trunk. For the Trunk Name, enter piaf-main. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.50 with the actual IP address of your piaf-main server. Replace password with a very secure alphanumeric password. Leave the other entries as they are.

With either or both trunks, you have the option of tightening up how calls placed from the other server are routed. To force all calls to go out through the Google Voice trunk, just change context=from-internal to context=gvoice. If you want extensions on the other server to be able to call extensions on The Incredible PBX directly, leave the context entry the way it is shown.

While we don’t recommend it, if you’re going to have multiple Asterisk servers connecting to The Incredible PBX to place Google Voice calls and you’re too lazy to create separate trunks to support each server, you can eliminate the IP address checking mechanism in Asterisk by replacing host=192.168.0.50 with insecure=port,invite. The security implications should be obvious.

Setting Up The Other Asterisk Server. There are two steps in setting up any other server that you wish to interconnect with The Incredible PBX. First, you have to create a compatible trunk to handle the calls. Then we’ll add an Outbound Route to send certain calls to Incredible PBX for processing. If you’re using SIP on the Incredible PBX, then you have to use SIP on the other Asterisk server. Same goes for IAX. We’ll set up both a SIP and IAX trunk on the PIAF main server just to show you what the entries should look like. And, to repeat, you really don’t need both of these. If your other Asterisk server is located on the same private subnet as Incredible PBX, use the SIP trunk. If you need access to Incredible PBX from elsewhere, use the IAX setup. To begin, login to FreePBX on your other PIAF server using maint and the password you set up with passwd-master. To create a trunk, first choose Setup, Trunks.

To create a SIP trunk, click Add SIP Trunk. For the Trunk Name, enter incredible-pbx. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.212 with the actual IP address of your incredible-pbx server. Replace password with the same secure alphanumeric password you used on the Incredible PBX SIP trunk to which you will be connecting. Leave the other entries as they are.

To create an IAX trunk, click Add IAX2 Trunk. For the Trunk Name, enter incredible-pbx. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.212 with the actual IP address of your incredible-pbx server. Replace password with the same secure alphanumeric password you used on the Incredible PBX IAX trunk to which you will be connecting. Leave the other entries as they are.

You’ll notice in the Dial Rules, we’ve used 48 (which is GV on a phone) as the prefix to be dialed on your other Asterisk server to route calls out through Google Voice on The Incredible PBX. So, to place a call from your other Asterisk server via Google Voice, a user would dial something like this: 48-678-987-6543. Before the call leaves the Asterisk server, the 48 prefix will be stripped off. You can make this prefix anything you’d like. Just be sure to use the same prefix when you set up the Outbound Route in the next step.

Adding an Outbound Route. The final configuration step is to add a new outbound route on your other Asterisk server to actually send calls to The Incredible PBX. As noted, we use a dialing prefix so that we can identify the calls to be sent. Create a new route called GoogleVoice and make your entries look like the following if you’re using IAX. If you’re using SIP, just change Trunk Sequence 0 to SIP/incredible-pbx. Click Submit Change and reload FreePBX when prompted.

Keep in mind that FreePBX processes Outbound Routes in top down order, and the first matching route is the only route that is used to place the call even if the call fails. So the trick here is to move your new GoogleVoice route up the list so that it’s at least above the default calling route (which is a route with no specified dial patterns to match) and any other routes consisting of 12 or 13-digit dial strings which might match our GoogleVoice dial patterns.

IAX Firewall Adjustments. If you’re using the IAX method above, you’ll need to adjust the IPtables firewall rules on Incredible PBX to allow communications with your other Asterisk server. If your other Asterisk server is PBX in a Flash, you may need to add a similar entry in the IPtables rules on that machine as well. In addition, you’ll need to map UDP 4569 on your hardware-based firewall to the private IP address of your Asterisk server. Otherwise, calls will never make it past your firewall.

On each server, edit /etc/sysconfig/iptables and add an entry with the IP address of the other server with which you’ll be communicating. If your Incredible PBX is on a different public network than your other server, we’d need to add an entry near the end of the file and above COMMIT allowing IAX communications with the public (not private!) IP address of the piaf-main server assuming that server is outside the LAN, e.g. something like this:

-A INPUT -p udp -m udp -s 222.68.100.150 –dport 4569 -j ACCEPT

If you’re using IAX and both servers are on the same private subnet or interconnected private subnets, then the entry might look like this:

-A INPUT -p udp -m udp -s 192.168.0.50 –dport 4569 -j ACCEPT

Once you’ve saved your change, restart the firewall: service iptables restart

Testing Things Out. Now you’re ready to place a test call. Pick up an extension on your piaf-main system and dial 48-800-322-7300. You’ll be greeted by American Airlines courtesy of Google Voice. The CallerID of your outbound calls will be your Google Voice number regardless of the extension or server from which the call originates. Enjoy!

Originally published: Monday, November 15, 2010

Introducing The Incredible PBX for Asterisk 1.8

Adding Skype to The Incredible PBX

Adding Incredible Backup… and Restore to The Incredible PBX

Adding Remotes, Preserving Security with The Incredible PBX

Remote Phone Meets Travelin’ Man with The Incredible PBX

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

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…

2010 Bargain of the Year: Nortel 1535 Color SIP Videophone

Tuesday, October 19, 2010 / 51 Comments on 2010 Bargain of the Year: Nortel 1535 Color SIP Videophone

We try not to get overly excited by new discoveries, but once in a while there comes along a VoIP deal that probably never will be repeated. Now’s the time. Here are a Baker’s Dozen reasons why you should buy a boatload of these Nortel IP 1535 phones before they’re all gone. Just make a bid of ~$60. We’ve given you a hint below on the going rate. 😉

WARNING: There are reports that some of the phones from various merchants do NOT include WiFi even though the ad may say otherwise. If you need WiFi, be sure to carefully read the merchant’s ad AND verify that the phone you are ordering has WiFi before purchasing.

Nortel’s top-of-the-line $700 phone can be had for about $60
SIP-compatible and works with Asterisk® and sip2sip.info among others
H.263-compatible color videophone works flawlessly
Wired and 802.11 b/g WiFi is easily configured
Supports both U.S. and European power cords out of the box
Speakerphone rivals the best speakerphones on the market
Integrated apps include browser, email, calendar
Music and video storage supported using SD/MMC cards
Configurable voicemail button for easy access to any mailbox
Language support for English, French, Spanish, German and more
Tons of integrated multimedia capabilities
Robust STUN and proxy support so they work from anywhere
One year warranty on the phones from the eBay merchant

Connectivity Options. Once you have your phones, there are almost limitless SIP connectivity options including direct connections to many of our favorite providers: Vitelity, voip.ms, Future Nine, and Axvoice. But today we want to address two other connectivity options: sip2sip.info or as an Asterisk extension on your PBX in a Flash system, both of which give you color videoconferencing out of the box.

Using sip2sip.info with the Nortel 1535. If you haven’t discovered sip2sip.info, it’s one of the few VoIP freebies left in the universe. By simply providing your name and email address, sip2sip.info will give you a free SIP a URI that lets anyone on the planet call you via SIP at no cost. In addition, all calls to numbers registered with ENUM are free as well. For example, to call numbers in the U.S. listed with e164.org, just dial 001NXXNXXXXXX. You can talk as long and as often as you like. The call can be pure audio, or it can be an H.263 video call. It’s simple to set up and use. And, once you have your phone configured with sip2sip.info, it’s incredibly easy to add a free DID from IPkall and then a free local DID from Google Voice. Then, presto, you have a local phone number for inbound calls that will never cost you a dime. If you make most of your outbound calls from a cellphone, then this is a perfect solution for a free home telephone number where anyone can reach you. And it includes a free voicemail account that will deliver the voicemails to your registered email address whenever you miss a call. We actually travel with one of these phones preconfigured with a local number in our favorite towns. When we go to a different place, it’s easy to change the local phone number. Update: You also can obtain a free SIP URI from GetOnSIP.

There’s only one trick to the sip2sip.info setup. Once your credentials are emailed to you, log into your account and change your password to a very secure but all-numeric password.

Using Asterisk with the Nortel 1535. We have a personal preference for Asterisk, and it’s a perfect fit with these phones. Just add these entries to sip_general_custom.conf in /etc/asterisk, and video support comes to life in all versions of PBX in a Flash once you restart Asterisk:

rtptimeout=120
videosupport=yes
allow=h263

Then you’re ready to set up your extensions to support the Nortel 1535. Here are the settings we use, and they work equally well with the X-Lite 4 client if you’d like to try some test video calls on your server:

dtmfmode=rfc2833
canreinvite=yes
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
disallow=all
allow=h263,ulaw,gsm

Configuring the Nortel 1535. All of the manuals for these phones still can be downloaded from Nortel’s web site. With the exception of the early phones which were configured for Turkey, here is the setup that works for us with sip2sip.info and Asterisk. Our special thanks to the dozens of gurus on the PBX in a Flash Forum who assisted with sorting all of this out. If you get stumped on any of this, the thread link provided has loads of additional information.

The two buttons at the top of the phone do most of the heavy lifting. The left one is the equivalent of the Enter key on a keyboard. The right one is the Back key. The other two keys of importance are * and #. * is used to enter special characters such as the period, slash, etc. # is used to change the keyboard type: ABC, Abc, 123, etc. Be sure you always have the correct keyboard type for the type of data you are entering. Pressing the Green button twice redials the last number called. The function key to the right of the number 3 connects you to voicemail. The function key to the right of the number 6 accesses the web browser.

Before you can configure the phone, you have to log in as Admin (Menu, Settings, System Settings, Admin, Login). The password is 1234. Then back out one level and set your Date/Time preferences. The most important one is to enable Network Time. For the Server Setting, enter time.nist.gov for a reliable NTP server. Then back out a level and choose Enable.

You’ve got to set up network connectivity before the phones will work obviously. They come preconfigured for a wired connection with DHCP support. That’s a good way to begin. Once everything is working reliably, you can switch to WiFi if desired. The only trick to WiFi is that you need to set your WiFi Type (Menu, Settings, Profile, WiFi, Settings, Wireless Settings, Authentication, Type) and then the WiFi Password for the chosen type before choosing your WiFi network (Menu, Settings, Profile, WiFi, Settings, Wireless Settings, WiFi Scanning). Once you have those set up, back out one level and choose Apply. Then back out one more level and choose Enable. You’ll be prompted to confirm you wish to restart the WiFi network. Then you’re all set.

