Category: MP3 Devices

2013 Greatest Hits: Lenny Returns for an Encore Performance

Nothing in the VoIP community this year quite captured the hearts and minds of geeks around the world like Brian West’s “Lenny.” For anyone that’s ever been dogged by obnoxious telemarketers or that’s had to deal with less than lucid tech support inquiries, Lenny was a godsend. Finally, we all had a place to send those poor souls while getting our daily chuckle listening to the results. If you’re late to the party and missed all the fun, then start today by listening to some of the recordings posted on ItsLenny.com and Reddit. Our personal favorite has got to be the “security expert” explaining the discovery of a vulnerability in Lenny’s network:

0:00

As if Brian needed another feather in his cap after FreeSwitch™, what made Lenny an instant hit was the ability to reroute telemarketing and blacklisted callers directly to ItsLenny.com headquarters for processing. The site provided numerous local phone numbers around the world as well as a SIP URI. For those in the PBX in a Flash™ and FreePBX® community, it was especially easy because of the Lorne Gaetz Lenny Blacklist Mod. By simply entering the SIP URI of Lenny, all of your telemarketers were immediately rerouted to Lenny. And then one day, The Music Died.

What? No more Lenny? Were we all destined to return to the screaming monkeys?

Well, not so fast. We got in touch with Brian to inquire about Lenny’s health. Brian explained that he was seeking a more robust home for our pal because of the tremendous response and worldwide usage of the ItsLenny.com site.

Brian also graciously offered permission to use the Lenny recordings for those that wanted to host their own “Lenny” during the interim. And that brings us to today. We’re not sufficiently proficient in FreeSwitch to offer an interim solution on that platform. And, for our shortcomings, we apologize. But what we can do is provide an Asterisk® alternative that you can host on your own server until Lenny returns to his former glory in his new home.

Introducing Lenny Encore! We’ve actually got a number of new creations to introduce today. First, we’ll give you a short law school lesson on the do’s and don’ts of recording phone calls. Second, today’s Lenny Encore dialplan code introduces the Asterisk BackgroundDetect function which actually waits for someone to speak and then proceeds when silence ensues. It’s not perfect, but it helps with applications like this and for applications that seek to detect the presence of answering machines when making robocalls. Third, we’ll show you how to use the Lenny Blacklist Mod in FreePBX to redirect blacklisted callers to any extension you wish rather than merely playing a congestion or Zapateller Special Information Tone (SIT). Fourth, we’ll show you how to record calls in Asterisk with one line of dialplan code. Fifth, we’ll document for the first time how to create a button on almost any SIP phone to reroute ringing (unanswered) incoming calls to another extension. Sixth, we’ll review how to safely set up your own SIP URI and Free DID to enable Lenny Encore access from anywhere. And, finally, we’ll provide you some links to take Lenny Encore for a test drive before you install anything. Please don’t use these links as a destination for your blacklist. The links will only be available for a few weeks. Now let’s get started.

Law School 101: Recording Phone Calls. For openers, this is not legal advice! Consult your own attorney for that. This is merely background information to hopefully alert you to some of the pitfalls which await should you decide to start recording phone conversations. One of the first things you learn in law school is that there’s a difference of legal opinion on almost every topic. That’s why both sides pay lawyers which is a good thing… for lawyers. So it is with the law pertaining to the recording of phone calls. Let’s start with the ABC’s of phone recording. Whether you can legally record a phone call between you and someone else depends upon several things: (A) the location of the person making the call, (B) the location of the person receiving the call, and (C) how the call makes the journey from Point A to Point B.

In some jurisdictions, you probably can’t record a phone call at all because you can’t legally operate an Asterisk server. In other jurisdictions, you can record a call if you give yourself permission to record your conversations with others. In a few jurisdictions (including at least a dozen states in the United States), both parties have to consent before you can record a phone call. In some of those, providing an announcement that you’re recording the call will suffice while in others you have to explain why you’re recording the call and allow the caller to opt out. At least in the United States, if the call crosses state lines then federal law may control; however, there may also be federal agency rules and regulations that impose additional constraints on interstate calls. In law school, there’s a full-semester course devoted to Conflict of Laws. What you need to know is that normally (but not always) the law of the jurisdiction in which the call is initiated controls. Clear as mud? You bet. Here’s the state-by-state and country-by-country breakdown of the rules for those of you that are curious. The moral of this story should be clear:

UNLESS YOUR INITIALS ARE NSA, DON’T RECORD PHONE CALLS UNLESS YOU’VE CONSULTED A LAWYER AND CAREFULLY EXPLAINED WHO THE CALLING PARTIES WILL BE, WHAT YOU INTEND TO RECORD, WHERE EACH POTENTIAL CALLER WILL BE CALLING FROM, WHEN YOU WILL BE RECORDING THE CALLS, WHY YOU ARE DOING IT, AND HOW YOU WILL BE RECORDING THE CALLS. And this isn’t going too well for the NSA either!

