Posts tagged: firewall

Knock Three Times: Pain-Free Remote Access to Your Asterisk or Linux Server

No. We’re not going to make you relive the 1970′s with us today although now you can listen to this Number 1 Hit and a million others for free with Amazon’s new Prime Music. No, we don’t get a commission if you sign up for Amazon Prime. Yes, we make millions when you buy something from Amazon using our links. Thank you! What we have for you today is a Number 1 Utility, and it works on virtually any Linux platform. If your fraternity or sorority had a secret knock to gain access, then you already know the basic concept. Port Knocker (aka knockd) from Judd Vinet is a terrific utility that runs as a daemon on your server and does just what you’d expect. It listens for knocks. When it detects three knocks on the correct three ports in the proper sequence and from the same IP address, it opens the IPtables Linux Firewall for remote access from that IP address to your server for a predefined period of time. This would allow you to log into your server with SSH or make SIP phone calls using a softphone registered to your remote Asterisk® server. What makes Port Knocker especially useful is the existence of knocking clients for virtually any smartphone, tablet, or desktop computer. For the Travelin’ Man, it’s another must have utility.

We introduced a turnkey implementation of Port Knocker in Incredible PBX for Ubuntu 14 late last week. If you were a pioneer earlier in the week, go back and install it again to take advantage of Port Knocker. Or better yet, follow along and we’ll show you how to install it on your own RedHat/CentOS or Ubuntu/Debian server in just a couple of minutes.

Prerequisites. We’ve built open source installation scripts for both the RedHat/CentOS platform as well as the Ubuntu/Debian operating systems. These knockd installers assume that you have a fully functional and locked down IPtables firewall with an existing WhiteList of authorized users. We’d recommend Travelin’ Man 3 if you need to deploy this technology and haven’t done so already. Last week’s Incredible PBX for Ubuntu 14 already includes Travelin’ Man 3 whitelisting technology. Read the article for full details.

Today’s knockd installers are fairly generic but, if you’re running a version of CentOS earlier than 6.x or Ubuntu earlier than 14 or Debian.anything, be advised that we haven’t tested these installers on those platforms so you’re on your own. Finally, if your server is sitting behind a hardware-based firewall (as we ALWAYS recommend), then you’ll also need to map three TCP ports from your hardware-based firewall to your server so that legitimate “knocks” can find their way to your server. These ports need not be opened in your IPtables firewall configuration! We’re just knocking, not entering. :-)

Overview. As configured, today’s installation scripts will install and preconfigure knockd to load automatically when you boot up your server. Three random TCP ports will be assigned for your server, and this port sequence is what remote users will need to have in order to gain access. Yes, you can change almost everything. How secure is it? Well, we’re randomizing the 3-port knock sequence using over 3,900 ports so you can do the math to figure out the odds of a bad guy guessing the correct sequence. HINT: 3900 x 3900 x 3900. Keep in mind that these “knocks” must all be received from the same IP address within a 15-second window. So sleep well but treat the port sequence just as if it were a password. It is! Once a successful knock sequence has been received, the default Port Knocker configuration will open all ports on your server for remote access from the knocking IP address for a period of one hour. During this time, “The Knocker” can log in using SSH or make SIP calls using trunks or extensions on the server. Port Knocker does not alleviate the need to have legitimate credentials to log into your server. It merely opens the door so that you can use them. At the bewitching (end of the) hour, all ports will be closed for this IP address unless “The Knocker” adds a whitelist entry for the IP address to IPtables during the open period. Yes, all of this can be modified to meet your individual requirements. For example, the setup could limit the range of ports available to “The Knocker.” Or the setup could leave the ports open indefinitely until another series of knocks were received telling knockd to close the IPtables connection. Or perhaps you would want to leave the ports open for a full day or a week instead of an hour. We’ll show you how to modify all of the settings.

Server Installation. To get started, log into your server as root and download and run the appropriate installer for your operating system platform.

For RedHat/Fedora/CentOS/ScientificLinux servers, issue the following commands:

cd /root
wget http://nerdvittles.com/wp-content/knock-R.tar.gz
tar zxvf knock*
rm knock-R.tar.gz
./knock*

For Ubuntu/Debian servers, issue the following commands:

cd /root
wget http://nerdvittles.com/wp-content/knock-U.tar.gz
tar zxvf knock*
rm knock-U.tar.gz
./knock*

For ARM-based servers, issue the following commands:

cd /root
wget http://nerdvittles.com/wp-content/knock-ARM.tar.gz
tar zxvf knock*
rm knock-ARM.tar.gz
./knock*

Server Navigation Guide. On both the RedHat/CentOS/Fedora and Ubuntu/Debian platforms, the knockd configuration is managed in /etc/knockd.conf. Before making changes, always shutdown knockd. Then make your changes. Then restart knockd. On RedHat systems, use service knockd stop and start. On Ubuntu, use /etc/init.d/knockd stop and start. By default, knockd monitors activity on eth0. If your setup is different, on Ubuntu, you’ll need to change the port in /etc/default/knockd: KNOCKD_OPTS="-i wlan0". On RedHat, the config file to modify is /etc/sysconfig/knockd and the syntax: OPTIONS="-i venet0:0".

In /etc/knockd.conf, create an additional context to either start or stop an activity. It can also be used do both as shown in the example code above. More examples here. There’s no reason these activities have to be limited to opening and closing the IPtables firewall ports. You could also use a knock sequence to turn on home lighting or a sprinkler system with the proper software on your server.

To change the knock ports, edit sequence. Both tcp and udp ports are supported. seq_timeout is the number of seconds knockd waits for the complete knock sequence before discarding what it’s already received. We’ve had better luck on more servers setting tcpflags=syn. start_command is the command to be executed when the sequence matches. cmd_timeout and stop_command tell knockd what to do after a certain number of seconds have elapsed since the start_command was initiated. If you’re only starting or stopping some activity (rather than both), use command instead of start_command and stop_command to specify the activity.

IPtables 101. The default setup gives complete server access to anyone that gets the knock right. That doesn’t mean they get in. In the PIAF World, it means they get rights equivalent to what someone else on your LAN would have, i.e. they can attempt to log in or they can use a browser to access FreePBX® provided they know the server’s root or FreePBX credentials.

If you would prefer to limit access to a single port or just a few ports, you can modify command or start_command and stop_command. Here are a few examples to get you started.

To open SSH access (TCP port 22):

/sbin/iptables -A INPUT -s %IP% -p tcp --dport 22 -j ACCEPT

To close SSH access (TCP port 22):

/sbin/iptables -D INPUT -s %IP% -p tcp --dport 22 -j ACCEPT

To open a range of SIP ports (UDP 5060 to 5069):

/sbin/iptables -A INPUT -s %IP% -p udp --dport 5060:5069 -j ACCEPT

To close a range of SIP ports (UDP 5060 to 5069):

/sbin/iptables -D INPUT -s %IP% -p udp --dport 5060:5069 -j ACCEPT

Here’s a gotcha to be aware of. If you’re using the Travelin’ Man 3 WhiteList setup on your server, be especially careful in crafting your IPtables rules so that you don’t accidentally remove an existing Travelin’ Man 3 rule in closing some port with knockd. You will note that the syntax of the knockd commands is intentionally a bit different than what you will find in your Travelin’ Man 3 setup. This avoids clobbering something accidentally.