Now you’re ready to configure your VoIP settings (Menu, Settings, VoIP Settings). Start with the domain of your server: sip2sip.info or the FQDN of your Asterisk server (Menu, Settings, VoIP Settings, Misc., Domain Name). While still in Misc., adjust the Codec Priority for video (Menu, Settings, VoIP Settings, Misc., Codec priority, Video). Choose First and change it to None. Choose Second and change it to H.264. Then choose First again and change it to H.263. Asterisk only supports H.263 so it has to be the first priority, or video won’t work. Then back out until the top left of the screen shows VoIP Settings. Choose User Information and enter your username for Username, Display Name, and Authentication name. For Asterisk, it’s your extension number. For sip2sip.info, it’s your 10-digit number beginning with 223. Enter your account password for Authentication pwd. Back out to VoIP Settings and enter the IP address of your server for Proxy, Proxy Address. For sip2sip.info, it’s 81.23.228.129. For Asterisk, it’s the public IP address of your server. While still in Proxy, choose STUN. For STUN Server IP Address, enter 75.101.138.128. Then Enable the STUN Server. Finally, back out to VoIP Settings again and choose Registration. Set the Expiry Timer to 3600. Then choose Register to connect your phone to your desired server. Done!

Using sip2sip.info with Asterisk. We were so impressed with the simplicity and functionality of sip2sip.info that we decided to also set up a sip2sip.info trunk on our Asterisk server. This is a very secure way to enable a SIP URI on your Asterisk server without exposing your server to SIP vulnerability. The only additional step with PBX in a Flash is to lock down external SIP access to the IP address of sip2sip.info. For setup instructions, see this thread on the PBX in a Flash Forums.

Configuring Voicemail Access. It’s easy to configure these phones to access any existing voicemail system. The only trick is that the number to call for voicemail access must be all numeric. On Asterisk systems, this means *98 won’t work! So, in FreePBX, first set up a Misc. Destination called Voicemail-Read and use *98 as the Dial String. Then set up a Misc. Application called VoiceMailRead and enter 86245 as the Feature Code. Then choose Misc Destination: Voicemail-Read as the Destination.

On the phone, choose Menu, Settings, VoIP Settings, Misc., Voice Mail, Voicemail Number and enter 86245. You can leave the Mailbox ID and password blank on Asterisk-based systems, and you’ll be prompted for them. Or you can fill in either the mailbox number or both the mailbox number and password, and your entries will be passed to Asterisk to access the desired voicemail box.

To access Voicemail from the phone, press the function key just to the right of the number 3 on the phone.

Using the Nortel 1535 Browser. While it’s not the best browser on the planet, these Nortel phones do have a decent web browser that can be used to retrieve current content such as news, weather, and sports scores. To set up a web link, choose Menu, Services, Web Browser, and choose one of the four links. Here are a couple entries to get you started. Others can be found in this thread on the PBX in a Flash Forums. Remember to use the Top Left function key as the Enter key in browser links! HINT: While in one of your four preconfigured web sites, if you press the Right Button just above the directional arrow keys, you can navigate to additional web sites.

mundy.org/news.php – Latest Yahoo! News
google.com/m – Google Mobile

To access the Browser, press the function key just to the right of the number 6 on the phone.

Accessing Email on the Nortel 1535. Both POP3 and IMAP email access are supported on the phone. And a number of boilerplate email messages already are preconfigured for sending using your chosen email provider. You can set up additional ones using the Template option. To set up email, go to Setup, Messages, Account Settings.

Nortel 1535 Organizer. These phones also include a very capable Address Book and Calendar. Entries can be imported using a standard SD/MMC card. We’ll leave the rest for you to sort out. Or take the guess work out of the experiment and read Nortel’s excellent documentation. Enjoy!

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

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…

It’s TeleYapper 5.0: The Ultimate RoboDialer for Asterisk

Sunday, October 10, 2010 / 21 Comments on It’s TeleYapper 5.0: The Ultimate RoboDialer for Asterisk

We don't normally take a month off at Nerd Vittles which should tell you something about today's 10/10/10 column. We're pleased to introduce TeleYapper 5.0, a completely rewritten, Asterisk® 1.4 and 1.6.2-compatible version of our telephone broadcasting service.¹ Using Cepstral text-to-speech, TeleYapper 5.0 brings individualized, text-based messaging and customized reminders coupled with the ability to capture recorded responses from every call.

WARNING: Because of changes in Cepstral, this application now requires an additional $200 license from Cepstral. We no longer recommend Cepstral for obvious reasons and will have a comparable system using Google's new Speech-to-Text application soon. Our apologies.

As part of the message delivery process, you now can customize and capture any one of four different responses from those that are called. And TeleYapper 5.0 will email you a CSV and/or XML file with the RoboResponse™ results when the calling process is completed including a list of failed calls and calls that were answered by an answering machine. In addition, you can have TeleYapper email certain call results to various individuals as the calls are processed if your requirements demand it.

For those with multiple outbound trunks, TeleYapper 5.0 supports simultaneous calls using multiple trunks. And now there are significant enhancements that detect answering machines and real people. This lets you deliver customized messages depending upon whether an actual human answers the phone.

Version 5 has been tested extensively with the Gold, Silver, and Bronze editions of PBX in a Flash 1.7.5.5, which provides support for the latest and greatest versions of Asterisk 1.4 and 1.6.2. And it should work well with other Asterisk aggregations with MySQL, Cepstral TTS support, and FreePBX 2.5 or later.

Overview. For those that have never used TeleYapper, here's a quick summary of how the new version works. It's an automated message broadcasting service commonly known as a call blasting or phone blasting system. In addition to loads of creepy uses, phone blasting has legitimate purposes as well. TeleYapper is licensed in several different ways for the following purposes: prerecorded phone messages for neighborhood association announcements, medical appointment reminders, school closings, tornado alerts, little league practices, municipal government reminders. It's free to use for non-profit, civic, and non-political purposes provided you don't solicit money or seek to sway someone's opinion or encourage a particular vote on an issue or candidate. All other uses require a commercial license. For commercial, political, and medical applications, please review our licensing terms below.

How it Works. Step #1 is to create a CSV or XML export from your favorite database application with the information that will be used to send the messages or reminders. This could be as simple as a list of phone numbers or as complex as a listing of doctors and patients with the dates, times, and places of their next medical appointments together with special patient instructions for activity preceding their visit, e.g. "Please remember to start flossing a month before your next dental appointment."

Step #2 is to create a config file with the robodial settings as well as the text which will actually be spoken during each customized call. If you remember form letters from your word processing days, TeleYapper's config file offers the same flexibility. A message can be as simple as "Take cover immediately. A tornado has been spotted at the end of your street." Or it could be a medical appointment reminder such as the following:

Hi. This is Allison from Charleston Family Clinic calling to confirm Jan's appointment with Doctor Quack on Tuesday, October 5th, 2010, at 10:30 a.m. in our Charleston office. Please remember not to eat or drink anything after midnight on the night before your scheduled appointment.

To confirm your appointment, press 1. To reschedule your appointment, press 2. To cancel your appointment, press 3. If we have reached you in error or if you do not wish to receive further automated medical appointment reminders, press 4. To hear this message again, please press 5 now.

And you can create a separate message which would be delivered in the event an answering machine takes the call:

Hi. This is Allison from Charleston Family Clinic calling to confirm Jan's appointment with Doctor Quack on Tuesday, October 5th, 2010, at 10:30 a.m. in our Charleston office. Please remember not to eat or drink anything after midnight on the night before your scheduled appointment.

If you need to change or cancel your appointment or if we have reached you in error, please call our office at your earliest convenience. The number is 800-123-4567. Goodbye.

Step #3 is to use your web browser to access a password-protected web page that will let you upload your CSV or XML data and your config file to kick off the dialing spree. Once the files have been uploaded, everything else is automatic.

Step #4 is to sit back and relax while TeleYapper executes your instructions and calling list. When the calling has been completed, the email address in your config file will be sent both CSV and XML reports of the results of all the calls. Either of these reports is suitable for import and manipulation using most spreadsheet applications.

Status Codes. Every call that is processed gets a status code entry whether the call is successful or fails. A status code of 0 means a call failed to both phone numbers provided for a particular callee. The second phone number is entirely optional. A status code of 5 means the call was answered but no response was provided by the called party. This typically would mean the call was picked up by an answering machine although it could mean Granny answered the call using a rotary dial phone. 🙄 Status codes of 1 through 4 have whatever meaning you choose to assign to each option when setting up a configuration for a particular calling campaign.

Legalese. TeleYapper 5.0 is free for use by non-profit, civic, and non-political organizations provided you absolve us from all financial and other responsibility in conjunction with your use of the software. Non-profit use further requires that no financial benefit be derived from the substance of the calls. Simply stated, your Little League team can use the software at no cost to remind kids to attend practice, but it cannot be used to solicit charitable contributions or to sell doughnuts without obtaining a commercial license.

By using this software, you also agree to strictly comply with federal and state regulations including 16 C.F.R. Part 310. In addition, you agree to assume all risks associated with use of the software. NO WARRANTIES EXPRESS OR IMPLIED INCLUDING ITS FITNESS FOR USE OR MERCHANTABILITY ARE PROVIDED WITH THIS SOFTWARE.

WARNING: With certain limited exceptions, most robocalling now requires prior written approval from those being called. See this link for a summary of the federal requirements. Be advised that improper use of this software may subject the user to penalties of up to $16,000 per call plus monetary damages to injured consumers.

Licensing. You are licensed to use this software under certain conditions. You do not own it. We do, and we also own the copyright. It is licensed for use under the terms of the Creative Commons Attribution Non-Commercial license. A Plain English summary is available here. We've done this primarily to do our part to stamp out the telemarketing creeps of the world. Those wishing to use TeleYapper for commercial or political purposes must first request and then purchase a commercial license after outlining your proposed terms of use. Telemarketers need not apply! For doctors, lawyers, and others falling outside the scope of our free license who wish to obtain a commercial use license, please contact us for pricing and details. Be sure to summarize your intended use in your request together with a sufficient factual summary to demonstrate that your use is in compliance with 16 C.F.R. Part 310. Please also indicate whether you will require assistance with installation and setup.

Prerequisites. As mentioned, you'll need a Linux-based Asterisk aggregation such as PBX in a Flash to use TeleYapper 5.0. This means you need a system with Asterisk 1.4 or 1.6 as well as FreePBX 2.5 or higher. For quality reasons, we strongly recommend you purchase a commercial Cepstral text-to-speech license for your server. While Flite would technically work, most folks don't respond well to calls from Egor so we have customized the code for use solely with Cepstral. You'll find Cepstral installation instructions in this Nerd Vittles article. The TeleYapper 5.0 code also relies heavily on Apache and PHP, both of which are included in every PBX in a Flash system.