6 P.M. UPDATE: A couple of serious bugs were discovered in the initial release. If you’ve already installed Lenny Remake, please replace the original dialplan code using the following commands. Skip this step if you have not previously installed Lenny Remake. The first-time install instructions below have been corrected to remove the problem. Our apologies.


cd /tmp
wget http://pbxinaflash.com/lsupport.tgz
tar zxvf lsupport.tgz
rm lsupport.tgz
sed -i '\:// BEGIN Lenny Remake:,\:// END Lenny Remake:d' /etc/asterisk/extensions_custom.conf
sed -i '/\[from-internal-custom\]/r /tmp/lenny.txt' /etc/asterisk/extensions_custom.conf
rm lenny.txt
rm 3.gsm
asterisk -rx "dialplan reload"
amportal a r

Installing Lenny Encore for the First Time. Now for the fun stuff. We’ve only tested this on PBX in a Flash servers running Asterisk 1.8 and Asterisk 11. For other platforms, there may be some prerequisites that you have to address. On the PIAF platform, log into your server as root. Then create and run a shell script that looks like this:

#!/bin/bash

mkdir /var/lib/asterisk/sounds/lenny
chown asterisk:asterisk /var/lib/asterisk/sounds/lenny
cd /var/lib/asterisk/sounds/lenny
wget http://pbxinaflash.com/Lenny.tgz
tar zxvf Lenny.tgz
rm Lenny.tgz

cd /tmp
wget http://pbxinaflash.com/lsupport.tgz
tar zxvf lsupport.tgz
rm lsupport.tgz
sed -i '\:// BEGIN Lenny Remake:,\:// END Lenny Remake:d' /etc/asterisk/extensions_custom.conf
sed -i '/\[from-internal-custom\]/r /tmp/lenny.txt' /etc/asterisk/extensions_custom.conf
rm lenny.txt
mv 3.gsm /var/lib/asterisk/sounds/lenny
cd /var/lib/asterisk/sounds/lenny
chown asterisk:asterisk *
chmod 755 *

echo " " >> /etc/asterisk/extensions_custom.conf
echo "[bridgit]" >> /etc/asterisk/extensions_custom.conf
echo "exten => 4,1,Pickup(701@from-internal)" >> /etc/asterisk/extensions_custom.conf
echo "exten => 4,2,Pickup(777@from-internal)" >> /etc/asterisk/extensions_custom.conf
echo " " >> /etc/asterisk/extensions_custom.conf

asterisk -rx "dialplan reload"
amportal a r

echo "Try it out by dialing 53669 from any extension on your PBX."

In the [bridgit] section of the code (at the bottom of the script), you’ll see two extensions in bold: 701 and 777. These represent a phone extension and ring group on your server that handle incoming calls from telemarketers. We’ll explain it in more detail shortly. For now, change the numbers to match your setup before you run the script. If you want to manage telemarketing calls from additional extensions with SIP phones, just add additional lines to the [bridgit] context incrementing the line numbers as you go, e.g. 4,3 then 4,4, etc.

Installing Lenny Blacklist MOD. To automatically reroute blacklisted callers to Lenny Encore, you’ll need to modify the blacklist processing setup in FreePBX. To do this, you first have to install the Lennny Blacklist MOD. Download it to your desktop from the Download Now link. Next, add it to FreePBX in the usual way: Admin -> Module Admin -> Upload Modules. Choose the Lenny Blacklist MOD on your Desktop. Once its imported, click on the Local Module Admin link to install and enable it. Once it’s enabled, open it under Other -> Lenny Blacklist MOD. Configure it to match what’s shown below:

Recording Calls with Lenny Encore. By default, Lenny Encore will do its thing with no call recording. If you and your lawyer think recording is a good idea, here’s how to enable it. Log in as root and edit extensions_custom.conf in /etc/asterisk. Simply uncomment the three lines near the top of the file that look like what’s shown below and reload your dialplan:


;exten => 53669,n,MixMonitor(/tmp/Lenny/${RECORDING}.wav)
;exten => 53669,n,NoOp(Recording will be available: /tmp/Lenny/${RECORDING}.wav)
;exten => 53669,n,Playback(en/this-call-may-be-monitored-or-recorded)

This gets the recordings saved to the /tmp/Lenny directory on your server, but these file collections can grow large. We recommend emailing them to yourself in MP3 format once a day and then deleting them. Here’s how to set this up:

cd /root
wget http://nerdvittles.com/convert2mp3.tar.gz
tar zxvf convert2mp3.tar.gz
nano -w convert2mp3.sh

When the editor opens, plug in your email address for delivery of the files and then save the modified script. Now add an entry to /etc/crontab that looks like this:

6 1 * * * root /root/convert2mp3.sh >/dev/null 2>&1

Reroute Ringing Calls to Lenny Encore. We’ve never seen this documented for Asterisk so here’s a bonus for this week. Have you ever wanted to reroute an incoming call to another extension while it was ringing so that you didn’t have to answer, tell the caller to hold, and transfer the call? Well we have, too. That’s especially true in the case of telemarketers and politicians.