Monitoring Activity. Here are the two best tools to monitor knockd activity to make certain your setup is performing as expected. The knockd log (/var/log/knockd.log) will tell you when a knocking attempt has occurred and whether it was successful:
[2014-07-06 14:44] starting up, listening on eth0
[2014-07-06 15:29] 79.299.148.11: opencloseSSH: Stage 1
[2014-07-06 15:29] 79.299.148.11: opencloseSSH: Stage 2
[2014-07-06 15:29] 79.299.148.11: opencloseSSH: Stage 3
[2014-07-06 15:29] 79.299.148.11: opencloseSSH: OPEN SESAME
[2014-07-06 15:29] opencloseSSH: running command: /sbin/iptables -A INPUT -s 79.299.148.11 -p tcp --dport 22 -j ACCEPT

Next, verify that the IPtables command did what it was supposed to do. iptables -nL will tell you whether port 22 access was, in fact, enabled for 79.299.148.11. The entry will appear just above the closing Chain entries in the listing:

ACCEPT     tcp  --  79.299.148.11         0.0.0.0/0           tcp dpt:22

Two things typically can go wrong. Either the knock from a client computer or cellphone wasn’t successful (knockd.log will tell you that) or IPtables didn’t open the port(s) requested in your knockd command (the iptables -nL query will show you that). In the latter case, it’s usually a syntax error in your knockd command. Or it could be the timing of the knocks. See /var/log/knockd.log.

Port Knocker Clients. The idea behind Port Knocker is to make remote access easy both for system administrators and end-users. From the end-user perspective, the simplest way to do that is to load an app on the end-user’s smartphone so that even a monkey could push a button to gain remote access to a server. If the end-user’s cellphone has WiFi connectivity sitting behind a firewall in a hotel somewhere, then executing a port knock from the smartphone should open up connectivity for any other devices in the hotel room including any notebook computers and tablets. All the devices typically will have the same public IP address, and this is the IP address that will be enabled with a successful knock from the smartphone.

Gotta love Apple’s search engine. Google, they’re not…

There actually are numerous port knocking clients for both Android and iOS devices. Here are two that we’ve tested that work: PortKnock for the iPhone and iPad is 99¢ and PortKnocker for Android is free. Some clients work better than others, and some don’t work at all or work only once. DroidKnocker always worked great the first time. Then it wouldn’t work again until the smartphone was restarted. KnockOnD for the iPhone, which is free, worked fine with our office-based server but wouldn’t work at all with a cloud-based server at RentPBX. With all the clients, we had better results particularly with cloud-based servers by changing the timing between knocks to 200 or 500 milliseconds. How and when the three knocks are sent seems to matter! Of all the clients on all the platforms, PortKnocker was the least temperamental and offered the most consistent results. And you can’t beat the price. A typical setup is to specify the address of the server and the 3 ports to be knocked. Make sure you have set the correct UDP/TCP option for each of the three knocks (the default setup uses 3 TCP ports), and make sure the IP address or FQDN for your server is correct.

Another alternative is to use nmap to send the knocks from a remote computer. The knock.FAQ file in your server’s /root directory will tell you the proper commands to send to successfully execute a connection with your server’s default Port Knocker setup. Enjoy!

Originally published: Monday, July 7, 2014


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! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.



Need help with Asterisk? Visit the PBX in a Flash Forum.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

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.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

  1. 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: Meet the New Kid on the Block

As much as we loved the moniker, the Orgasmatron build was in desperate need of a name change to more accurately describe its true heritage. We didn't look too far for just the right name. Meet The Incredible PBX!

Thanks to the Zero Internet Footprint™ design, it's the most secure Asterisk®-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server.1 And you won't find a more full-featured Personal Branch Exchange™.

NEWS FLASH: The Incredible PBX is now available for Asterisk 1.8! Go here.

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

The Incredible PBX is much more than just a name change. In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features tailored to meet the needs of the individual: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we'll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

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 5.4, Asterisk 1.4, 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.

Prerequisites. Here's what you'll need to get started:

  • Broadband Internet connection
  • $200 PC3 on which to run The Incredible PBX or a Proxmox VM
  • dLink Router/Firewall. Low Cost: $35 WBR-2310  Best: DGL-4500
  • Free Google Voice account (Available in U.S. without an invite at this link)
  • Free SIPgateOne residential account (U.S. cell to get SMS invite) OR
  • Free IPkall IAX account (recommended for international users)

Installing The Incredible PBX. The installation process is simple and straight-forward. Just don't skip any steps. Here are the 5 Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls:

1. Install the latest version of PBX in a Flash
2. Download & run The Incredible PBX installer
3. Set up your two provider accounts
4. Configure a softphone or SIP telephone
5. Run the configure-gv credentials installer

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. We recommend you install the latest 32-bit version of PBX in a Flash. This new build works much better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 5.5 operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use virtually identical payloads for all versions of PBX in a Flash.

Download the 32-bit, PIAF 1.6 version from Google, SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Or, if you are feeling really adventurous or if you have new, bleeding edge hardware, try our new 32-bit, PIAF 1.7 build which features CentOS 5.5. This new release is available from SourceForge or Google Docs. The MD5 checksum for the PIAF 1.7 build is 184cdb00142ccdd814b11de23fb00082.

Download the brand-new 32-bit PIAF 1.7.5.5. from SourceForge or one of our download mirrors. Burn the ISO to a CD. Then boot from the installation CD and type ksalt press the Enter key to begin.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can't find the kickstart file. Just tab to OK and press Enter. Don't change the name or location of the kickstart file! This will get you going. Think of it as a CentOS 'feature'. :-)

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose A choose PIAF-Silver option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes
status

When prompted, change the ARI password to something really obscure. You're never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time. Write down the dynamic IP address assigned to your server after running the status command. You'll need it shortly.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source with The Incredible PBX. The version of Asterisk installed from our payload file is very stable.

Running The Incredible PBX Installer. Log into your server as root and issue the following commands to download and run The Incredible PBX installer:

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

Have another 15-minute cup of coffee. It's a great time to consider a modest donation to the Nerd Vittles project. You'll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Here's a short video demonstration of the Incredible PBX installer process:

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. If you are eligible, we strongly recommend a SIPgate One residential account for The Incredible PBX. However, you may elect to use an IPkall account as an alternative. Both are free; however, you cannot register The Incredible PBX to IPkall's servers so you'll need to punch a hole in your firewall to receive incoming calls from Google Voice and IPkall. This step is not necessary with SIPgate accounts since there is a permanent registered connection between The Incredible PBX and SIPgate's servers!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we'd recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You'll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don't worry. You can erase your cellphone number from your account once it is set up and working properly. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn't matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you'll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You'll need these in a few minutes to complete the configuration of The Incredible PBX. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you're using IPkall as your intermediate provider, first log in to your hardware-based firewall/router and map UDP port 45694 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don't worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you're using SIPgate instead of IPkall.

After your firewall is properly configured, you'll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we'll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don't have an available number, your request apparently goes in the bit bucket. You'll know because IPkall typically turns these requests around in a few minutes. Don't worry about the mothership entry. We'll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it's dynamic, then you'll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you've got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you'll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It's worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it's a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Configuring Google Voice. Google Voice no longer is by invitation only so, if you're in the U.S. or have a friend that is, head over to the Google Voice site and register. After you've chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type.

Google places a test call to your number so you'll have to delay it a bit for IPkall. If you're using SIPgate, go ahead and tell Google to place the test call which will be forwarded to your cellphone. Enter the two-digit code that's displayed when you're prompted to do so. With IPkall, wait until we finish running the credentials configurator below.