Installing Cepstral. Cepstral installation is not the simplest application to get working with Asterisk so here are the commands for those running 32-bit systems with Asterisk 1.4 or 1.6.2. For details on purchasing and registering Cepstral (and a discount) and for 64-bit installs, read our previous article including the comments.

For Asterisk 1.4 systems running under 32-bit CentOS, log into your server as root and issue the following commands accepting the Cepstral defaults. Be sure to create the Cepstral directory when prompted!
cd /root wget http://nerd.bz/bnTVjX 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* cd app_swift-1.4.2 make make install 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" ls /opt/swift/voices swift --reg-voice

For Asterisk 1.6.2 systems running under 32-bit CentOS, log into your server as root and issue the following commands accepting the Cepstral defaults. Be sure to create the Cepstral directory when prompted!
cd /root wget http://nerd.bz/bnTVjX 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-1.6.2.tar.gz tar -zxvf app_swift* cd app_swift-1.6.2 make make install 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" ls /opt/swift/voices swift --reg-voice

Installing TeleYapper 5.0 The real beauty of PBX in a Flash as an Asterisk platform is demonstrated by the ease with which you can install new applications such as this one. The drill is very simple. You download an install script, make it executable, and run it. Less than a minute later, the TeleYapper install is done. Here are the commands to execute to install TeleYapper 5.0 after logging into your PBX in a Flash system as root. On other systems, you are well advised to carefully review the install script and tailor it to meet the individual requirements of the platform on which you are installing it.
cd /root wget http://bestof.nerdvittles.com/applications/teleyapper5/teleyapper5.pbx chmod +x teleyapper5.pbx ./teleyapper5.pbx

The TeleYapper Database. We use the MySQL database management system to manage the list of callees for TeleYapper to dial. It can handle a database of almost any size and generally stands up well in performance comparisons with Oracle. So you're covered on the database front. For most users, you never should need to access the MySQL database directly. TeleYapper 5.0 handles the importing of CSV or XML files for processing, manages the call queue, and processes and emails CSV and/or XML-formatted reports to you when the calls are completed.

The install script creates the MySQL database to support TeleYapper 5.0. Should you need or want to manage the database directly, the easiest tool to use is phpMyAdmin which is accessible through the Tools tab in FreePBX on PBX in a Flash systems. You'll need to login as maint with your maint password to access phpMyAdmin. After phpMyAdmin loads, click on the reminders database in the left column. Then click the reminders table entry in the left column to open the file. Unless you really, really know what you are doing and appreciate how much coding will be required to support new or different fields in the reminders file, don't improve it.

Here's the layout of the MySQL database table for TeleYapper 5.0:

id - System generated record ID
acctno - Account Number (12 alphanumeric characters)
provider - Provider Name (30 alphanumeric characters)
recipient - Recipient Name (30 alphanumeric characters)
apptdt - Appointment Date (MM/DD/YY format)
appttime - Appointment Time (HHMM format using 24-hr clock)
apptplace - Appointment Location (30 alphanumeric characters)
instructions - Free-form text (65535 alphanumeric characters)
phone1 - Primary Phone (NNN-NNN-NNNN or NNNNNNNNNN)
phone2 - Alternate Phone (NNN-NNN-NNNN or NNNNNNNNNN)
status - Status: 0=failedcall 5=ansmachine 1,2,3,4=user-defined
failedcalls - System Generated Number of Failed Calls

Tweaking PHP for TeleYapper. Depending upon your PHP setup and the number of calls you plan to process, you may need to adjust the default PHP resource settings on your server. The main reason is because TeleYapper generates a custom sound file for every call to be processed before the calling ever starts. If you plan to make thousands of calls, this can take some time. The PHP settings are stored in /etc/php.ini. You must log in as root and restart Apache after making changes to these settings: service httpd restart. The settings that matter are the following:

max_execution_time = 30 (we recommend 900 which is 15 minutes to process)
max_input_time = 60 (we recommend 300 which is 5 minutes to upload a file)
memory_limit = 100M (OK as is)

post_max_size = 8M (we recommend 100 megabytes which should be ample)

file_uploads = On (OK as is on most systems)
upload_max_filesize = 100M (we recommend 100 megabytes which should be ample)

Tweaking Crontab. TeleYapper relies upon a cron job to kick off its calling sprees so you'll need the following entry in your /etc/crontab file unless you used the install script which inserts it automatically:

* * * * * root /var/www/html/appt-reminders/gen-reminders.php > /dev/null 2>&1

Formatting CSV Data For Import. You don't necessarily need an external database in order to use TeleYapper 5.0 although it is designed to support almost any database or spreadsheet application in the marketplace so long as it can export data in CSV or XML format. A CSV (comma-separated values) or XML file is the middleware that makes everything work. Each line in a CSV file represents an entry to be processed by TeleYapper 5.0 when the CSV file is uploaded. Each item in a line is called a field. Every field begins and ends with double-quotes, and fields are separated from each other with commas. Do NOT include any quotation marks in your actual text, or you'll get a disaster. All fields are required, by the way, but only the Phone1 field must have an actual entry. The remaining fields may each consist of nothing more than a pair of double-quotes. Note also that the id, status, and failedcalls fields (shown in red below) must consist of a pair of double-quotes and nothing more. Here's the actual CSV format which must be used, and all of the data must appear on the same line so disregard the WordPress formatting below:

"id","acctno","provider","recipient","apptdt","appttime","apptplace","instructions","phone1","phone2",
"status","failedcalls"

Here's what the CSV entry used for our sample medical reminder shown near the top of this article would look like. We've excluded the special instructions and Phone2 entries below only to simplify the display because of constraints inherent in our blog formatting:

"","12345","Quack","Jan","10/05/10","1030","Charleston","","4049876543","","",""

The XML Alternative. If you'd prefer to upload XML file templates for your calls instead of CSV data, a sample XML file is included in the distribution to show you the proper formatting. Here's a sample entry that matches the CSV data above:

<reminders>
   
    <reminders>
       <id></id>
       <acctno>12345</acctno>
       <provider>Quack</provider>
       <recipient>Jan</recipient>
       <apptdt>10/05/10</apptdt>
       <appttime>1030</appttime>
       <apptplace>Charleston</apptplace>
       <instructions></instructions>
       <phone1>4049876543</phone1>
       <phone2></phone2>
       <status></status>
       <failedcalls></failedcalls>
    </reminders>
</reminders>

Direct Uploading with SAMBA. If you've activated SAMBA on your Asterisk server, you can upload TeleYapper files for processing directly. Be sure to name your CSV or XML file as reminders.csv or reminders.xml. And name your config file: config.php. Copy the files to the /var/www/html/appt-reminders/upload directory on your Asterisk server. That's all there is to it. If you need hints on SAMBA installation, see our Best of Nerd Vittles tutorial. Pay particular attention to the sections on Security Considerations and Firewall Settings. Before using the SAMBA, be sure to upload some test CSV/XML files using the web interface. There is no error checking when you use the SAMBA option!

Configuring TeleYapper 5.0 Calling Scripts. Now let's address how we transform a CSV or XML entry such as the ones shown above into a personalized phone call to Jan, the actual patient in our example. Every TeleYapper session can have an individual configuration file associated with it. If none is specified, then a default configuration is used. In this way, you can customize call procedures and calling scripts for different tasks. The easiest approach is to always upload a config file with your CSV or XML data file. Then you won't get unexpected results when the calling begins.

HINT: It's a very good idea to create a sample upload with your own phone number and some sample configuration data to test things out before you start calling thousands of clients.

A default configuration file (config.default.php) as well as sample CSV and XML templates (reminders.csv and reminders.xml) come with TeleYapper 5.0 and can be found on your Asterisk server in the /var/www/html/appt-reminders directory. Make a copy of them, and move the copies to your Mac or PC. Then, using TextEdit or Notepad, open the files and have a look. Before addressing other configuration options in config.php, let's tackle the setup procedure for calling scripts.

The actual boilerplate message to be delivered to the called party is stored in $msg. Notice that you can substitute data out of your database in the boilerplate template by enclosing any desired fields in braces. Just make sure the fieldname exactly matches one of the fields in the reminders database. So our entry for the sample call above would look like this:

$msg="Hi: This is Allison from Charleston Family Clinic calling to confirm an appointment for {recipient}, with Doctor {provider}, on {apptdt}, at {appttime}, in our {apptplace} office. {instructions}";

Just a comment that, for those with large data processing systems, you may find it more convenient to generate the actual text for each reminder on your mega-machine. In this case, all of the data (up to 65,535 characters) could be loaded into the instructions field for each callee. So each upload record might consist of nothing more than phone numbers and instructions. In this scenario, the $msg entry in config.php would look like this: $msg="{instructions}";

The key press choices that are provided to the called party are configured using the $options field which would look like this for our example:

$options = "To confirm your appointment, please press 1. To reschedule your appointment, press 2. To cancel your appointment, press 3. If we have reached you in error or if you do not wish to receive appointment reminders, press 4. To hear this message again, please press 5 now.";

Don't confuse the 5 option which is automatically included in the TeleYapper dialplan code with status code 5 which means an answering machine picked up a call. Status code 5 is system-generated and is not stored based upon a callee choosing to listen to a recorded message more than once. The two 5's are not the same even though options 1-4 are actually used to define what the first four status codes mean on your system.