As part of the Lenny Encore dialplan code, we’ve added the necessary piece to get this working on many SIP phones with a spare button that can be pressed to dial a number. Many phones call it a Speed Dial entry. Just create a Speed Dial entry for your phone that looks like this:

Now, when the CallerID shows an annoying caller is ringing, just press the Lenny key!

But suppose you want to make this more generic. If you’d like to be able to press the Lenny key and be prompted for the extension number to which to forward the incoming call, then edit the 536691 dialplan code (as we did with call recording) and uncomment the following lines:


;exten => 536691,n,Flite("After the beep, enter extension or press pound for Lenny.")
;exten => 536691,n,Read(SENDTO,beep,7)
;exten => 536691,n,GotoIf($["foo${SENDTO}" = "foo"]?5:6)

If you hit the Lenny key while an incoming call is ringing and enter an extension number followed by #, then that’s where the call will go. If you just hit #, then Lenny Encore gets the call.

Taking Lenny Encore for a Test Drive. We’ve set up a temporary site to let you try Lenny out before installing on your own server. Just call 1-206-424-6913 or use either of the following SIP URIs: 2233435945@sip2sip.info or lenny@nerdvittles.com. Our next article shows you how to do it yourself!

Upgrading Lenny Encore. This project is still a work in progress. What that means is the code is changing almost daily. You can replace your setup with the latest code by following the 6 p.m. update procedure documented above. This will reset your system to NO RECORDINGS in addition to loading the latest dial plan code. Your feedback is, of course, always appreciated. Come join the fun!

More Lenny Encore to Come! Well, that’s enough to keep you busy this week. Next week (now available!), we’ll walk you through setting up a safe SIP URI and free DID to handle inbound calls for Lenny or any other purpose on your PBX in a Flash server.


Deals of the Week. There are a few amazing deals still on the street, but you’d better hurry. First, for new customers, Sangoma is offering a board of your choice from a very impressive list at 75% off. For details, see this thread on the PIAF Forum. Second, a new company called Copy.com is offering 20GB of free cloud storage with no restrictions on file size uploads (which are all too common with other free offers). Copy.com has free sync apps for Windows, Macs, and Linux systems. To take advantage of the offer, just click on our referral link here. We get 5GB of extra storage, too, which will help avoid another PIAF Forum disaster. Finally, O’Reilly has over 1,000 Packt Ebooks on sale for 50% off until August 15. Only 3 days left!

Originally published: Monday, August 12, 2013




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


 

Don’t miss the first-ever FreePBX World on August 27-28 at the Mandalay Bay in Las Vegas. For complete details, see this post on the FreePBX blog.


 

We are pleased to once again be able to offer Nerd Vittles’ readers a 20% discount on registration to attend this year’s 10th Anniversary AstriCon in Atlanta. Here’s the Nerd Vittles Discount Code: AC13NERD.


 
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…

11/11/11: To Celebrate Nerd New Year’s, Please Welcome…

Nerd Vittles Daily Dump

Just click on the image above to visit the site. Content is updated at least twice daily. As always, we welcome your content suggestions. Enjoy!

Originally published: Friday, November 11, 2011


Great News! Google Plus is available to everyone. Sign up here and circle us. Click these links to view the Asterisk feed or PBX in a Flash feed on Google+.




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…

7 Steps to Skytopia: Pain-Free Calls with Skype and Asterisk

As you probably know, Digium® announced that Skype for Asterisk® would not be available for sale or activation after July 26, 2011. Here we are in November. So what to do? If you're looking for a commercial solution, you're S.O.L. But, if you have a non-commercial PBX for personal use1, then keep reading. We'll walk you through, step-by-step, getting Skype integrated into your PBX in a Flash or Incredible PBX environment. It's easy, but it's a manual process. If you follow the steps below in order, you'll be up and running in about 15 minutes.

Prerequisites. For today's project, we're assuming you have an existing Incredible PBX server running CentOS 5.7. If not, here's our tutorial to get you up and running quickly. You'll also need a keyboard, mouse, and monitor. We strongly recommend a dedicated server such as an Atom-based PC. If you're using a virtual machine, then you'll need a sound card alternative. Try this: /sbin/modprobe snd-dummy.