While you're still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:

  • Call Screening - OFF
  • Call Presentation - OFF
  • Caller ID (In) - Display Caller's Number
  • Caller ID (Out) - Don't Change Anything
  • Do Not Disturb - OFF

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

If you're using SIPgate and you've confirmed your number, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to The Incredible PBX. We're ready to insert your credentials and SIPgate/IPkall information into The Incredible PBX. You'll need several pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 10-digit SIPgate or IPkall RingBack DID. You'll also need to reenter your passwd-master password which is used to configure CallerID Superfecta. Finally, you'll need to tell the configurator whether you're using a SIPgate or IPkall account. In the case of SIPgate, you'll also be prompted to enter your SIP ID and SIP password. These are NOT the same as your account credentials!!

Log back into your server as root and issue the following command to kick off the configurator: ./configure-gv.x. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!! Once you've checked and double-checked your entries, press Enter and The Incredible PBX setup will be completed. You'll need to press Enter again when the script finishes to reboot your PBX. After the reboot, your system will have randomly-generated passwords for every extension and voicemail box that is preconfigured on your system. The DISA password also has been changed. We generate five-digit passwords. If you will sleep better with longer passwords, be our guest. They are easily reset using the FreePBX web interface described elsewhere in this article.

Finally, log back into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you'll want a real SIP telephone, and you'll find lots of recommendations on Nerd Vittles. For today, let's download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

If you're using SIPgate as your provider with Google Voice, you're ready to place a test call. If you're using IPkall, we still need to verify your IPkall number with Google Voice. Return to Google Voice and tell it to place the test call to your IPkall number which you've already entered as your destination number. Your softphone will ring momentarily. Enter the two-digit code provided by Google Voice, and you're all set.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let's place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. Save the file and restart Asterisk with the command: amportal restart.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today's VoIP world. Start by reading our Primer on Asterisk Security. We've secured all of your passwords except your root password and your passwd-master password, and we're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Choosing a VoIP Provider. For this week, we'll point you to some things to play with on your new server. Then, in the subsequent articles below, we'll cover in detail how to customize every application that's been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we'd recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

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

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. Be safe!

Kicking the Tires. OK. That's enough tutorial for today. Let's play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O - Incredible PBX Demo (running on your PBX)
  • 1234*1061 - Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 - Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 - ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 - ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 - Schedule a hotel-style wakeup call from any extension
  • 1061*1061 - PBX in a Flash Support Conference Bridge
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)



Click above. Enter your name and phone number. Press Connect to begin the call.


Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and The Incredible PBX adds a bunch of additional options. See if you can find all of them. Also check out Tweet2Dial which uses Twitter to make Google Voice calls, send free SMS messages, and manage your Incredible PBX.

Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You'll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you've heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud as well. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It's perfect for off-site backups which we'll cover in a few weeks.

Don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

Originally published: Monday, April 19, 2010

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

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 The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Part II.

Continue reading Part III.

Continue reading 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.

Coming Soon. We haven't forgotten. We'll cover setting up multiple Google Voice accounts for simultaneous calling on multiple channels very soon. And the new (free) Skype Gateway to Asterisk for The Incredible PBX is now available. The FreePBX components already are in place to support inbound and outbound calling via Skype. You can even try a test call to our Aspire One Revo today by dialing nerdvittles from your favorite Skype client. Beginning today, this article will be available on http://IncrediblePBX.com. Then Nerd Vittles will return to our (almost) weekly schedule of new articles. 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.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest...

  1. Requires a SIPgate One account. []
  2. 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. []
  3. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. []
  4. Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device "inside" your firewall for further processing. []

The Incredible PBX: Adding Remotes, Preserving Security

Unlike most Asterisk®-based PBXs which are insecure as installed and leave it to you to implement sufficient safeguards to preserve the integrity of your system, the Incredible PBX is delivered with rock-solid, air-tight security already in place. Because it is designed to operate behind a hardware- based firewall, what you'll be doing when you want to add functionality with the Incredible PBX is loosening security rather than tightening it. The trick, of course, is to do it in a way that doesn't compromise the overall integrity of your system. As delivered, the Incredible PBX relies upon four layers of network security: a hardware-based firewall of your choice1, a preconfigured IPtables software-based Linux firewall, preconfigured Fail2Ban to monitor your logs for suspicious activity and to block specific IP addresses when abuse is detected, and random passwords for all extensions and DISA connections.

If you installed the Incredible PBX using SIPgate as the intermediate provider with Google Voice, then your hardware-based firewall should have no ports opened and forwarded to your server. If you used IPkall, then only UDP 4569 has been opened and forwarded to your server. And the Incredible PBX IPtables setup for IAX restricts access to just a few IP addresses to support IPkall.

There are obviously situations in which you will want or need additional connectivity. The most likely one involves activation of SIP telephones at remote locations, such as a branch office, or Grandma's house or a relative in college. The other obvious use is with cellphones and PDAs that support SIP clients such as Android phones, iPhones, and iPads.2

What we'd recommend you not do is open the SIP floodgate to your PBX by providing unrestricted inbound SIP access, but we'll show you how if you really want or need this functionality. As desirable as this can be, it is accompanied by an array of security issues that really are not worth the risks unless you know what you're doing and you're willing to stay on top of security updates and keep your system patched.

Let's first tackle how to provide limited inbound SIP functionality without selling the farm. If the remote site has a fixed IP address, the procedure to allow remote access to your server is fairly straight-forward: just map the SIP ports on the hardware-based firewall to your server (UDP 5000:5082 and UDP 10000:20000) and then restrict SIP access using IPtables to the remote IP address as well as the subnet of your private LAN. You can decipher your private subnet by running status. If your server's IP address is 192.168.0.123, then your private subnet would be 192.168.0.0. The IPtables firewall settings are stored in /etc/sysconfig/iptables. Edit that file and find the line that looks like this:

-A INPUT -p udp -m udp --dport 5000:5082 -j ACCEPT

Delete or comment out this entry with a leading # and insert new entries that look like the following using the public IP address(es) you wish to add plus the private subnet:

-A INPUT -p udp -m udp -s 141.146.20.10 --dport 5000:5082 -j ACCEPT
-A INPUT -p udp -m udp -s 141.146.20.11 --dport 5000:5082 -j ACCEPT
-A INPUT -p udp -m udp -s 192.168.0.0/255.255.0.0 --dport 5000:5082 -j ACCEPT


After making the changes, save the file: Ctrl-X, Y, then Enter. Then restart IPtables: service iptables restart.

Unfortunately, in many situations, the remote phone or cellphone uses an Internet connection with a dynamic IP address. So we don't know the actual IP address that will be assigned. There are a number of solutions to this problem, and we'll rank them in our order of preference. First, spend the $200 and install another Incredible PBX at the remote site. Then the two servers can be linked with IAX connections between the servers making connectivity between the systems totally transparent. Second, install VPN routers at both sites and use a private IP address to establish connectivity with the host system. In this situation, you will have the equivalent of a fixed IP address for the remote device which makes it the equivalent of the fixed IP address solution above. Third, install OpenVPN on your host system and purchase a SIP phone or cellphone that supports VPN connectivity. Most of the high-end SNOM SIP phones have this functionality as do Android phones, iPhones, and iPads. With this setup you also have the equivalent of a fixed IP address, even though it's on a virtual private network. Fourth, talk to the Internet service provider at your remote site and obtain the range of IP addresses that DHCP hands out to those using their services... or just make an educated guess.3

BEFORE Activating Full SIP Connectivity. OK. We hear you. You travel for a living, and the IP address of your cellphone changes hourly, all day, every day of the year. Then, yes, you are a candidate for a full-fledged Asterisk server with unlimited SIP access. Before covering how, let's review what responsibilities go with running such a server. Bear in mind that one compromised SIP password or otherwise vulnerable application on your server (including Asterisk, FreePBX, SSH, and hundreds of others), and you may very well be the proud owner of a whopping phone bill. And we're not talking hundreds of dollars. It could very well be tens of thousands of dollars. And it doesn't take weeks or months. It could be a few hours.