As we mentioned, the system has the smarts to usually figure out if an answering machine took the call. When it detects this, the $ansmach message is played instead of $options. A sample entry might look like this:

$ansmach = "If you need to cancel or reschedule this appointment, if we reached you in error, or if you do not wish to receive appointment reminders in the future, please call 777-123-4567 at your earliest convenience. Thank you for your assistance. Goodbye.";

Finally, for each of the four choices (1 through 4), there is a response message which is played if the callee chooses that option. Here's a sample template to get you started:

$chose1 = "Thank you for making Charleston Family Clinic your medical home. Your appointment has been confirmed. Goodbye.";
$chose2 = "Thank you. A representative will be calling you to reschedule your appointment. Goodbye.";
$chose3 = "Thank you for making Charleston Family Clinic your medical home. Your appointment has been cancelled. Goodbye.";
$chose4 = "Thank you. We will update our systems and apologize for the call. Goodbye.";

Thus, when a callee responds to the boilerplate call by pressing 1, $chose1 is played in response. If an email address has been entered for $chose1email, then a copy of the log entry for that call is sent to the specified email address using the customized email subjects (shown below) in addition to being placed in the master call log. The same process occurs when the other options are chosen. Particularly with medical appointment cancellations, it may be important to receive immediate notification when an appointment is canceled or a patient requests a change in scheduling. So the software includes the flexibility to generate instant emails to various email addresses depending upon which option is pressed. As noted, the optional instant emails will be generated using the email subjects entered for the following fields in your customized configuration file:

$chose1subj = "APPOINTMENT NOTIFICATION CONFIRMED BY PHONE";
$chose2subj = "APPOINTMENT RESCHEDULING REQUEST BY PHONE";
$chose3subj = "APPOINTMENT CANCELLATION REQUEST BY PHONE";
$chose4subj = "APPOINTMENT SCHEDULING ERROR REPORTED BY PHONE";
$chose5subj = "APPOINTMENT NOTIFICATION LEFT ON ANSWERING MACHINE";

Uploading Data & Config Files to TeleYapper. Simple web pages are used to upload CSV and XML data with config files to TeleYapper 5.0. WARNING: These web pages have NOT been sanitized for use on the Internet. They are designed for use on your local area network behind a secure firewall. On PBX in a Flash systems, the web pages are password-protected and require a valid user account login for access. This will NOT be the case on other Asterisk aggregations without tweaking your Apache configuration. Sample entries can be found in teleyapper.conf in the /var/www/html/appt-reminders directory. On PBX in a Flash systems, you can log in using maint, wwwadmin, or meetme accounts. Or you can create an additional account to use with TeleYapper 5.0:

htpasswd /usr/local/apache/passwd/wwwpasswd teleyapper

There are separate web pages depending upon whether you wish to upload CSV or XML data. For CSV data, the web address is http://ipaddress/appt-reminders/uploadcsv/. For XML data, the web address is http://ipaddress/appt-reminders/uploadxml/. Substitute the private IP address of your Asterisk server for ipaddress. Here's a sample of the CSV web form. You can, of course, substitute your own logo on the right if desired.

Other TeleYapper 5.0 Config Options. In addition to the boilerplate text for TeleYapper calls, there are a number of other settings which can be adjusted to meet your individual requirements.

The database settings should never need adjusting so just leave them alone. They look like this:

$db="reminders";
$fi="reminders";
$dbuser="root";
$dbpass="passw0rd";

You can manually set a starting and ending time to begin and end the calling sequence for a particular upload. Never set these in the default configuration! Only set them in a config file to be uploaded. If the entries are blank, calls will commence shortly after the upload completes and will end when all of the entries have been processed. Note that there is no current flexibility to schedule individual calls based upon the time of the appointment. This typically would be handled by selecting particular records for processing in your primary database. For example, for medical appointments, you would select records in which an appointment is scheduled for tomorrow and then upload the list to TeleYapper which would place the calls today. We probably will expand this functionality down the road, but it's not there yet. So it's up to you to upload call lists which basically are ripe for calling now.

If you wish to use the $startcalls and $endcalls features in your custom config files, the syntax should look like this: YYYYMMDD,HHMM where YYYY is a 4-digit year, MM is a 2-digit month, DD is a 2-digit day of the month, HH is the 2-digit hour based upon a 24-hour clock (aka Military Time), and MM is the 2-digit minute. Note that calls will not end precisely at the $endcalls time. Any existing calls already in process will be completed including redials and calls to an alternate $phone2 number. This process can take up to 10 minutes to complete.

CAUTION: Be very careful using the $startcalls option! Nothing precludes your scheduling a thousand reminder calls to kick off at 0200 which is 2 a.m. Not really a good thing if job security matters to you.

To restart the calling process on the following day, log into your server as root and switch to the /var/www/html/appt-reminders directory. Then edit config.php and adjust the $startcalls and $endcalls for the remaining calls. Then run: ./gen-calls.php. Any existing database entry with a status=0 will be called when the calling process resumes. You can monitor the calling process by running: ./showcalls.sh. Press Ctrl-C to terminate the call display. It usually takes a minute or two for the first call to be placed.

$callerid is used to set the CallerID of outbound calls if your telephony provider supports it.

$trunk is used to set the outbound dialing trunk for calls. The default works for most purposes.

$channel is used to set the outbound dialing channel for calls. The default works for most purposes.

$maxcalls and $spacing are used to set the number of simultaneous calls and spacing between calls respectively. Be very careful with these settings. You must have sufficient outbound trunks to handle the number of simultaneous calls you schedule with $maxcalls, or you will get circuit busy conditions which are recorded as calls to busy numbers. Keep in mind that TeleYapper tries every call twice with 2 minutes of separation. So, if you only have two outbound trunks, don't set $maxcalls above 1, or you will get trunk busy conditions whenever original calls to an individual fail, i.e. line busy or no answer situations. In addition, remember that TeleYapper 5.0 supports a second phone number for each called party. These are triggered whenever the original two calls to the primary number fail and must also be considered in setting $maxcalls properly. If your logs show a disproportionate number of failed calls (status=0), this may be a tell-tale sign of trunk busy conditions.

$waittime is the number of seconds a call to any given number will ring. 45 seconds is about 7 rings.

$email is the email address that will be used to send the logs at the completion of the calling process. $chose1email through $chose5email are the optional email addresses if you want instantaneous feedback on certain types of status results. This means you get an immediate email if a certain call results in a certain status code. Leave the ones blank for $status conditions on which you want no immediate feedback and simply wait for the logs to arrive.

$csvreport and $xmlreport are used to set which type of completion report you wish to receive. If you want both of them, set them both to 1. Otherwise, set the one you don't want to 0.

The Old Fashioned Way. For those of you that preferred the older method of entering data directly into MySQL, you still can use phpMyAdmin or some other front-end tool to enter the data directly into the reminders.reminders table. Just leave the id field blank since it automatically gets generated by MySQL. And either leave the status and failedcalls fields blank or set them to 0. They also are system-generated. Once you have your data in place, log into your server as root, and...

cd /var/www/html/appt-reminders
Configure config.php for your calling campaign
Run ./gen-mysql.php to kick off TeleYapper 5.0

In Closing... Finally, let us issue our usual tinkerer's warning. Don't delete anything from the /var/www/html/appt-reminders directory tree. Just because you don't know its function doesn't mean it doesn't have one. Aside from that, the documentation above should get you started today. Be advised that TeleYapper 5.0 still is a work in progress. So check back every week or so for new comments on this article to see what's been changed, added, or fixed since you originally downloaded the application. Enjoy!

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

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...

Special thanks to my dear wife, Mary, who did much of the system design work for this project, and to Community Health Centers of Florida for underwriting some of the design and development costs. [↩]

TweedleD Back From the Dead Using Twitter OAuth

Tuesday, August 31, 2010

Twitter Direct Messages and SMS Instant Messages are great ways to send yourself important notes that you want to keep track of… privately. Today, we’ll restore TweedleD to the living and let you harness the power of Twitter and SMS to announce every call on your Asterisk® server with the name and number of the caller as well as the DID of the incoming call.

If you’re one of the ~~gazillion~~ handful of folks using one of our Twitter applications, then you awakened to an unpleasant surprise earlier this morning. Neither Tweet2Dial nor TweedleD would connect to Twitter! That meant you could no longer use Twitter to place Google Voice calls, to send Google Voice SMS messages, or to manage your Asterisk server. Nor could you send yourself direct messages when new incoming calls hit your Asterisk server. Unfortunately (for us), the Twitter folks have discontinued use of basic authentication to log into and use Twitter’s API. Instead, you now must use the more secure (and more complicated) OAuth authentication mechanism. Actually, this is a good thing in the grand scheme of things because it means you no longer have to give out your actual Twitter account name and password to anyone. So let’s move on to how to put Humpty back together again.

We’ve actually been wrestling with this a good long while. The basic problem is generating the necessary new credentials for use on servers that are tucked safely behind hardware-based firewalls. Without getting too deep in the weeds, here’s the drill. To use OAuth, you basically need two sets of credentials. First, you need your own (not ours!) application-specific credentials known in the Twitterverse as your consumer key and consumer secret. These are generated by Twitter and normally hide inside your application. They are never made public. You also need an access_token and access_secret. But these can only be obtained once Twitter generates a request_token that is passed back to your application. There’s also a request_token_secret that is used by Twitter to verify that you are who you say you are before the access_token and access_secret are provided. If this sounds complicated, you’d be right. Now add the fact that our particular Twitter apps are sitting behind a firewall on your server, and you can begin to appreciate why OAuth complicates things with apps running on private networks.

So here’s the deal. To use the new OAuth-compatible versions of Tweet2Dial and TweedleD, you’ll need four pieces of information for each of these applications: a consumer_key, consumer_secret, access_token, and access_token_secret. Because the two applications typically use two different Twitter accounts, you cannot interchange these four pieces.

The traditional procedure for getting these four pieces of data works like this. First, you go to the Twitter apps web site and register each application while logged into your Twitter account that will host the application. For each app, you must specify the following. In return, Twitter will provide a consumer_key and consumer_secret for the application.

Application Name
Description
Application Web Site
Organization
Web Site
Application Type
Callback URL
Default Access Type
Use Twitter for Login?

Second, you must create an application on the public web to log into Twitter using your new credentials. This log in process will produce the remaining pieces necessary to generate an access_token and access_token_secret specifically for you. Can you do all of this? Obviously, only you can answer that question, and here’s the best guide we’ve found to walk you through the process. Suffice it to say, unless you are a seasoned programmer, it’s a hairy procedure with lots of opportunities for disaster. But be our guest and try your hand at it.

What’s Plan B? The alternative to Step #2 above is to log into a special web site we’ve created to generate the necessary credentials for you. But this means you have to provide (and trust us not to store or use) your consumer key and consumer secret. You also don’t want someone looking over your shoulder while you’re obtaining the remaining credentials. Unfortunately, without your consumer key and consumer secret, we can’t obtain an access_token and access_token_secret for you. And without those, nothing works.

To get started, you’ll need to register your new application with Twitter. For this to work, the Application Web Site address and Application Callback URL in Step #1 need to point to our web site, not yours. Once you get the secret codes, you can change the web links for your application to any fake address you care to make up. Just don’t delete the Twitter app you’ve created.

If it makes you sleep any better, this isn’t your bank account we’re talking about, it’s a specific Twitter application, one that we happened to write for you. So, if you like Plan B and you’re comfortable with our assurance that none of your confidential keys or passwords are being harvested, continue reading. Otherwise, use your own devices for getting the four necessary credentials. Then you can download the new OAuth-compatible version of TweedleD, and you’re off to the races.

Today we’re going to tackle TweedleD and get it back in operation. In coming weeks, we’ll tackle Tweet2Dial as well.

TweedleD Setup. Let’s begin with TweedleD which lets you send Twitter DMs and SMS alerts to announce every incoming call on your Asterisk server. This tutorial assumes you previously have installed the original version of TweedleD. If not, start there. Now let’s proceed. First, you’ll need to register the application with Twitter.