UPDATE: We've revised this article a bit to accommodate PIAF2 with CentOS 6.2 and Incredible PBX 3. Keep in mind that Skype is a 32-bit application so we strongly recommend a 32-bit platform if reliability matters to you.

Step 1. For inbound Skype calling to work with other implementations including generic PBX in a Flash systems, you'll need to create a SIP URI for your Asterisk server: mothership@127.0.0.1. You do NOT need to expose the SIP port(s) of your Asterisk server to the Internet, and we strongly recommend that you don't! We've previously explained how to set up a SIP URI in this article. The Incredible PBX includes this SIP URI functionality out of the box.

Step 2. You'll also need Java 1.5. To see if it's included in your distribution, issue the following command: rpm -q jdk. If your particular Asterisk distribution doesn't have JAVA 1.5 or higher installed (rpm -q jdk), here's how to do it. Go to the Oracle Technology Network, sign up for a free Oracle web account and log in. While still logged in, accept the binary code license agreement, and click on this link to download jdk-6u12-linux-i586-rpm.bin. Then copy the file to /root on your Asterisk server. Make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin) and then run it. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Whew!

Step 3. You'll also obviously need a dedicated Skype account for your Asterisk server. If you don't have one to spare, download the Skype software for your Mac or Windows PC, and sign up for a free account. You can try out your account by calling our demo hotline: nerdvittles. Get this working on your Mac or PC before proceeding! Then be sure you log out and disable automatic logins on reboot, or you'll have a problem down the road with two machines trying to log in to a single Skype account.

Step 4. Now we're ready to install the remaining software components that your server will need to access Skype. Log into your Asterisk server as root and issue the following commands.

cd /root
mkdir skype
cd skype
wget http://download.skype.com/linux/skype_static-2.1.0.47.tar.bz2
tar jxvf skype_static*
yum -y install xorg-x11-server-Xvfb
yum -y install qt4
yum -y install xterm
yum -y install libXScrnSaver.i386 < == use this for CentOS 5.x
#yum -y install libXScrnSaver <== use this for CentOS 6.x
wget http://incrediblepbx.com/siptosis.tgz
cd ..
wget http://incrediblepbx.com/skype-start
chmod +x skype-start
cp skype-start skype/.
cd /
tar zxvf /root/skype/siptosis.tgz
cd /root/skype

If you'd prefer to avoid all the typing, you can issue the following commands to download a script that will do all the heavy lifting for you. This is for CentOS 5.x systems only:

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

For PIAF2 systems running CentOS 6.x, use this instead:

cd /root
wget http://incrediblepbx.com/skype-setup2
chmod +x skype-setup2
./skype-setup2

Step 5. Now there are a few steps to manually configure the software components so that the entire Skype startup process can be automated when your server boots in the future. To begin, you'll need to fire up X-Windows which puts your server in graphics mode. This is the only mode that Skype understands. While logged into your server as root, issue the following command: xinit

NOTE: If xinit won't start on your particular machine, you may need to create /etc/X11/xorg.conf. Here's a generic config file that should work fine for CentOS 5.x systems:

Section "ServerLayout"
Identifier "X.org Configured"
Screen 0 "Screen0" 0 0
EndSection

Section "Device"
Identifier "Card0"
Driver "vesa"
EndSection

Section "Screen"
Identifier "Screen0"
Device "Card0"
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
EndSection

For PIAF2 users, some have reported issues on Atom machines with seeing a display at all after xinit loads. If this happens to you, don't panic. Simply log into your server from a PC or MAC using SSH. Then run: vncserver :1. Set a password for VNC, and then use a VNC client on your PC or Mac to access VNC at the IP address of your server on display port 1. Now you can continue with Step 6, below.

Step 6. Now we're ready to start up Skype, and get it properly configured. There are two important requirements. First, we want to make sure your credentials are saved for automatic login in the future. And second, we want to configure Skype to run in a minimized state each time it restarts. To begin, click in the white graphics window on your screen using your mouse and issue these commands:

cd /root/skype/skype_static-2.1.0.47
./skype

Click on the Accept button to accept the Skype license agreement. Once Skype loads, enter your Skype Name and Password. Before clicking on Sign In, be sure to check the Automatic Sign In box so that you'll be logged in automatically in the future. Once you're logged in, click on the blue S in the lower left corner of the window to access the Skype Main Menu. Then click Options. When the General tab displays, check the box which says Start Skype minimised in the system tray. Then click the Apply button. To test things out, click on the Sound Device tab and then Make a Test Call. Once you're sure everything is working, click the Close button. Now click on the blue S again and click Quit to shut down Skype.

Step 7. Now we're ready to integrate Skype into the SipToSis middleware so that Asterisk can communicate with Skype. Issue the following commands to start Skype in background mode and then start SipToSis. Be sure to write down the PID for Skype in case we need to kill the app if something goes wrong.