Baker's Dozen SIP Security Checklist

1. Keep Asterisk Current & Patched
2. Keep FreePBX Current & Patched
3. Make Frequent Backups
4. Visit PBX in a Flash Forums Regularly
5. Subscribe to PBX in a Flash RSS Feed
6. Secure Alphanumeric Extension Passwords
7. Secure DISA, VMail, Root, FreePBX Passwords
8. Lock Down Extensions with Deny/Permit
9. Turn Off Recurring Payments with Providers
10. Restrict Trunks to 1-2 Simultaneous Calls
11. Tighten Dialplan by Removing Wildcards
12. Eliminate Intl & Toll Calls With Providers
13. Check FreePBX Call Logs Daily for Abuse

Baker's Dozen SIP Security Checklist. Before opening the floodgates, let's review what you need to do. First, you'll need to run the very latest version of Asterisk... all the time. This means you need to monitor asterisk.org, and keep your system up to date by running update-scripts, update-source, and update-fixes regularly. The default version of Asterisk on current PBX in a Flash and Incredible PBX builds is extremely reliable, but it contains SIP and IAX vulnerabilities which should not be exposed directly to the Internet! Second, you need to run the latest version of FreePBX and apply all patches as they are released. Third, you need to make frequent backups appreciating that sometimes the Asterisk and FreePBX developers get things horribly wrong, and stuff that used to work no longer does. Believe it or not, they're human! Fourth, you need to visit the PBX in a Flash Forums daily and keep abreast of security alerts and bug reports on CentOS, Asterisk, and FreePBX. Fifth, you need to subscribe to the PBX in a Flash RSS Feed which provides regular security alerts when there are reported problems. Sixth, you need to really secure your extension passwords with very long, complex alphanumeric passwords. Ditto for your root and FreePBX passwords! Seventh, for DISA and voicemail, these passwords need to be numeric, complex, and extra long. Eighth, you need to lock down as many of your extensions as possible with deny/permit settings to restrict the IP addresses of those extensions. If you only have one or two remote SIP extensions with dynamic IP addresses, then all of the rest should have deny/permit entries! Ninth, turn off recurring payments with all of your telephony providers and keep minimal funds available in all of your accounts. This means you'll have to monitor these accounts to make sure they are not deactivated for lack of funds. Tenth, restrict all of your trunks to one or at most two simultaneous calls to reduce your call exposure in the event someone breaks into your system. Eleventh, tighten up your Trunk Dial Rules and eliminate any entries that would permit calls to anywhere in the world! If you don't regularly make international calls, there's absolutely no reason to have such entries in your dialplan. If you still have Ma Bell PSTN lines, this is even more important. In fact, consider eliminating long distance access to all of these trunks. Twelfth, where possible, configure your provider accounts to eliminate international and toll calls of all varieties. Finally, check your FreePBX call log every day to make certain no one is making calls on your nickel.

If you are unwilling or unable to perform these Baker's Dozen steps while continuing to monitor the sites provided and recheck your setup regularly (at least every week), don't activate unrestricted SIP access to your server.

Other Options. Consider using an intermediate provider such as voip.ms to provide SIP URI access to your server. Keep in mind that having a registered connection between your server and a VoIP provider alleviates the need to punch a hole in your firewall. So the idea here is to sign up for an inexpensive voip.ms account and set up the trunk connection with your server as either an IAX or SIP account with an always-on connection. Then voip.ms gives you the option of activating a SIP URI as part of a subaccount setup. Just create an internal extension on their server, and this will generate a SIP URI, e.g. 123456666@sip.us4.voip.ms where 12345 is your voip.ms account number and 6666 is the internal extension you created. This lets you connect directly with your server through the SIP URI from anywhere once you map this subaccount to an extension or IVR on your server. The charge for SIP URI calls is only $.001 per minute. The last step is to use this SIP URI in your remote SIP phone to connect back to your server. You can take advantage of the full range of Asterisk functions once these calls reach your server including IVRs and DISA. The approach is not only simple to implement, but it's also safe and economical.

There are some other alternatives as well. Use something like Google Voice or Ooma to redirect calls to your cellphone when you're traveling. Or buy an Ooma for Grandma or a MagicJack for Joe College. These options also are safe, secure, and quite inexpensive.

Just Released: Remote Phone Meets Travelin' Man

Activating Inbound SIP on Your Server. If you still are hell-bent on opening SIP access to your server, the Incredible PBX already is preconfigured to support it. Just map the SIP ports on your hardware- based firewall to your server (UDP 5000:5082 and UDP 10000:20000). Once activated, anyone can reach you through the following SIP URI using the actual public IP address of your server: mothership@12.34.56.78. You also can adjust the e164 trunk in FreePBX to route inbound calls to any destination desired. Then register your phone number on e164.org and others can call you at no cost using your traditional phone number. Enjoy!


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

Remote Phone Meets Travelin' Man with The 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.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest...

  1. We, of course, continue to recommend a dLink Router/Firewall. Low Cost: $35 WBR-2310  Better: DIR-825  Best: DGL-4500 []
  2. We recommend the free SipAgent client for Android devices and the commercial Acrobits Softphone for iPods and iPads. []
  3. Adding an entry like the following would dramatically reduce the likelihood of a SIP attack: -A INPUT -p udp -m udp -s 141.146.0.0/255.255.0.0 --dport 5000:5082 -j ACCEPT []

Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk

It’s been an exciting couple of weeks watching the overwhelmingly positive response to our release of Orgasmatron 5.1. With this version, we introduced a new Asterisk® security model that took into account the ever-increasing security risks posed by exposing web and telephony servers to direct Internet access. The bottom line is this. If your telecom requirements still can be accomplished by placing a server securely behind a $35 hardware-based Internet firewall with no Internet exposure, then it makes absolutely no sense to dangle such a tempting target in front of the world’s most nefarious creeps.

News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

Our experience suggests that the only trade off with this new approach is the inability to receive anonymous SIP calls… a small price to pay considering the potential financial and computer risks involved. You still can place outbound VoIP calls as well as placing and receiving calls using any of the phone numbers registered on your new PBX in a Flash server. And, thanks to Google Voice, SIPgate, and IPkall, all inbound calls are free, and all outbound calls to numbers in the U.S. and Canada are free as well.

If a SIP URI and your own Freenum/ISN number are simply features you can’t live without, sign up for a voip.ms IAX account, and you’ll get a SIP URI for free. Inbound SIP URI and Freenum/ISN calls will set you back $1 for every 1,000 minutes billed in 6 second increments.

Or you can sign up for a free IP Freedom CallCentric account and configure a new SIP trunk in FreePBX by following these directions. Once configured, your new server SIP URI will be 1777xxxxxxx@in.callcentric.com where xxxxxxx is your assigned 7-digit CallCentric number.

Keep in mind that a new security vulnerability has been found with either Asterisk or FreePBX almost monthly. The chart below tells you why. With virtually limitless attack surfaces because of the number of interrelated components in CentOS, Asterisk, and FreePBX comes enormous and recurring potential for remote compromise of these systems. Rather than play this cat-and-mouse security game with the underworld, the Orgasmatron design changes the paradigm. It lets you use any (secure or insecure) version of Asterisk and FreePBX without worrying about any outside attacks. Do passwords on your new server matter? Not really… unless there is someone inside your firewall that you don’t trust. :roll: Are we going to secure them anyway? Absolutely. But instead of the constant worry over new security vulnerabilities, Orgasmatron 5.2 lets you enjoy exploring the world of Asterisk and VoIP telephony with an incredibly rich feature set that you won’t find anywhere else, period! We’ll resist making any other device analogies, but the idea here is to protect the good guy (you!) while keeping the bad guys out. No penetration. No worries. Simple as that.