Step #1 is to use a browser to log into the Twitter account that you plan to use to generate the Direct Messages. If you can’t remember which Twitter account you used to originally set this application up, log into your Asterisk server as root and find the $username entry near the top of nv-twitter.php in the /var/lib/asterisk/agi-bin directory. Your password for this Twitter account will be there as well.

Step #2 is to open a new tab with your browser and visit Twitter apps. Choose the Register a New Application option and fill out the form like this:

Application Icon: Your choice
Application Name: TweedleD
Description: Asterisk Incoming Call Announcer
Application Web Site: http://pbxinaflash.com/oauth/index.php
Organization: Your Name or Company Name
Web Site: http://anything-you-like.com
Application Type: Browser
Callback URL: http://pbxinaflash.com/oauth/callback.php
Default Access Type: Read & Write
Use Twitter for Login: Leave unchecked

Click the Save button once you’ve entered all the data and completed the Captcha code. Once your app is registered, call it up and write down your Consumer key and Consumer secret. By the way, if these ever get compromised, you can generate new ones. But it means you’ll have to repeat the rest of this exercise since the other credentials will change as well.

Step #3 is to visit our OAuth Credentials Generator web site using your browser. Now plug in your Consumer key and Consumer secret. Then fill out the Captcha code and click the Submit button. If you got the Captcha code right, you’ll see your entries redisplayed with an option to Login with Twitter. Just click on the link, sign in to Twitter if you’re not already signed in, and click the Allow button when prompted whether to Allow TweedleD access to your new web site. If you get some funky error message from Twitter, then you didn’t enter your Consumer key and secret correctly. Try again. Otherwise, you’ll then get a screen that displays some information about your Twitter account as well as your new Access Token and Secret. Write these down, too.

Access Token: 32438037iO5cYUq4h0BJD4Z6Un5phaZHZ2zJ4P4LQ2t6TX8fpU

Access Secret: PQrhUlM9nnJwIrYCNTF07ai3vlXgMD3uf3qmmWJp6o

Step #4 is to download and install the new TweedleD application. Log into your Asterisk server as root and issue the following commands:
cd / wget http://bestof.nerdvittles.com/applications/TweedleD2.tgz tar zxvf TweedleD2.tgz rm TweedleD2.tgz cd /var/lib/asterisk/agi-bin nano -w nv-twitter.php
Step #5 is to configure TweedleD for use using your new Twitter credentials and your SMS email address. You’ll also need to choose whether to activate tweets, SMS messages, or both by setting $tweet and $sms to 1 if you want either or both of them activated.

If you’re activating Twitter, set $tweet=1 and fill in the 4 credentials that you wrote down previously: $consumer_key, $consumer_secret, $access_token, and $access_token_secret. In the $user4msg option, enter the Twitter account name to which the direct messages should be sent.

To activate SMS messaging as well, set $sms=1 and enter your SMS email address in the $smsaddress field using one of the examples provided.

Save the file, and you’re ready to try things out. Just make a call to your Asterisk server and the Twitter message announcing the call should arrive shortly thereafter. For syntax on other Twitter API calls, see the Twitter API Documentation. Enjoy!

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

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…

It’s PBX in a Flash 1.7.5.5: The Lean, Mean Asterisk Machine

Monday, July 12, 2010 / 11 Comments on It’s PBX in a Flash 1.7.5.5: The Lean, Mean Asterisk Machine

It’s been 18 months since a new version of PBX in a Flash was officially released. And we’ll explain the reasons why it’s quite unnecessary with our product in a few minutes. But, today, we’re proud to introduce the latest and greatest version 1.7.5.5 of PBX in a Flash featuring your choice of Asterisk® 1.4 or 1.6.2 with Zaptel or DAHDI support and FreePBX 2.6. It’s lean, mean, and incredibly flexible.

You don’t get the kitchen sink with the base PBX in a Flash ISO installs. Instead you get a rock-solid CentOS 5.5 operating system with the latest CentOS kernel on which to build an Internet telephony server that meets your specific needs. If we had to sum up this new release in a word, it would be refined. Newer hardware devices now are supported, and Mondo backups and other scripts have been tweaked to work with these new devices including Atom-based machines which are proving to be the ideal telephony platform for SOHO and small business deployments. As usual, documentation was not an afterthought. There’s a new installation tutorial and our award-winning knol has been updated to cover everything you’ll ever want to know about PBX in a Flash. And there’s loads of additional documentation on the PBX in a Flash web site. For the reading impaired, there’s even a 7-minute YouTube video to walk you through the installation process.

The installation procedure has been simplified. For most users, downloading the ISO, burning the ISO to a CD, booting from the CD, and pressing the Enter key is all the complexity you’ll face with a new PBX in a Flash install. For experts and resellers, there are the familiar options to perform network installs or to select different disk architectures including software RAID. Newer device drivers can be loaded as part of the installation process as well. And TM1000’s EndPoint Manager automatically configures almost any telephone on the planet for use with PBX in a Flash. All it takes is a quick download from SourceForge. For those with a physical handicap, you now can install the complete system with no user intervention by typing ksauto at the first prompt.

Overview. For those that prefer quick checklists to long articles, here’s the 30-minute, annotated, Baker’s Dozen PBX in a Flash 1.7.5.5 installation drill:

1. Download PBX in a Flash ISO
2. Burn ISO to a CD-ROM
3. Install system behind secure firewall
4. Boot target machine to be reformatted from CD
5. Press Enter key at first prompt
6. Choose keyboard for your country
7. Choose timezone for your location
8. Create a secure root password
9. Choose GOLD, SILVER, or BRONZE edition
10. Login as root & run update-scripts
11. Run update-fixes
12. Run passwd-master
13. Load FreePBX Modules OR Install Incredible PBX

A Better Mousetrap. Asterisk-based LAMP aggregations thankfully are more plentiful today, but we think we have a better mousetrap. Here are a few reasons why? First, PBX in a Flash is the only distribution that is totally source-based with Asterisk compiled from source as part of the install. What that means is when you purchase add-on hardware and it has a problem for some reason, all of the tools are already in place for you to contact the manufacturer or reseller and have them reconfigure or recompile whatever is necessary on your system to get you back in business quickly. It also means that most of our applications are compiled from source on your specific hardware which assures a more reliable and stable software platform on which to build your telephony system.

Second, we don’t release PBX in a Flash ISOs every other week. We don’t have to. Every time a new security patch is released for Asterisk, the "other guys" have to create a new RPM or ISO to support it. That means your system is vulnerable for weeks or months while that process is underway. In some cases, it means installing a new ISO and starting over. I wish I had a nickel for every time I reinstalled and basically started over with Asterisk@Home or trixbox. With PBX in a Flash, you simply type update-source and then update-fixes at the command prompt, and your system is brought current without missing a beat. The total server downtime is typically under 15 minutes!

Third, PBX in a Flash uses a two-step install process that all but eliminates the ISO obsolescence issues that have plagued other distributions. The PBX in a Flash ISO is used to install either the 32-bit or the 64-bit CentOS 5.5 operating system and kernel. When that process completes and after performing a yum update on CentOS 5.5, the installer then searches multiple sites on the Internet for our "payload files" which contain the latest, greatest versions of Asterisk to meet your specific requirements. The payload script also installs FreePBX and many of the customized features that make PBX in a Flash unique. If you need additional functionality, we have an entire web site, pbxinaflash.org, dedicated to add-on scripts. Most of these add-on scripts are available by typing help-pbx at the command prompt. All of them install without user intervention in a minute or two. Using this design, most bugs are eliminated as well without your having to do much of anything. Translation: More time to enjoy your production-quality VoIP PBX… and less all-nighters! Finally, if you’re new to Asterisk or just want to take advantage of a decade of expertise from the PIAF developers, just load the Incredible PBX over the top of your new PBX in a Flash install. In just 15 minutes, you’ll have an incredibly secure, turnkey PBX with dozens of add-on apps that can make and receive unlimited free calls in the U.S. and Canada thanks to Google Voice.

And, speaking of security, PBX in a Flash is the only distribution that brings you multiple layers of security out of the box. There’s the preconfigured Linux IPtables firewall. And, in addition, there’s the latest and greatest version of Fail2Ban which blocks malicious intruders attempting to guess your passwords and break into your system. We also strongly recommend adding a hardware-based firewall/router to block all access to your system unless you really know what you’re doing. Does all of this matter? Well, it’s your phone bill. Here’s a link to our article about a company that recently received an unexpected $120,000 phone bill in the mail. So you decide. If you read nothing else before embarking on your VoIP adventure, read our Primer on Asterisk Security!

So today we’re proud to introduce the 1.7.5.5 release of PBX in a Flash. It’s still the Lean, Mean Asterisk Machine designed to meet the needs of hobbyists as well as business users. And FreePBX 2.6 provides a rock-solid, graphical user interface to Asterisk that competes with any commercial PBX on the planet.

Getting Started with PBX in a Flash 1.7.5.5. Begin by downloading either the 32-bit or 64-bit ISO image for PBX in a Flash from SourceForge, Google, or from one of our download mirrors. Torrents are also available. And don’t worry. If you try to run the 64-bit install on a system that doesn’t support it, it’ll just sit there so you’ve got nothing to lose by trying the Ferrari first. Once you’ve got the ISO image in hand, use your favorite tool to burn it to a bootable CD. This next step is the most important. Do some reading!! There also are loads of helpful tutorials that are free for the downloading from our support site. Before you begin the install process, be aware that all drives (including USB devices) on your target system will be erased as part of the install process. So be sure to use a dedicated server for PBX in a Flash.

Update: A new PBX in a Flash installer is now available for USB Flash Drives.

What About Hardware? If you’re new to all of this, let us recommend you try either one of Dell’s entry-level PowerEdge servers or one of the newer Intel Atom-based small-footprint PCs or netbooks such as the Acer Aspire One or Acer Aspire Revo. On sale pricing is typically in the $200-$300 range. You can save an additional 2% plus $5 by using our coupon link in the right margin. Any of these systems is just about perfect for a home or small business server.

Basic Install. Once you have your new system, just insert the CD containing the ISO and then reboot the machine you wish to dedicate to PBX in a Flash. After reading this tutorial and the initial prompts and warnings, choose an option and press the <Enter key> to begin the installation. Choose your default keyboard and then choose your time zone and leave the UTC system clock option unchecked. Next choose a root password for your new system. Make it secure, and write it down (not on your shoe). IMPORTANT: Your server must have its system clock set correctly and be connected to the Internet before the install process begins! In about 15 minutes depending upon the speed of your PC, the machine will reboot when the installation of CentOS 5.5 is complete. Be sure to eject the CD at this point, or your system will boot again from the CD and start over.