./skype &
cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype. Before clicking Yes, be sure to click the Checkbox to Remember This Selection for future connections! When you click Yes, you'll see the SipToSis CLI indicating that it's waiting for a Skype call.

If you've installed this on an Incredible PBX, Skype should now be functional. From another Skype account, just call the Skype Name that you used to set this up, and your Asterisk extensions should start ringing. To test outbound Skype calling, use an X-Lite softphone connected to an extension on your Asterisk server and dial *echo123 to access Skype's call testing service or *nerdvittles to access our demo.

All that remains is to configure your server to automatically start Skype and SipToSis whenever your system is restarted. Here's how. Press Ctrl-Alt-F2 to get a new login prompt on your server. Log in as root and issue the following command:

echo "/root/skype-start" >> /etc/rc.d/rc.local

Now reboot your server and make sure everything is working.

Navigation Tips. Here are a few navigation tips for managing your Asterisk console on CentOS systems once Skype has been installed:

1. Ctrl-Alt-F2 gets you a new login prompt for your server

2. Ctrl-Alt-F7 gets you back to the SipToSis/Skype session. You can kill SipToSis by holding down Ctrl-C for several seconds. To find the Skype PID: pidof skype. To kill Skype: kill pid#. To restart Skype: skype & and to restart SipToSis, just issue the command again: ./SipToSis_Linux

3. Ctrl-Alt-F9 gets you to the Asterisk CLI.

Setting Up Speed Dials for Skype Friends. One of the wrinkles with Skype is that Skype uses names for its users rather than numbers. If you don't have a SIP URI-capable softphone, there's still an easy way to place calls to your Skype friends using FreePBX®. Just add a Speed Dial number to your FreePBX dialplan. Choose Extension, then select the Custom type, provide an Extension Number which is the Speed Dial number (this could actually spell your friend's name using a TouchTone phone), enter a Display Name for your friend, and add an optional SIP Alias. Then insert the following in the dial field replacing joeschmo with your friend's actual Skype name. Save your entries and reload the dialplan when prompted.
SIP/joeschmo@127.0.0.1:5070

Security Warning. One final note of caution. Do NOT expose UDP port 5070 to the Internet unless you first secure this port with a username and password to avoid Internet intruders using your gateway as a free Skype dialing platform! You do not need 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your Asterisk server so we recommend you keep it securely behind at least a hardware-based firewall.

FreePBX Design. For those not using Incredible PBX, here is the FreePBX setup that Incredible PBX uses and that we recommend. For outbound Skype calls, you have two choices.

1. To place a call to a regular phone number using SkypeOut (which costs you money), you'll simply dial 8 plus the area code and number. Our foreign friends will have to adjust their dialplans and /siptosis/SkypeOutDialingRules.props accordingly. Today's setup assumes 10-digit phone numbers!

2. To place a call to a Skype username using a softphone that supports SIP URI dialing such as X-Lite, you simply precede the Skype username with an asterisk, e.g. *echo123 will connect you to the Skype Call Testing Service or *nerdvittles will connect you to the Nerd Vittles Skype demo.

For incoming Skype calls, the default setup routes those calls to a SIP URI: mothership@127.0.0.1. Whether you point this URI to an extension, ring group, or IVR is up to you. In the default Incredible PBX build, the mothership URI is pointed to the Stealth AutoAttendant, an IVR that plays a welcoming message and then transfers the call to a ring group if no digit is pressed by the caller.

Configuring FreePBX. To put this setup in place, use a web browser to access FreePBX on your Asterisk server. You'll need to create a Custom Trunk and then an Outbound Route.

1. Choose Setup, Add Trunk, Add Custom Trunk. Fill in the form so that it looks like the following using your own CallerID number obviously:

When you're finished, click the Submit Changes button and then reload the dialplan when prompted.

2. Next choose Setup, Outbound Routes, Add Route. Fill in the form so that it looks like this:

When you're finished, click the Submit Changes button. Be sure to move this new OutSkype route to the top position in your Outbound Routes listing in the right margin! Then reload the dialplan when prompted.

3. If you're not using Incredible PBX, add a new DayNight Control 1 option while you're still in FreePBX. Just specify where you want calls routed for Day mode and Night mode. Then, here's the easy way to activate SIP URI support on your Asterisk/FreePBX server. Copy the [from-sip-external] context from the extensions.conf file in /etc/asterisk. Now copy the content into extensions_override_freepbx.conf. Be sure to preserve the context name in brackets! On a FreePBX 2.8 system, make it look like the following. The additions we're making are shown in bold below:

[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but
; I do know what I'm doing. This is correct.
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => mothership,1,Goto(app-daynight,1,1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)

Finally, reload your Asterisk dialplan, and we're finished with Asterisk and FreePBX setup:

asterisk -rx "dialplan reload"

Fedora Builds. For those using recent Fedora builds, these systems have a full implementation of X-Windows and KDE. Just start the system in mode5 (graphics mode), log in, run Skype in one window and start up SipToSis in a terminal window using the commands in Step 7 above. Authorize external use of Skype when prompted.