In our former life working for a living, we actually procured and managed multimillion dollar PBXs as part of our “other duties as assigned.” Without qualification, we can tell you that the feature set that Orgasmatron 5.2 brings to the table for free runs circles around anything you could buy (then or now) in the commercial marketplace. And, at one time or another, we purchased every Nortel feature good money could buy. There’s one other difference. Orgasmatron 5.2 runs swimmingly on a $200 Atom-based PC that you can purchase at any Best Buy as well as hundreds of other stores including Amazon, NewEgg, and Buy.com. We paid more than $200 to provision an additional extension on our Nortel switch! You, of course, can add as many extensions as you like. De nada.

So, why a new version of Orgasmatron in only a few weeks? Well, it’s not security-related. In fact, there is nothing wrong with continuing on with Orgasmatron 5.1. Unfortunately, it relied exclusively upon SIPgate to make free Google Voice calls in the U.S. and Canada. And SIPgate required an invite using an SMS message from a U.S.-based cellphone. That pretty well knocked out all of our friends living outside the United States. Today’s version fixes that by letting anyone sign up for a free IPkall phone number in Washington state. All you need is a valid email address. The setup process is a bit more complex because IPkall doesn’t support registered connections to their servers. But we’ll walk you through the additional steps and, once completed, your server will be just as secure as the SIPgate approach we set up with Orgasmatron 5.1. And few, if any, Linux skills are required to set up or manage Orgasmatron 5.2. As we’ve noted previously, if you can handle slice and bake cookies, you’ve got the necessary skillset! Be aware this is about a one-hour project, and you need to track through the article carefully, or the entire house of cards comes down.

New Asterisk Security Model. Orgasmatron 5.2 maintains our design goal of running an absolutely secure Asterisk PBX from behind a hardware-based firewall with either NO INBOUND PORTS exposed to the Internet with SIPgate or an IP-address-restricted IAX port for IPkall. Don’t defeat this security mechanism by exposing additional ports on your PBX in a Flash server to Internet access. And choose your NAT-based firewall/router carefully. All of these devices are not created equally. Not only do some perform better than others, but certain models are notoriously bad at handling NAT-based routing tasks, a critical requirement in the Asterisk VoIP environment. In almost every case of problems with one-way audio, the real culprit can be traced back to a crappy router. For $35, you really can’t go wrong with the dLink WBR-2310. If you want traffic shaping functionality as well, take a look at dLink’s Gaming Router, our personal favorite.

As long as your router, Google Voice, SIPgate, and IPkall passwords are secure, you can sleep like a baby. We use an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada because Google Voice actually places two calls to connect you to your destination. First, you get a call back. And then the party you’re calling is connected. The SIPgate or IPkall trunk is used by Google Voice to call you back so the inbound call is always free. We handle the interconnection magic with Asterisk transparently so your calls appear to be processed as if you were using a standard telephone to dial out. Just refrain from using extension 75 in Asterisk for personal conferencing!

The choice is yours. You can use SIPgate with no incoming ports exposed to your server from the Internet. Or you can use IPkall and map UDP port 4569 (IAX2) on your hardware-based firewall to the internal IP address of your new PBX in a Flash server. Even with the IPkall setup, we’ve locked down IPtables (our Linux firewall) to restrict IAX access to several specific IP addresses so your server remains absolutely secure. We’ve also included support for FonicaTec’s IAX offering for those that want a backup IAX provider. We’ll have much more to say about IPtables in coming weeks.

If you’ve already installed Orgasmatron 5.1 and it’s working for you, do you need to upgrade? NO. With the exception of the new IAX support for IPkall, the code in Orgasmatron 5.2 is identical.

We, of course, continue to recommend that you sign up with Vitelity so you have an alternate communications vehicle in the event of a problem with your free service. Vitelity also can provide 911 emergency service for your home or home office. You can save a little money while supporting the PBX in a Flash project by using the links at the end of this article.

Swiss Army Knife Inventory. There’s no need for a Swiss Army Knife if you don’t know what all the blades are for. So, for those that are wondering what’s included in the Orgasmatron 5.2 build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with CentOS 5.4, Asterisk 1.4, 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 the scripts that are provided.

Prerequisites. Here’s what you’ll need to get started:

  • Broadband Internet connection
  • Rock-solid NAT router/firewall. Recommend: $35 dLink WBR-2310
  • $200 PC on which to run PBX in a Flash or a Proxmox Virtual Machine
  • Free Google Voice account (HINT: Under $2 on eBay)
  • Free SIPgateOne residential account (Use cell to get SMS invite) OR
  • Free IPkall IAX account

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

Today’s Drill. The installation process is straight-forward, but a little different than the Orgasmo 5.1 scenario because of the need to accommodate IPkall. Just don’t skip any steps. In a nutshell, here are the 6 Steps to Free Calling and an incredibly versatile, preconfigured Asterisk PBX:

1. Install the latest version of PBX in a Flash
2. Run the Orgasmatron 5.2 Installer
3. Configure a softphone or SIP telephone
4. Configure Providers for Orgasmatron 5.2
5. Enter your Google Voice and SIPgate/IPkall credentials
6. Change existing passwords to secure your system

Installing PBX in a Flash. Here’s a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta on a new Atom-based PC. This beta is virtually identical to version 1.4 except it uses CentOS 5.4 instead of CentOS 5.2. This means it works better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use the identical payload for versions 1.3, 1.4, 1.5, and 1.6 of PBX in a Flash. The beta label simply means we haven’t had time to sufficiently test CentOS. But this is not a Microsoft-style beta so fear not!

Download the 32-bit, PIAF 1.6 version from Google, SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot from the installation CD and type ksalt to begin.

WARNING: This install will completely erase, repartition, and reformat ALL disks on your system! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can’t find the kickstart file. Just tab to OK and press Enter. Don’t change the name or location of the kickstart file! This will get you going. Think of it as a CentOS ‘feature’. :-)

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose A option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes

When prompted, change the ARI password to something really obscure. You’re never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source on the Orgasmatron builds because of parking lot issues in the latest releases of Asterisk.

Running the Orgasmatron 5.2 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.2 installer:

cd /root
wget http://pbxinaflash.net/orgasmo52.x
chmod +x orgasmo52.x
./orgasmo52.x

Have another 15-minute cup of coffee. It’s a great time to consider a modest donation to the Nerd Vittles project. You’ll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Now run passwd-master1. Set your FreePBX passwords to something very secure but different from your Linux root password.

Next, type status2 and press Enter. Write down the IP address of your new server.

If you’re using IPkall, now’s the time to log in to your hardware-based firewall/router and map UDP port 45693 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don’t worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you’re using SIPgate instead of IPkall.

For good measure, we recommend you reboot your server at this point. The command to type is simple: reboot4

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone, and you’ll find lots of recommendations on Nerd Vittles. For today, let’s download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using 82812661 as the password for extension 701 and the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Don’t Forget! After you change your extension passwords later in this tutorial, you will need to update the password entry in X-Lite, or you will no longer be able to place calls! In fact, you will get locked out of your server for 90 minutes after three failed password attempts. So put this on a sticky note so you don’t forget, or you’ll regret it in about 15 minutes.