After the reboot, the system will boot CentOS 5.5 and then prompt you to choose the version of Asterisk you’d like to install. Here are the three choices:

A – GOLD with Asterisk 1.4.21.2 and Zaptel
B – SILVER with latest Asterisk 1.4 version and DAHDI
C – BRONZE with latest Asterisk 1.6.2 version and DAHDI

If you plan to expose your server to the Internet in any way, we recommend you choose the SILVER version which is the most secure. And just to repeat, if you don’t have Internet connectivity, then the installation cannot complete. When the installation finishes, reboot your system and log in as root. The IP address of your PBX in a Flash system will be displayed once you log in. If it’s blank, type service network restart after assuring that you have Internet connectivity and access to a DHCP server that hands out IP addresses. Typing ifconfig should display your IP address on the eth0 port. Write it down. We’ll need it in a minute.

Now that you’ve logged in as root, you should see the IP address displayed with the following command prompt: root@pbx:~/. If instead you see bash displayed as the command prompt and it’s not green, then the installation has not completed successfully. This is probably due to network problems but also could be caused by the time being set incorrectly on your server. You can’t compile Asterisk if the time on your computer is a date in the past! For this glitch you basically have to start over. If it’s a network issue, fix it and then reboot and watch for the eth0 connection to complete. Assuming it doesn’t fail the second time around, the installation will continue. Likewise, if you do not have DHCP on your network, the installation will fail because the PBX will not be given an IP address.

Three Steps to Complete the Install. There are three important things to do to complete the installation. First, run the following commands after logging into your new server as root with your root password:

update-scripts (gets the latest PIAF scripts)
update-fixes (applies PIAF security patches and bug-fixes)
passwd-master (sets your FreePBX maint password)

Second, from the command prompt, run genzaptelconf or gendahdiconf if you have ZAP/DAHDI hardware. This sets up your hardware as well as a timing source for conferencing. If you’re using additional hardware for your Asterisk system, we recommend removing any modem before you install the cards. This will help avoid interrupt conflicts.

Third, decide how to handle the IP address for your PBX in a Flash server. The default is DHCP, but you don’t want the IP address of your PBX changing. Phones and phone calls need to know how to find your PBX, and if your internal IP address changes because of DHCP, that’s a problem. You have two choices. Either set your router to always hand out the same DHCP address to your PBX in a Flash server by specifying its MAC address in the reserved IP address table of your router, or run netconfig at the command prompt and assign a permanent IP address to your server. Be aware that netconfig no longer is a part of CentOS 5.5. Run install-netconfig to reinstall it. If you experience problems with the process, see this message thread on the forum.

If you’ve used one of the dLink firewall/routers we recommend and you plan to install the Incredible PBX, you can skip the rest of this article. We’ve done all of the work for you!

The Incredible PBX Inventory. For those wondering what’s included with The Incredible PBX, here’s a feature list of components you get in addition to the base install of PBX in a Flash with CentOS 5.5, Asterisk, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using provided scripts.

A2Billing (/root/nv/install-a2billing)
Amazon S3 Cloud Computing
AsteriDex
CallerID Superfecta (FreePBX Module adds Names to CID Numbers)
CallWho for Asterisk
Cepstral TTS for 32-bit, Asterisk 1.4 1 (/root/nv/install-cepstral.sh)
Preconfigured Email That Works with SendMail
Extensions (16 preconfigured with random passwords)
Fax Module using nvFax
FONmail
FreePBX Backups
Gizmo5 (Free Calls to Gizmo5 users worldwide: 1747xxxxxxx*1089)
Google Voice (preconfigured for free U.S./Canada calling)
Hamachi VPN (/root/nv/install-hamachi.x)
Hotel-Style Wakeup Calls (FreePBX Module)
ISN: FreeNum SIP Calling from Any Phone
MeetMe Conference Bridge (just dial C-O-N-F)
Mondo Full System Backups (/root/nv/install-diskbackup.x)
NewsClips from Yahoo
ODBC Database Support
PogoPlug Cloud Computing
Reminders by Phone and Web
SIP URI Outbound Calling (call any SIP URI worldwide for free)
Skype Inbound & Outbound Calling (Available 4/26)
TeleYapper
Tide Reports with xTide
Trunk Lister Script (/root/nv/trunks.sh)
Trunks (Vitelity, Fonica, SIPgate, IPkall, and ENUM)
Twitter Interface (Make Free Calls and Send SMS Messages)
Weather by Airport Code
Weather by ZIP Code
Worldwide Weather
Zaptel Updater (/root/nv/zaptel-update.sh)

If you’ve decided to roll your own and skip The Incredible PBX, then let’s continue…

Getting Rid of One-Way Audio. There are some settings you’ll need to add to /etc/asterisk/sip_custom.conf if you want to have reliable, two-way communications with Asterisk: nano -w /etc/asterisk/sip_custom.conf. The entries depend upon whether your Internet connection has a fixed IP address or a DHCP address issued by your provider. In the latter case, you also need to configure your router to support Dynamic DNS (DDNS) using a service such as dyndns.org. If you have a fixed IP address, then enter settings like the following using your actual public IP address and your private IP subnet:
externip=180.12.12.12 localnet=192.168.1.0/255.255.255.0
If you have a public address that changes and you’re using DDNS, then the settings would look something like the following:
externhost=myserver.dyndns.org localnet=192.168.0.0/255.255.255.0

(NOTE: The first 3 octets in the above localnet entries need to match your private IP addresses!)

Once you’ve made your entries, save the file: Ctrl-X, Y, then Enter. Reload Asterisk: amportal restart. If you assigned a permanent IP address, reboot your server: shutdown -r now.

Be aware that some people experience problems with the externhost approach outlined above. If your provider only gives you a dynamic IP address, you still can use the externip approach above so long as you have a method to frequently verify your IP address. The approach we actually use on our home network is to run a little script every 5 minutes. If it finds that your outside IP address has changed, it will automatically update your sip_custom.conf file with the new address. To use our approach, create a file in /var/lib/asterisk/agi-bin names ip.sh. Here’s the code:²
#!/bin/bash # File to log the IP Address IPFILE='/var/log/asterisk/externip' # Your local lan ip block localnet=192.168.1.0 # Nothing else needs to be changed. if [ ! -f "$IPFILE" ]; then echo "creating $IPFILE" echo first_time_usage > $IPFILE fi lastip=`cat $IPFILE` externip=$(curl -s -S --user-agent "PIAF 1.4"↩ http://myip.pbxinaflash.com | awk 'NR==2') if [ $externip != $lastip ]; then # Writes new IP address (if it has changed) to file. echo "$externip" > $IPFILE echo "externip=$externip" > /etc/asterisk/sip_custom.conf echo "localnet=$localnet/255.255.255.0" >>↩ /etc/asterisk/sip_custom.conf echo "srvlookup=yes" >> /etc/asterisk/sip_custom.conf echo "nat=yes" >> /etc/asterisk/sip_custom.conf asterisk -rx "dialplan reload" ; else exit 0; fi exit;

On line 5, enter the internal subnet for your server as the localnet entry. This is usually 192.168.0.0 or 192.168.1.0. YMMV!

Save the file and give it execute permissions: chmod +x /var/lib/asterisk/agi-bin/ip.sh. Then make asterisk the file owner: chown asterisk:asterisk /var/lib/asterisk/agi-bin/ip.sh.

Finally, add the following entry to the bottom of /etc/crontab:
*/5 * * * * asterisk /var/lib/asterisk/agi-bin/ip.sh > /dev/null

Activating Email Delivery of Voicemail Messages. We’ve previously shown how to configure systems to reliably deliver email messages whenever a voicemail arrives unless your ISP happens to block downstream SMTP mail servers. Here’s the link in case you need it. As it happens, you really don’t have to use a real fully-qualified domain name to get this working. So long as the entry (such as pbx.dyndns.org) is inserted in both the /etc/hosts file and /etc/asterisk/vm_general.inc with a matching servermail entry of vm@pbx.dyndns.org (as explained in the link above), your system will reliably send emails to you whenever you get a voicemail if you configure your extensions in FreePBX to support this capability. You can, of course, put in real host entries if you prefer. For 90% of the systems around the world, if you just want your server to reliably e-mail you your voicemail messages, make line 3 of /etc/hosts look like this with a tab after 127.0.0.1 and spaces between the domain names:
127.0.0.1 pbx.dyndns.org pbx.local pbx localhost.localdomain localhost
And then make line 6 of /etc/asterisk/vm_general.inc look like the following:
serveremail=voicemail@pbx.dyndns.org
Now issue the following two commands to make the changes take effect:
service network restart amportal restart

The command "setup-mail" can be used from the Linux prompt to set the fully-qualified domain name (FQDN) of the mail that is sent out from your server. This may help mail to be delivered from the PBX. One of things mail servers do to reduce spam is to do a reverse lookup on where the mail has come from, checking that there is actually a mailserver at the other end. You can only do this if you have set up dynamic DNS or if you have pointed a hostname at your fixed IP address. Once you have done this, and assuming your ISP is cooperative, then you will receive your voicemails via email if you wish (this is set within FreePBX),and your PBX will email you when FreePBX needs an update. You set this feature in FreePBX General Settings.

If your hosting provider blocks downstream SMTP servers to reduce spam, here’s a simple way to use your Gmail account (free!) as your SMTP Relay Host. Then you never have to worry about this again!

Setting Passwords and Other Stuff. Be aware that major security issues are reported from time to time with FreePBX. We strongly recommend that you not use FreePBX admin security alone to protect your system from a web attack. It may compromise root access to your entire server. For this reason, we recommend that you log in as root and immediately run passwd-master after completing the update-scripts and update-fixes scenario. This establishes Apache htaccess security on your FreePBX web interface. After running this conversion utility, you can only log into the FreePBX admin interface with the username maint (not admin) and the password which you establish when you run the utility.

Other passwords can be set in your system with these commands:
passwd... reset your root user password passwd-maint... reset your FreePBX maint password passwd-wwwadmin... for users needing FOP and MeetMe access passwd-meetme... for users needing only MeetMe access passwd-webmin... for users needing WebMin access to your server (very dangerous!)

There’s also an Administration password that you can set in the KennonSoft UI that displays when you point your browser to the IP address of your server. Do NOT use the same password here that you use elsewhere as it is not overly secure.