Where To Go From Here. Well, those are the basics. You now can make one outbound Skype call at a time from your Asterisk server, and you can receive an inbound Skype call on any Asterisk extension when Skype users call your regular Skype name. If you want multiple Skype account support, then you'll need to do some tweaking. What you'll need is the STS Trunk Builder toolkit which is free, but proprietary. Enjoy!

Originally published: Tuesday, November 1, 2011


Great News! Google Plus is available to everyone. Sign up here and circle us. Click these links to view the Asterisk feed or PBX in a Flash feed on Google+.




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. Excerpt from the Skype Terms of Service: "Subscriptions are for individual use only. Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means). Each subscription is to be used for your own personal communication purposes only, to make calls to another individual. The use of the subscription for commercial gain, such as calling numbers specifically to generate income for yourself or others by placing such calls, is not permitted. Unusual call patterns may be considered indicative of such use and may result in us terminating your subscription and blocking your User Account in accordance with paragraph 11.2." []

Android 3 Deal of the Year: Acer Tab for Under $300

We’ve never done back-to-back reviews of similar devices, but this week’s Target ad changes all of that. As you might expect, Acer has covered all of the bases with their entry into the dual-core Android 3 tablet sweepstakes. You may recall that we weren’t huge fans of the Motorola Xoom which promised a lot and delivered a boatload of vaporware. The Acer Iconia Tab A500 is not the Xoom. You not only get a microSD slot and Flash that actually work, but Acer has thrown in an HDMI port that can output 1080p video as well as a USB port that lets you connect your favorite USB devices including external hard disks. It performs this magic with an 8-10 hour battery life. And this week (only at Target) you can pick up this WiFi-only device for half the cost of the Motorola Xoom. In fact, after the gift card, it’s only a dollar more than the single-core Vizio Tablet that we reviewed last week.

Update: See the comments for equivalent deals just announced at NewEgg and CompUSA.

It’s difficult to describe the feel of the Acer Tab. Suffice it to say, it’s dimensions coupled with its sleek and sculpted design put it in the league with the iPad2 unlike the Xoom which felt chunky and clunky despite being an ounce lighter than the Acer.

As we mentioned last week, we don’t dive too deeply into the technical weeds in our reviews. If you want the technical assessment, check out this PC World review. What we prefer to evaluate is real-world usage of these devices. The Acer Tab has stunning performance. In addition to reading email and browsing the web, here’s the suite of applications which we think matter to most folks. We want to watch videos from YouTube and NetFlix. We want to stream music from Google Music and Spotify and read our Kindle books. We like to use Skype. And, yes, we also like Flash video support which works perfectly on the Acer tablet.

In addition to running Android 3, the Acer Tab boasts impressive hardware specs running a 1GHz Nvidia Tegra 250 dual-core processor with 1GB of RAM and 16GB of ROM. Add another 32GB easily with the microSD slot. The 10.1-inch tablet has a 1280-by-800 pixel display with a 16:10 aspect ratio that’s perfect for HD video content. We always prefer testing devices with real-world video content that we’ve shot so we can compare it to performance on other devices. Our Pawleys Island Parade video didn’t disappoint. It’s performance and color were as good or better on the Acer Tab than on Apple’s top-of-the-line 27″ iMac featuring a quad-core 2.93 GHz Core i7 processor with 8GB of RAM plus L2 and L3 cache. The same can be said with playback of complex Flash video. Netflix unfortunately is still a few weeks off although rooted Acer devices reportedly run it just fine.

On the music front, it doesn’t get much better than the Acer Tab. With Google Music or Spotify, the music world is your oyster. And the silver lining is that the Acer Tab is the one and only device that includes Dolby Mobile audio. Once you adjust the equalizer to match your taste in music, you’ll have sound quality to match that 20-pound boombox gathering dust in your basement.

In the communications department, Skype performed well although video calls are not yet supported. That’s unfortunate given the impressive specs on the Acer Tab’s two cameras. The Iconia Tab has a 5-megapixel rear-facing camera with flash in addition to a 2-megapixel front-facing camera for video conferencing. Finally, making and receiving free phone calls using either an Asterisk® server with CSipSimple or Google Voice using a $50 Obihai device and the free ObiON client for Android both worked great.