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. And there’s really no reason you can’t use both if they’re available in your location. Do NOT use special characters in your provider passwords, or nothing will work! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we’d recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You’ll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don’t worry. You can erase your cellphone number from your account once it is set up. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn’t matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you’ll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You’ll need these in a few minutes to configure PBX in a Flash. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you’ve opted to use IPkall, here’s the drill. First, you’ll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we’ll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don’t have an available number, your request apparently goes in the bit bucket. You’ll know because IPkall typically turns these requests around in a few minutes. Don’t worry about the mothership entry. We’ll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it’s dynamic, then you’ll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you’ve got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you’ll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It’s worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

We’re going to be making some entries in FreePBX which is the web-GUI that manages PBX in a Flash. For now, we simply need to enter your new IPkall phone number so that incoming calls to your IPkall number will actually ring on your softphone. Later, we’ll make some further adjustments once we get Google Voice humming along.

Using a web browser from your desktop, log in to FreePBX 2.6 at the following link substituting your server’s private IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master).

When FreePBX loads, choose Setup, Trunks, ipkall (iax). In the USER Context field, enter your 10-digit IPkall phone number. Click Submit Changes, Apply Configuration Changes, Continue with Reload to save your settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it’s a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Now let’s test your new phone number. Call your IPkall number from a cellphone or some other phone. Your softphone should ring. Answer the call, and be sure you have voice in both directions! Do not proceed without success here, or the rest of the adventure is a waste of your time.

Configuring Google Voice. Google Voice still is by invitation only so the first thing you’ll need is an invite. If you’re in a hurry, then stroll over to eBay where you’ll find lots of them for under $2. Once you have your invite in hand, click on the email link to set up your account. After you’ve chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type. Trust us.

Google then will place a call to your number and ask you to enter a confirmation code that’s been provided. When your cellphone (SIPgate) or softphone (IPkall) rings, answer it and punch in the number. Wait for confirmation. Then hang up.

As we mentioned earlier, there’s no reason you can’t set up both SIPgate and IPkall forwarding numbers in Google Voice. Just repeat the drill with the other provider’s number if you wish to activate both numbers for use with Google Voice. They’re not both going to ring simultaneously as you will see in a minute.

While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:

  • Call ScreeningOFF
  • Call PresentationOFF
  • Caller ID (In)Display Caller’s Number
  • Caller ID (Out)Don’t Change Anything
  • Do Not DisturbOFF

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

Finally, place a test call to your new Google Voice number and be sure your cellphone or softphone rings. Don’t move forward until you’ve been able to successfully place a call to your phone by dialing your Google Voice number. Once this is working, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to PBX in a Flash. We’re ready to insert your Google Voice credentials and SIPgate/IPkall number into PBX in a Flash. You’ll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate or IPkall RingBack DID (beginning with a 1). Don’t get the 10-digit GV number mixed up with the 11-digit SIPgate/IPkall RingBack DID, or nothing will work. :-)

Log back into your server as root and issue the following command: ./configure-gv. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!

Configuring FreePBX. Now shift back to your Desktop and, using a web browser, log in to FreePBX 2.6 at the following link substituting your actual IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master). Depending upon which intermediate provider you’re using, do the following:

SIPgate Setup. When FreePBX loads, choose Setup, Trunks, sipgate. In Peer Details, replace both instances of sipID with your actual SipGate SIP ID. In Peer Details, replace sipPassword with your actual SipGate SIP Password. In Register String, replace sipID with your SipGate SIP ID, replace sipPassword with your SipGate SIP Password, and replace 3333333333 with your 10-digit SipGate Phone Number. When finished, the Register String should look something like the following:

7004484f0:B8TTW3@sipgate.com/4155201234

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

SIPgate and IPkall Setup. While still in FreePBX with your browser, click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate or IPkall Phone Number. In CallerID Number, replace 7777777777 with your 10-digit Google Voice Number.

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

Securing FreePBX. You’re almost done. While still in FreePBX, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit. After changing secret and Voicemail Password, repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.

Now change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue.

Don’t forget to adjust your X-Lite password to match the password entry you made for extension 701!

Orgasmatron Test Flight. The proof is in the pudding as they say. So let’s try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let’s place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. Save the file and restart Asterisk with the command: amportal restart.

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then, in the subsequent articles below, we’ll cover in detail how to customize every application that’s been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we’d recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

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

Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O – Nerd Vittles Orgasmatron Demo (running on your PBX)
  • 1234*1061 – Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 – Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P – Enter a five digit zip code for any U.S. weather report
  • 6-1-1 – Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 – Get the latest news and sports headlines from Yahoo News
  • T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
  • F-A-X – Send a fax to an email address of your choice
  • 4-1-2 – 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L – Record a message and deliver it to any email address
  • C-O-N-F – Set up a MeetMe Conference on the fly
  • 1-2-3 – Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 – ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 – ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 – Schedule a hotel-style wakeup call from any extension
  • 1061*1061 – PBX in a Flash Support Conference Bridge
  • 882*1061VoIP Users Conference every Friday at Noon (EST)


Click above. Enter your name and phone number. Press Connect to begin the call.


Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. For starters, you’ll want to activate CallerID Lookups in FreePBX. Choose Setup, CID Superfecta, Default and enter the maint password you created with passwd-master. Then choose Tools, Module Administration, CallerID Lookup, Enable, Process and Save the Settings. Then edit each of the Inbound Routes and choose CallerID Superfecta as the CID Lookup Source. Save your changes. Finally, choose Setup, CallerID Lookup Sources, CallerID Superfecta and be sure your maint password created with passwd-master is correct here, too. If not, update it. For additional tips, visit the forums.

Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You’ll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you’ve heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It’s also perfect for off-site backups!

Also check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

Continue reading Part II.

Continue reading Part III.

Continue reading 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 for Orgasmatron 5.1 and 5.2. 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.

Coming Attractions. In our next episode, we’ll walk you through the process of adding a second, third, fourth, and fifth Google Voice line to your server so that you’ll never run out of free calling on 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.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

  1. passwd-master is the PIAF utility for setting a master password for FreePBX access with the maint user account. []
  2. status is the PIAF utility program that displays the current status of most major applications running on your server. []
  3. Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device “inside” your firewall for further processing. []
  4. reboot is the Linux command for restarting your server. It’s functionally equivalent to shutdown -r now. []

Introducing Phone Genie for Asterisk (Email Edition)

From Our Disney Cruise Family Scrapbook Almost two years ago, we introduced Phone Genie for Asterisk®. It let you reconfigure your Asterisk system remotely using your favorite web browser. This included the ability to set and adjust call forwarding, call waiting, and Do Not Disturb for any Asterisk extension. In addition, you could enter Asterisk CLI commands and execute a number of Linux system commands, all from the convenience of your web browser. Phone Genie for Asterisk remains one of the all-time favorite downloads of our readers.

Unfortunately, you don't always have access to a web browser when you're away from your Asterisk server. So today we introduce the perfect complement to the original Phone Genie with our new Email Edition. By following this quick tutorial, you can configure your Asterisk server to respond to any Asterisk CLI command which can be sent from almost any email client on the planet. And we'll perform all this magic with less than a dozen lines of bash scripting. Asterisk CLI commands have almost limitless possibilities. Use Phone Genie to check the status or change the functionality of just about any component on your server.

How It Works. The best way to explain how all of this works is to use a simple example. Let's assume you've left home and forgot to transfer your inbound calls for extension 701 to your cellphone. What we'll do is send a simple email message to a special user account on your Asterisk server that we've set up specifically to handle email directives for your server. Unlike most email addresses, we want this one to be unintuitive so strangers aren't sending messages to your server all the time. Let's assume the address is kxt1498@myserver.dyndns.org for this example. Using any email client, just address a message to that account. For the subject of the message, we'll use the following:

Asterisk: database put CF 701 6781234567

It doesn't really matter whether you include a message with the email. As long as the subject of the email is in the proper form, that's all that matters. The command above activates call forwarding for extension 701 and sends the calls to 6781234567. The command uses standard Asterisk CLI syntax.