Configuring WebMin. WebMin is the Swiss Army Knife of Linux. It provides TOTAL access to your system through a web interface. Search Nerd Vittles for webmin if you want more information. Be very careful if you decide to enable it on the public Internet. You do this by opening port 9001 on your router and pointing it to the private IP address of your PBX in a Flash server. Before using WebMin, you need to set up a username and password for access. From the Linux prompt while logged in as root, type the following command where admin is the username you wish to set up and foo is the password you’ve chosen for the admininstrator account. HINT: Don’t use admin and foo as your username and password for WebMin unless you want your server trashed!
/usr/libexec/webmin/changepass.pl /etc/webmin root password

To access WebMin on your private network, go to http://192.168.0.123:9001 where 192.168.0.123 is the private IP address of your PBX in a Flash server. Then type the username and password you assigned above to gain entry. To stop WebMin: /etc/webmin/stop. To start WebMin: /etc/webmin/start. For complete documentation, go here.

Updating and Configuring FreePBX. FreePBX 2.6 is installed as part of the PBX in a Flash 1.7.5.5 implementation. This incredible, web-based tool provides a complete menu-driven user interface to Asterisk. The entire FreePBX project is a model of how open source development projects ought to work. And having Philippe Lindheimer’s as the Captain of the Ship is just icing on the cake. All it takes to get started with FreePBX is a few minutes of configuration, and you’ll have a functioning Asterisk PBX complete with voicemail, music on hold, call forwarding, and a powerful interactive voice response (IVR) system. There is excellent documentation for FreePBX which you should read at your earliest convenience. It will answer 99% of your questions about how to use and configure FreePBX. For the one percent that is not covered in the Guide, visit the FreePBX Forums which are frequented regularly by the FreePBX developers. Kindly post FreePBX questions on their forum rather than the PBX in a Flash Forum. This helps everybody. Now let’s get started.

Now move to a PC or Mac and, using your favorite web browser, go to the IP address you deciphered above for your new server. Be aware that FreePBX has a difficult time displaying properly with IE6 and IE7 and regularly blows up with older versions of Safari. Be safe. Use Firefox. From the PBX in a Flash Main Menu in your web browser, click on the Administration link and then click the FreePBX button. Once FreePBX loads, click the Module Administration option in the left frame. Now click Check for Updates online in the upper right panel. Next, click Download All which will select all but two modules for download and install. Scroll to the bottom of the page and click Process, then Confirm, then Return. Now repeat the process once more, then Process, Confirm, Return, Apply Config Changes, and Continue with Reload. Finally, scroll down the Modules listing until you get to the Maintenance section. Click on each of the following and choose Install: ConfigEdit, Sys Info, and phpMyAdmin. Then click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. All three of these tools now are installed in the Maintenance section of the Tools tab of FreePBX. You now have an up-to-date version of FreePBX. You’ll need to repeat the drill every few weeks as new updates are released. This will assure that you have all of the latest and greatest software. To change your Admin password, click on the Setup tab in the left frame, then click Administrators, then Admin in the far right column, enter a new password, and click Submit Changes, Apply Configuration Changes, and Continue with reload. We’re going to be repeating this process a number of times in the next section so… when instructed to Save Your Changes, that means "click Submit Changes, Apply Configuration Changes, and Continue with reload." Finally, don’t worry about the warnings alerting you that you’re using default passwords. Your system is behind a secure firewall, and these passwords are only accessible to someone that has access to your system and has your root password.

Choosing Internet Telephony Hosting Providers for Your System. Before you can place calls to users outside your system or to receive incoming calls, you’ll need at least one provider (each) for your incoming phone number (DID) and incoming calls as well as a provider for your outbound calls (terminations). We have a list of some of our favorites here, and there are many, many others. You basically have two choices with most providers. You can either pay as you go or sign up for an all-you-can-eat plan. Most of the latter plans also have caps on minutes so it’s more akin to all-they-care-for-you-to-eat, and there are none of the latter plans for business service. In the U.S. market, the going rate for pay as you go service is about 1.5¢ per minute rounded to the tenth of a minute. The best deal on DIDs is from Vitelity. They charge $3.99 a month for a DID with unlimited, free incoming calls. There’s a link to the Nerd Vittles discount on this service for PBX in a Flash users below.

Before you sign up for any all-you-can-eat plan, do some reading about the service providers. Some of them are real scam artists with backbilling and all sorts of unconscionable restrictions. You need to be careful. Our cardinal rule in the VoIP Wild West is never, ever entrust your entire PBX to a single hosting provider. As Forrest Gump would say, "Stuff happens!" And life’s too short to have dead telephones, even if it’s a rarity.

Setting Up FreePBX to Make Your First Call. There are four components in FreePBX that need to be configured before you can place a call or receive one from outside your PBX in a Flash system. So here’s FreePBX for Dummies in less than 50 words. You need to configure Trunks, Extensions, Outbound Routes, and Inbound Routes. Trunks are hosting provider specifications that get calls delivered to and transported from your PBX to the rest of the world. Extensions are internal numbers on your PBX that connect your PBX to telephone hardware or softphones. Inbound Routes specify what should be done with calls coming in on a Trunk. Outbound Routes specify what should be done with calls going out to a Trunk. Everything else is bells and whistles.

Trunks. When you sign up with most of the better ITHP’s that support Asterisk, they will provide documentation on how to connect their service with your Asterisk system. If they have a trixbox tutorial, use that since it also uses FreePBX as the web front end to Asterisk. Here’s an example from les.net. And here’s the Vitelity support page although you will need to set up an account before you can access it. We also have covered the setups for a number of providers in previous articles. Just search the Nerd Vittles site for the name of the provider you wish to use. You’ll also find many Trunk setups in the trixbox Trunk Forum. Once you find the setup for your provider, add it in FreePBX by going to Setup, Trunks, Add SIP Trunk. Our AxVoice setup (which is all entered in the Outgoing section with a label of axvoice) looks like this with a Registration String of yourusername:yourpassword@sip.axvoice.com:
allow=ulaw authname=yourusername canreinvite=no context=all-incoming defaultip=sip.axvoice.com disallow=all dtmfmode=inband fromdomain=sip.axvoice.com fromuser=yourusername host=sip.axvoice.com insecure=very nat=yes secret=yourpassword type=friend user=phone username=yourusername

And our Vitelity Outbound Trunk looks like the following (labeled vitel-outbound) with no registration string:
allow=ulaw&gsm canreinvite=no context=from-pstn disallow=all fromuser=yourusername host=outbound1.vitelity.net secret=yourpassword sendrpid=yes trustrpid=yes type=friend username=yourusername

Extensions. Now let’s set up a couple of Extensions to get you started. A good rule of thumb for systems with less than 50 extensions is to reserve the IP addresses from 192.x.x.201 to 192.x.x.250 for your phones. Then you can create extension numbers in FreePBX to match those IP addresses. This makes it easy to identify which phone on your system goes with which IP address and makes it easy for end-users to access the phone’s GUI to add bells and whistles. To create extension 201 (don’t start with 200), click Setup, Extensions, Generic SIP Device, Submit. Then fill in the following blanks USING VERY SECURE PASSWORDS and leaving the defaults in the other fields for the time being.

User Extension … 201
Display Name … Home
Outbound CID … [your 10-digit phone number if you have one; otherwise, leave blank]
Emergency CID … [your 10-digit phone number for 911 ID if you have one; otherwise, leave blank]
Device Options
secret … 1299864 < -- make this unique AND secure! dtmfmode ... rfc2833 Voicemail & Directory ... Enabled voicemail password ... 1299864 <-- make this unique AND secure! email address ... yourname@yourdomain.com [if you want voicemail messages emailed to you] pager email address ... yourname@yourdomain.com [if you want to be paged when voicemail messages arrive] email attachment ... yes [if you want the voicemail message included in the email message] play CID ... yes [if you want the CallerID played when you retrieve a message] play envelope ... yes [if you want the date/time of the message played before the message is read to you] delete Vmail ... yes [if you want the voicemail message deleted after it's emailed to you] vm options ... callback=from-internal [to enable automatic callbacks by pressing 3,2 after playing a voicemail message] vm context ... default

Now create several more extensions using the template above: 202, 203, 204, and 205 would be a good start. Keep the passwords simple. You’ll need them whenever you configure your phone instruments.

Extension Security. We cannot overstress the need to make your extension passwords secure. All the firewalls in the world won’t protect you from malicious phone calls on your nickel if you use your extension number or something like 1234 for your extension password because the SIP and IAX ports typically are exposed to allow connections to your providers. In addition to making up secure passwords, the latest version of FreePBX also lets you define the IP address or subnet that can access each of your extensions. Use it!!! Once the extensions are created, edit each one and modify the permit field to specify the actual IP address or subnet of each phone on your system. A specific IP address entry should look like this: 192.168.1.142/255.255.255.255. If most of your phones are on a private LAN, you may prefer to use a subnet entry like this: 192.168.1.0/255.255.255.0 using your actual subnet, of course.

Outbound Routes. The idea behind multiple outbound routes is to save money. Some providers are cheaper to some places than others. We’re going to skip that tutorial today. You can search the site for lots of information on choosing providers. Assuming you have only one or two for starters, let’s just set up a default outbound route for all your calls. Using your web browser, access FreePBX on your server and click Setup, Outbound Routes. Enter a route name of Everything. Enter the dial patterns for your outbound calls. In the U.S., you’d enter something like the following:
1NXXNXXXXXX NXXNXXXXXX
Click on the Trunk Sequence pull-down and choose your providers in the order you’d like them to be used for outbound calls.Click Submit Changes and then save your changes. Note that a second choice in trunk sequence only gets used if the calls fail to go through using your first choice. You’ll notice there’s already a 9_outside route which we don’t need. Click on it and then choose Delete Route 9_outside. Save your changes.

Inbound Routes. We’re also going to abbreviate the inbound routes tutorial just to get you going quickly today. The idea here is that you can have multiple DIDs (phone numbers) that get routed to different extensions or ring groups or departments. For today, we recommend you first build a Ring Group with all of the extension numbers you have created. Once you’ve done that, choose Inbound Routes, leave all of the settings at their default values and move to the Set Destination section and choose your Ring Group as the destination. Now click Submit and save your changes. That will set up a default incoming route for your calls. As you add bells and whistles to your system, you can move the Default Route down the list of priorities so that it only catches calls that aren’t processed with other inbound routing rules.

General Settings. Last, but not least, we need to enter an email address for you so that you are notified when new FreePBX updates are released. Scroll to the bottom of the General Settings screen after selecting it from the left panel. Plug in your email address, click Submit, and save your changes. Done!

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. 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 known as a Sipura SPA-3102. 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. This device also supports connection of your PBX to a standard office or home phone line as well as a telephone.