There’s only one word you’ll need to remember to take advantage of this Target deal: H-U-R-R-Y! This is a one-week only special, and Target offers no rainschecks. So call around until you find one. You won’t be sorry. And, as usual, Target offers a 90-day, no questions asked return policy which is second to none.

Google+ Invites Still Available. Need a Google+ invite? Drop us a note and include the word “Google+” and we’ll get one off to you. Come join the fun!

Our Favorite Android Apps. We’ve listed a few of our favorite apps below for those just getting started with Android. Enjoy!


Originally published: Tuesday, August 16, 2011




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…

How Good Can a $298 Android Tablet Be?

Pretty damn good in the case of the new 8″ Vizio Tablet. While it’s not going to take any speed awards when compared with the new Galaxy Tab 10.1, it does have a 1GHz processor with 512MB of RAM which delivers respectable performance with incredible battery life that rivals any iPad. Storage capacity is limited to 2GB, but you can add a 32GB microSD and meet any computing demands you may have. Currently the device is WiFi only.

As you might expect, Vizio knows a thing or two about televisions, and there’s a silver lining with the Vizio Tablet. Not only is an IR blaster included in the hardware, but you also get a giant TV remote that controls any combination of TVs, cable and satellite boxes, DVD and BluRay devices, and about 95% of the other video and audio components you will find on the planet. And it works as well or better than any of the pricey, high-end touchscreen (with a little screen) TV remotes that would easily put you in the Poor House. Say goodnight, Logitech. There’s also a front-facing 640×480 camera which easily suffices for video conferencing. No current video conferencing apps work, by the way, but it’s only been on the street for a week. The best news of all, you can pick one up at Costco or WalMart if you want one today. Or order it from Amazon if you prefer tax-free.

We don’t dive too deeply into the technical weeds in our reviews. If you want the technical assessment, check out this SlashGear review. What we prefer to evaluate is real-world usage of these devices. The Vizio Tablet passes with flying colors. In addition to reading email and browsing the web, here’s the suite of applications which we think matter to most folks. We want to watch videos from YouTube and NetFlix. We want to stream music from Google Music and Spotify and read our Kindle books. We like to use Skype. Sorry, Apple, we also like Flash video support which works perfectly on the Vizio Tablet even though it’s currently running Gingerbread.1

Last, but not least, being a phone nerd, we obviously want to make and receive free phone calls using either an Asterisk® server with CSipSimple or Google Voice using a $50 Obihai device and the free ObiON client for Android. Both work great!

Of course, the usual Android favorites including Google+ with the exception of (the currently non-functioning) Huddle for video conferencing with up to 10 participants, Maps, Navigation, and Google Talk all work flawlessly. Gallery is perfectly synched with your Picasa photo collection which now can store unlimited photos at no cost through Google Plus. If you want to actually take professional photographs and make feature films, this isn’t the device for you. With the exception of Skype which is not yet available for this device (which was just released), everything else we’ve mentioned works great especially if you’re living on a budget. And, with the addition of Huddle in Google+, the absence of Skype support really doesn’t much matter any more. If you happen to need a Google+ invite, here’s a link compliments of Nerd Vittles. Finally, and pardon us for repeating, if you’re sick of wrestling with a half dozen remotes to watch television, this device is worth its weight in gold. You’ll be asking yourself why no one but Vizio was smart enough to think of it.

Vizio also had a better idea when it came to the Android user interface. As you can see in the photo above, there’s a top section where you can install your Favorite Apps. Immediately below that is your entire Applications collection. At the very bottom, there are five buttons which you can assign to your Must-Have Apps such as email, your web browser, the Google Market, Settings, and whatever else you happen to like.

Another nice touch that hasn’t been mentioned in many of the reviews is that Vizio has added a new keyboard option. If you remember the ergonomic keyboards that had the keys divided into two sections, Vizio has done much the same thing on the touchscreen which greatly improves typing for those that actually learned how. This keyboard, of course, can be toggled on and off depending upon your personal taste.

In conclusion, we think Vizio has hit a home run with this device. The price point, the feature set, the form factor, and the incredible battery life are just about perfect. We’ve listed a few of our favorite Android apps below to get you started. Enjoy!


Originally published: Wednesday, August 10, 2011




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. Honeycomb has been promised for down the road. []

Welcome to Frontier Days

One of my favorite vacations as a kid was spent enjoying Frontier Days in Cheyenne, Wyoming. If you’ve never been with your family, you need to add this to your Bucket List. It’s a week-long celebration that you’ll never forget. To commemorate this year’s event which is going on right now, we decided to celebrate by staging our own Frontier Days here at Nerd Vittles. It provides you an opportunity to join with us in kicking the tires of all the new stuff we’re working on this summer to write about in the fall. In the grand tradition of Cheyenne’s Frontier Days, expect a wild ride! If you’re a bit squeamish about knowing how sausage is made, today’s introduction to new projects may not be your cup of tea. For the pioneers, it’s Party Time! So let’s get started.