On your Asterisk server, we'll have a simple bash script that runs every minute or two to check for new emails in the kxt1498 user's mailbox. If it finds a new message, it will parse the subject line, make certain there is a password match, and then send the command (unaltered) to the Asterisk Command Line Interface for processing. Here's an overview of all the CLI commands. The results of executing the command will be emailed to the address you've configured in the script. This works as both confirmation that your command has been executed and a security alert that your Asterisk system has been accessed using the Email Edition of Phone Genie. In the above example, you would receive an email at the address you've configured in the script with a subject of PhoneGenie. The body of the email would look like this:

Updated database successfully...database put CF 701 6781234567

Prerequisites. This software assumes you are using one of the Asterisk aggregations built on CentOS 5. We've tested it with PBX in a Flash. You'll also need an SMTP server (SendMail or Postfix) that is configured to send and receive emails to and from destinations on the Internet. You do not need a POP3 or IMAP mail server! We've tested this with Asterisk 1.4, but it should work fine with Asterisk 1.6 as well. FreePBX 2.5 or later is required for some functions.

Security Warning. Before we begin, let's pause for a moment to review the enormity of your problems if you do this wrong and to remind you that YOU ARE PROCEEDING AT YOUR OWN RISK. PBX in a Flash in particular is shipped with all outside access to your SMTP server blocked. We've obviously got to remove that layer of security for this software to function properly. But you need to be especially careful with SMTP servers because they can be used to relay SPAM to the entire world if you fiddle with settings that you don't understand. So... DON'T MAKE IMPROVEMENTS THAT AREN'T COVERED HERE UNLESS YOU KNOW WHAT YOU'RE DOING!

This software also gives certain email messages elevated privileges on your Asterisk server so that Asterisk itself can be reconfigured. If you compromise the email account name and password for this application, anybody worldwide can pretty much destroy the functionality of your server. In addition, calls to a certain extension could be rerouted to a very expensive destination on a cruise ship sailing around the world. If your dialplan permitted these calls and you had an account with automatic replenishment from a credit card or bank account, you've got a very expensive problem on your hands. That's one reason that reliable email notification of every Phone Genie transaction is critically important. If you're not getting timely notifications of each Phone Genie transaction, DO NOT USE THIS SOFTWARE until that problem is resolved!

Should you detect that your system has been compromised by receiving an email that indicates a command has been executed on your Asterisk server that you did not initiate, you should immediately disable or remove the script so that no further Phone Genie emails are processed on your server. Be sure to preserve any unprocessed Phone Genie emails for authorities as these may contain important information regarding the source of the emails. These email messages usually are deleted once Phone Genie completes execution of the associated Asterisk commands.

Overview. Here's the drill for today. First, we'll adjust both your hardware- based and IPtables firewalls to allow inbound email delivery to your Asterisk server. Second, we'll remove SendMail from your system and install and configure Postfix to handle the SMTP email chores. This will greatly simplify the security issues in locking down your server from unwanted emails. Depending upon your Internet service provider, installation of Postfix may break outbound email delivery from your server if your provider happens to block outbound traffic on port 25. We'll show you how to fix it. Third, we'll add a new user account on your Asterisk server that will be used exclusively to handle Phone Genie messages. Fourth, you're going to need a fully-qualified domain name for your Asterisk server so that email can be delivered reliably to your server. We'll walk you through getting this set up. Fifth, we'll install and configure the Phone Genie software and run some simple tests to make certain everything is working as it should. Sixth, we'll add the Phone Genie script as a cron job which will be run every couple of minutes to check for incoming Phone Genie emails. Finally, we'll review some of the Asterisk commands that can be executed using the Email Edition of Phone Genie for Asterisk.

Security Design. We've obviously given a great deal of thought to the security issues surrounding this application. The security model we've adopted works like this. First, for an email to get through to your Asterisk server, one and only one email address will work from the Internet. All other inbound email from the Internet will be rejected by Postfix. We strongly suggest you leave it that way. Your email address consists of the special username that we will create on your server plus a (hopefully new) fully-qualified domain name that points to your server. You are well advised to use and keep secret both a non-intuitive and complicated username AND a non-intuitive and complicated, fully-qualified domain name. Only this combination will let the email message through the Postfix filter! Using the correct username and a different FQDN that may also point to your server's correct IP address will nevertheless be rejected by Postfix. The third piece in the security model is the password. If you examine the sample Subject above, you will note that it begins with the word "Asterisk" followed by a colon, a space, and then the Asterisk CLI command. The word "Asterisk" is actually the password, and it can be changed to any password you like. So, if you change your password to FooBaR, then the subject of your message should look like this. Note that the colon followed by a space are also required!

FooBaR: database put CF 701 6781234567

Finally, it should be obvious but... DON'T SEND THESE EMAILS FROM AN UNTRUSTED CLIENT OR A PC IN A PUBLIC PLACE because your email message may get stored in a place that someone else could decipher how to access your server. If you wouldn't leave a $1000 bill beside the computer from which you're sending the email, don't send it! Otherwise, you may lose a good bit more than $1,000. To give you some idea of what's at risk with a compromised system, try sending the following email using your correct email address and password:

FooBaR: help

</sermon>

Firewall Configuration. For purposes of our example today, we're assuming that your Asterisk server is sitting behind a hardware-based firewall/router on a private subnet and that your Asterisk server includes a functioning software-based IPtables Linux firewall. This is the default PBX in a Flash setup that we always recommend. On your hardware-based firewall, you will need to redirect incoming TCP port 25 traffic to TCP port 25 on the private IP address of your Asterisk server. This change often requires a reboot of your firewall/router. Once that change is complete, log into your Asterisk server as root and edit /etc/sysconfig/iptables on PBX in a Flash systems. We need to add a new rule to IPtables which allows incoming TCP port 25 traffic through the firewall. Scroll to the bottom of the file and insert the following lines just above the COMMIT line:

# Allow inbound SMTP traffic on TCP port 25
-A INPUT -p tcp -m tcp --dport 25 -j ACCEPT

Save your additions to the file and then reload IPtables and your network:

service iptables stop
service iptables start
service network restart
service iptables status | grep "tcp dpt:25"

The last command should return an entry from IPtables showing TCP port 25 traffic is now being ACCEPTed into the server. If not, check your entries and repeat the process until this works.

Postfix Installation. Let's continue by removing SendMail from your server and installing Postfix. They both perform the same email functions, but the complexity of SendMail makes the likelihood of a configuration error too risky for us to sleep well. If you understand the intricacies of SendMail and feel comfortable implementing the security model we've described above, by all means, have at it. We'll be happy to share your results with the rest of our user community. In the meantime, here's the Postfix solution. While still logged into your server as root, issue the following commands to uninstall SendMail and install Postfix:

rpm -e --nodeps sendmail
yum -y install postfix

Choosing a Username and FQDN. Before we configure Postfix, you need to decide upon a user account name for your Asterisk server to manage Phone Genie messages. And you also need a fully-qualified domain name which points to the public IP address of your Asterisk server. As mentioned above, we strongly recommend that the username and FQDN be obscure and unguessable. For example, a combination of letters and numbers that don't spell words are good choices. Something like dlrpzh7b3@dhf34.nerdvittles.com will help you sleep well. If you don't have a static IP address and dedicated domain for your server that you can manage, then use an equally obscure FQDN from a provider such as dyndns.org. Something like dhf34.dyndns.org works. You then can configure your Asterisk server to automatically keep your dynamic IP address current. We're going to use these entries as examples below. Obviously, you should choose different entries!