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 or the snom 360 Softphone which is a replica of perhaps the best IP phone on the planet. 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. Visit the PBX in a Flash Forum for lots of suggestions on telephones. Our personal favorite and the phone that PBX in a Flash officially supports is the Aastra 57i or 57iCT which also includes cordless DECT phone. Do some reading before you buy.

Where To Go From Here. The PBX in a Flash script repository at pbxinaflash.org also has gotten a facelift. That should be your next stop because it is the home of all the goodies that make PBX in a Flash shine. Tom King, the ultimate scripting guru, manages that site. So check it often. You’ll also find all of our Nerd Vittles Goodies work with this new release. Most of our original collection work flawlessly with Asterisk 1.4 including AsteriDex, Yahoo News Headlines, Weather by Airport Code, Weather by Zip Code, Worldwide Weather Forecasts, Telephone Reminders, MailCall for Asterisk, and TeleYapper. We have not yet completed testing with Asterisk 1.6, but most should work. Complete documentation for each application also is provided at the link above. And, if you still have a DBT-120 Bluetooth adapter, you’ll be happy to learn that it works out-of-the-box with PBX in a Flash. Dust off our recent article on Proximity Detection, and you should be in business in under 10 minutes. Enjoy!

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…

For Asterisk 1.6 or for 64-bit systems with Asterisk 1.4 or 1.6, use the Cepstral install procedures outlined in this Nerd Vittles article. [↩]
Join the following line to the original line of code whenever you encounter the ↩ character. [↩]

The Incredible PBX: Remote Phone Meets the Travelin’ Man

Monday, June 28, 2010 / 18 Comments on The Incredible PBX: Remote Phone Meets the Travelin’ Man

Ever wrestled with one of those thorny problems for weeks only to wake up in the middle of the night with the answer? Thus was born Travelin’ Man, a web- based, one-click Asterisk® application that automatically reconfigures your Asterisk PBX to enable remote SIP phone access from your cellphone, iPad, remote PC, NetBook, or desktop telephone.

News Flash: Be sure to read our latest article introducing Travelin’ Man 3, a completely new security methodology based upon FQDN Whitelists and DDNS. In a nutshell, you get set-it-and-forget-it convenience and rock-solid VoIP security for your Cloud-based PBX or any PBX in a Flash server that’s lacking a hardware-based firewall and you get both transparent connectivity and security for your mobile or remote workforce.

If you’ve read the Incredible PBX series of articles on Nerd Vittles, you already know what a thorny problem remote phone access is if you want to preserve the overall security of your server. Indeed, our recommendation has been to leave SIP access closed on your hardware-based firewall because of the dangers inherent in activating remote SIP access. Now we have a better idea!

Today’s new approach works like this. First, we’ll run a little script that secures all of your extensions with permit entries locking down all these connections to the IP address range within your private network. Then we’ll open the SIP and RTP ports on your hardware and software firewalls and map these ports to your Asterisk server’s private IP address. With this setup, no one can attempt remote SIP logins to your server because Asterisk blocks all SIP extension connection attempts except those originating inside your LAN. To manage external phone connections to your server, the install script creates a new virtual Apache web server on your Incredible PBX using port 83. We’ll enable and map TCP port 83 on your hardware and software firewalls to your server as well. Web access with port 83 is limited to running the Travelin’ Man app to activate external phones.

Now we’re ready to set up access to your server for remote devices. For each extension you wish to enable for remote access, we’ll create a special web directory using an obscure, random file name which will serve as the web link for the Travelin’ Man web app. For example, in the diagram above, directory 184778 manages extension 501, directory 2389957h manages extension 701, and directory 6993h5j manages extension 702. This is accomplished by simply changing the extension number in the index.php script stored in each directory.

When one of these web links is accessed remotely, the PHP script will automatically reconfigure Asterisk to enable access to the designated SIP extension on your server using the remote IP address from which the web page was accessed. And, of course, there’s an additional layer of SIP security as well. You still need your extension credentials to actually log in to your server with a softphone to place and receive calls. The Travelin’ Man installation process takes only a couple minutes, and the remote SIP activation procedure takes just a couple seconds each time you want remote access from a different location. Here’s a quick example of how it actually works.

Let’s assume we want to use the new $3.95 Bria SIP softphone on an iPad to connect as extension 501 on our Incredible PBX back at home. The problem is that the dynamic IP address of your iPad changes at each new site on your itinerary. Some locations have WiFi while others only have 3G connections.

First, we’ll generate an icon to run Travelin’ Man from your iPad desktop. Use the same procedure with an iPhone or iPod Touch, and there’s a similar procedure for Android devices.¹ You only have to do this once. Start up Safari on the iPad to access the new port 83 web server at the random web address the installer created to support extension 501. That web address is something like this using your own FQDN²: http://myserver.dyndns.org:83/184778. After establishing the link once, we’ll hit the + button in Safari and choose Add to Home Screen. This creates the TravelMan icon on the iPad. See the screenshot below of our demo iPad setup which used extension 221 instead of 501.

Once configured, it’s just two clicks to enable your remote phone anywhere: click once on the TravelMan icon. When your IP address is confirmed, return to your Home Screen and click the Bria softphone icon to establish a SIP connection back to your server. Behind the scenes, the Travelin’ Man application will generate the required permit entry for your remote IP address mapping it to the designated extension on your server, and then it will reload your SIP settings to make your Asterisk server accessible to the Bria softphone in your hotel room. The entire process takes only a couple seconds.

If your company happens to have a dozen traveling salesmen, then you’d simply assign a dedicated extension to each employee and create secure directory names for each person (e.g. 2389957h and 6993h5j in diagram above) with a copy of the Travelin’ Man app configured for that employee’s extension number. Now your entire mobile workforce has connectivity back to the home office from any location on the globe. And, when an employee leaves the company and another arrives, just create a new name for the old employee’s web directory to preserve the security of your system (e.g. 184778 in our example becomes 78hd773). Keep in mind that each time the Travelin’ Man app is run for any extension, it wipes out any previously authorized IP address entry for that extension. Thus, the security of your Incredible PBX is always preserved.

Prerequisites. Before proceeding with today’s install, you must be running a stock install of Incredible PBX with PBX in a Flash behind a properly-secured, hardware-based firewall³. We recommend the latest version of Asterisk 1.4 because it addresses a SIP vulnerability that might cause you problems if malformed SIP packets are targeted at your server. The current release of PBX in a Flash (1.7.5.5 Silver) is ideal, but any version of PBX in a Flash can be brought current with Asterisk using the update-source and update-fixes tools. Travelin’ Man assumes that you have the Incredible PBX base install of extensions: 501 plus 701-715. You can obviously add more or remove some, but you’ll need to manually adjust sip_custom_post.conf to reflect your actual extension list after the install completes.

The installer has been encrypted for your/our own protection. In source form, the script would allow anyone to defeat the Incredible PBX requirement. Doing so would mean the required IPtables security component would not be in place and properly configured to protect the underlying system from attack. So we’ve opted to play Big Brother to avoid potential security problems for all of us down the road. This article clearly explains all the necessary components if some folks want to roll their own version. We just don’t want the responsibility if something goes horribly wrong. As Forrest Gump would say, "Shit Happens." 🙂 If you don’t believe it, check out the latest security scramble in the trixbox forums.

Installation. Now we’re ready to get started. So log into your Incredible PBX as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/travelinman.tar.gz
tar zxvf travelinman.tar.gz
./travelinman.x

NOTE: If you’re using PIAF2 with CentOS 6.2, you’ll need to use the updated version of Travelin’ Man because of a syntax change in the Apache config file:

cd /root
wget http://incrediblepbx.com/travelinman2.tar.gz
tar zxvf travelinman2.tar.gz
./travelinman2

The first step in the install procedure is to lock down access to all of your extensions to your private LAN subnet. In case you ever want to do this on another server not running the Incredible PBX, here’s a link to our privip.sh shell script that shows how to do it. This should work on most FreePBX-based Asterisk systems.

Once the extensions are locked down, the script will modify your IPtables and Apache configurations to permit web access on port 83. Next, it will adjust your Asterisk setup to support the Travelin’ Man permit scheme. This involves reworking of sip_custom_post.conf so that permit settings for individual extensions can be stored in files named 501.inc, 701.inc, etc. Finally, the installation procedure will set up a single web site to support extension 501 with a randomized directory name for remote access.⁴ This setup will be stored in /var/www/travelman. To activate support for additional extensions, you would simply copy the subdirectory giving it a new random name: cp -r dir1 dir2. Then edit config.php in the new subdirectory and change the $extension entry.

To complete the install, you must reconfigure your hardware-based firewall and map the following ports to the private IP address of your server:

TCP 83
UDP 5060
UDP 10000-20000

When the installation is completed, it will show you how to access the new web site for extension 501 using either a fully-qualified domain name or a public or private IP address. Now just follow the steps at the beginning of this article to set up your Android or iDevice, and test things out. Enjoy!

Reminders: Be sure to review the comments to this article and the related support forum thread for a week or two for late-breaking enhancements and issues. Also, Incredible PBX comes preconfigured with call forwarding activated for extension 501. Don’t forget to either disable it or set up a real call forwarding number for extension 501 if you want your cellphone to ring. From any extension on your server, just dial *72501 to set up call forwarding. To cancel call forwarding and pass calls directly to the registered 501 softphone, dial *74 and enter 501. Also be aware that the default RingAll ring group (700) configuration on Incredible PBX systems does not include extension 501. So add 501 if you want your remote extension to ring for incoming calls.

The Incredible PBX: Basic Installation Guide

Adding Skype to The Incredible PBX

Adding Incredible Backup… and Restore to The Incredible PBX

Adding Multiple Google Voice Trunks to The Incredible PBX

Adding Remotes, Preserving Security with Incredible PBX

Continue reading Basic Installation Guide, Part II.

Continue reading Basic Installation Guide, Part III.

Continue reading Basic Installation Guide, Part IV.

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

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…

To create a desktop icon for Travelin’ Man on Android devices, navigate to the link with your browser. Then save the link as a Bookmark by clicking the Star icon in your browser then click Add. Return to the Home Screen and, from the screen on which you wish to add the icon, touch and hold your finger on the screen. When the Add to Home Screen menu appears, choose Shortcuts then Bookmarks and select the link you previously saved. As with iDevices, you only have to do this once. [↩]
FQDN = Fully-qualified domain name [↩]
We recommend the dLink Router/Firewall. Low Cost: $35 WBR-2310 Best: DGL-4500 [↩]
If you’d like to download the web site code independently from the Travelin’ Man install procedure, here’s the link. [↩]

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