To create the new user account on your server using whatever name you have chosen, here are the commands to issue while still logged into your server as root. Just substitute your chosen username for dlrpzh7b3 in both commands. Be sure to choose a secure password, too.

useradd dlrpzh7b3
passwd dlrpzh7b3

Configuring Postfix. Now let's get Postfix set up for maximum protection. First, move to postfix directory: cd /etc/postfix. Now edit main.cf: nano -w main.cf. Search for the inet_interfaces line in the file: Ctrl-W, inet_interfaces =. Add a hash mark to the beginning of each uncommented inet_interfaces line so that your entries look like this:

#inet_interfaces = $myhostname
#inet_interfaces = $myhostname, localhost
#inet_interfaces = localhost

Next, search for mydestination in the file: Ctrl-W,mydestination =. Comment out each of the lines except the one that looks like this:

mydestination = $myhostname, localhost.$mydomain, localhost

Now add the private IP address of your Asterisk server and your FQDN chosen above to the line so that it looks like this. Don't forget the commas and keep everything on one line.

mydestination = $myhostname, localhost.$mydomain, localhost, 192.168.0.118, dhf34.nerdvittles.com

Finally, move to the last line in the file and make it look like this, all on one line:

smtpd_recipient_restrictions = check_recipient_access hash:/etc/postfix/access, permit_mynetworks, reject_unauth_destination

Save your changes to the file: Ctrl-X, Y, then Enter. Now edit /etc/postfix/access. Move to the very bottom of the file and add two new lines with the following entries using the actual email address and FQDN you chose above instead of the examples. The first line tells Postfix to allow emails addressed to the specified email recipient. The next line tells Postfix to reject all other emails addressed to anyone at this domain. Other domains and public IP addressing are blocked by our mydestination entry above.

dlrpzh7b3@dhf34.nerdvittles.com OK
dhf34.nerdvittles.com REJECT recipient rejected

Save your changes to the file: Ctrl-X, Y, then Enter. Now issue the following two commands:

postmap /etc/postfix/access
service postfix restart

Testing Postfix. Now comes the important part. We need to make sure that outbound emails from your Asterisk server are delivered. And we need to make sure that incoming emails ONLY to the one email address you've designated are received and that all other emails from the Internet are rejected. We can't stress enough how important all three of these tests are. If your Postfix implementation doesn't pass all three, DO NOT PROCEED!

Testing outbound email with Postfix is easy. While logged into your server as root, issue the following command using a destination email address (instead of yourname@gmail.com) where you regularly receive emails:

echo "Hi there" | mail -s Test yourname@gmail.com

Count to 20 and refresh your email's Inbox. If the message is there, you've passed Test #1. If not, check your junk mail folder. If it's still not there, try another email address if you have one. Still no cigar? Then your Internet Service Provider is probably blocking email generated from downstream email servers. For tips on remedying the problem, see this message thread on the PBX in a Flash forums. You might also want to review the Postfix tutorial on dyndns.com. Here's another good tutorial on setting up a Gmail relay using Postfix. And here's another excellent tutorial. Then run the test again until you achieve success.

Testing inbound email to your designated email address is Test #2. Use a web client and send an email message to dlrpzh7b3@dhf34.nerdvittles.com substituting the actual email address you have chosen for your server. Count to 20, log into your server as root and type the following command to retrieve email for user dlrpzh7b3: mail -u dlrpzh7b3. The server should report that you have one new message. Type "d 1" and then "q" to delete the message and quit the mail app. If no email arrives, check the Inbox on your sending client to see if the message bounced and, if so, why. Check your email entries in /etc/postfix/access and /etc/postfix/main.cf for typos and review the steps in Configuring Postfix above. Then repeat the test until you successfully send a message to your designated email address.

Testing inbound email to an unauthorized email address on your Asterisk server is Test #3. For this test, we want to make sure that an email sent to the root account on your server fails. What you'll need for this test is the FQDN that was chosen above. Then, using a mail client, send an email message to root@dhf34.nerdvittles.com using your actual FQDN. Count to 20, log into your server as root, and type: mail. The message you sent should NOT be in the Inbox. Repeat the test by sending a message to root and dlrpzh7b3 @the actual IP address of your Asterisk server. These, too, should both fail. Once you get a passing grade on all three tests, we can move on. The hard part is behind you!

Installing Phone Genie. While logged into your server as root, issue the following commands:

cd /root
wget http://pbxinaflash.net/source/nv/phonegenie.tgz
tar zxvf phonegenie.tgz
rm phonegenie.tgz

Configuring Phone Genie. While still logged into your server as root, edit phonegenie.sh. You will note that there are 3 fields that need to be configured at the top of the file: user, pw, and notify. The user field is the designated user account name that will be used for incoming emails (dlrpzh7b3 in our example). The pw field is the word in every email Subject that precedes the colon, space, and Asterisk CLI command (Asterisk in our example). The notify field is a reliable email address where you regularly receive emails promptly. This is where the results of your Phone Genie email commands will be sent. Choose this email address wisely, as if your bank account depended upon it. It does! Once you have filled in the 3 fields (preserving the quotation marks around each entry), save the file with your changes.

Testing Phone Genie. Now we're ready to try everything out. Using an email client, send an email message to dlrpzh7b3@dhf34.nerdvittles.com (using your actual Phone Genie email name and FQDN). For the Subject, enter the following (substituting the password you created above for Asterisk)... Asterisk: help

After counting to 20, log into your Asterisk server as root and issue the following command:

/root/phonegenie.sh

You should see a display of all of the Asterisk CLI commands and within a minute or so, you should receive an email with the same information at the email address you entered into the notify field in phonegenie.sh in the previous step.

Installing Phone Genie as a Cron Job. Once you have tested several Phone Genie emails manually and you're satisfied that everything is working reliably, you can set up the Phone Genie shell script as a cron job. It should be set to execute every minute or every couple of minutes throughout the day and night. Edit /etc/crontab and insert the command shown below to have the script execute every 2 minutes:

*/2 * * * * root /root/phonegenie.sh > /dev/null

Sample Phone Genie Commands. In addition to all of the traditional Asterisk CLI commands, Phone Genie also supports a number of commands that are specific to FreePBX. These additional commands let you configure call forwarding, call waiting, do not disturb, system speed dials, and blacklist entries on your Asterisk server. For Asterisk CLI command syntax, consult voip-info.org. For FreePBX command syntax, see the listing below. Enjoy!

database put CF 302 8338116666 * Call Forwarding Enable
database del CF 302 * Call Forwarding Disable

database put CFB 302 8238221234 * Call Forwarding on Busy Enable
database del CFB 302 * Call Forwarding on Busy Disable

database put CFU 302 8038445689 * Call Forwarding Unavailable Enable
database del CFU 302 * Call Forwarding Unavailable Disable

database put CW 302 ENABLED * Call Waiting Enable
database del CW 302 * Call Waiting Disable

database put DND 302 YES * Do Not Disturb Enable
database del DND 302 * Do Not Disturb Disable

database put blacklist 6781234567 1 * Blacklist a number
database del blacklist 6781234567 * Remove blacklisted number

database put sysspeeddials 99 6781234567 * Set up Speed Dial 99
database del sysspeeddials 99 * Remove Speed Dial 99
(NOTE: Be sure you enable Feature Code *0 prefix in FreePBX!)

We wish all of you a very Merry Christmas!




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.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest...

Ringbinder theme by Themocracy