Introducing Asterisk 10. At the top of our list is the brand new Asterisk®, formerly known as Asterisk 1.10. You’ll want to read Kevin Fleming’s announcement of the name change, and then read Malcolm Davenport’s summarization of the new product. Here are a few excerpts:

A major focus of the Asterisk 10 development cycle was Asterisk’s support for media types. In versions of Asterisk 1.8 and prior, Asterisk supported a rather limited number of codecs due to some architectural limitations. Plumbing was ripped out, kitchens were remodeled, girders were swapped, and Asterisk 10 now has a media architecture that’s capable of handling both a nearly unlimited number of codecs as well as codecs with more complex parameters…

Asterisk 10 [also] provides basic video conferencing support. That’s right, if you and your friends have video-capable SIP devices, that all speak the same video codec and profile, you can create multi-party video conferences.

Asterisk 10 can also improve your faxing experience. Asterisk 1.4 is capable of T.38 pass-through, where one T.38 capable endpoint can send a fax directly to another T.38 capable endpoint – usually a couple of SIP peers. Asterisk 1.6.X and 1.8 are capable of T.38 termination, where Asterisk can read/write TIFF files from/to T.38 endpoints. Now, with Asterisk 10, transparency between non-T.38 and T.38 is possible.

Whenever there are major plumbing changes, there usually are some major surprises awaiting those of us that depend upon Asterisk to actually make calls. That’s where you come in. Tom King has quickly put together a new PBX in a Flash 1.7.5.6.3 ISO that includes PIAF-Red, aka the new Asterisk 10. We encourage you to try it on a non-production machine, and report any problems both to us (on the PIAF Forum) and to Digium® (in the Bug Tracker). Here’s a download link to get you started. Here’s the new Cepstral TTS installer.

Introducing Incredible PBX 2.0. Frontier Days wouldn’t be complete without a new version of Incredible PBX. In this beta release, we’ve reworked Google Voice support and added one of the most requested features, the ability to enter dial strings for trunks in outbound routes the old-fashioned way.

On the Google Voice front, we’ve replaced the hard-coded Google Voice code in Incredible PBX 1.8 with Marcus Brown’s new FreePBX® module. It not only makes Google Voice usage optional, but it also lets you add and remove multiple Google Voice trunks to your heart’s content. And the setup process takes less than a minute to enter your credentials.

Incredible PBX 2.0 also includes Andrew Nagy’s new Swiss Army Knife Module for FreePBX. This module adds some of the most requested features that currently are missing from FreePBX 2.8 and 2.9:

  • Export a CSV file of your Dial Patterns from Outbound Dial Plans
  • Use Textbox Dial Patterns for Outbound Routes
  • Modified Blacklist Module allowing any value, not just numbers
  • Coming Soon: reg-exp black/white list module

If you’d like to take Incredible PBX 2.0 for a spin, here’s a download link with instructions. Be aware that this version is NOT suitable for use on any system that is not also protected by a hardware-based firewall. For example, don’t use it on a hosted server such as RentPBX.com just yet. We use a different security model on hosted and cloud-based systems, and it is NOT included in this build. Finally, Incredible PBX 2.0 is not yet compatible with Asterisk 10 and PIAF-Red, but we’re working on it.

Introducing Google+. Unless you’ve been sleeping under a rock, you probably have heard that Google has a new little product of its own. In less than 3 weeks, Google+ has grown to over 20 million users, and it’s still by invitation only. You can read our writeup of it on Nerd Vittles. Suffice it to say, it is a game changer for those of us in the technology business. It’s an almost perfect tool for carrying on a problem-solving dialog, and we plan to make extensive use of it in coming months to support PBX in a Flash and Incredible PBX. Don’t be shy. We’ve got plenty of invites. All you have to do is drop us a note and include the word Google+ so we’ll know what you need. We’re turning requests around in less than a day. One final hint. Use your real name on Google Voice, or the Soup Nazi may remove your account. It’s become a bit of a brouhaha at the moment… as one might expect during Frontier Days.

Introducing OS X Lion. Apple has not been asleep at the wheel either. Their new operating system release is extraordinarily good but only available as an over-the-air update to an existing OS X 10.6.8 system. You can read our writeup of the gotchas for a quick and painless install. And, if you’re in the market for a new notebook, we can’t say enough good things about the new MacBook Air. It’s in a league of its own.

Introducing Google Chromebooks. Last but not least, we need to say a few words about the amazing new Chromebooks running Google’s Chrome OS. As with cellphones, Google is not making the hardware. So you have a choice of Samsung or Acer at the moment. The Samsung model starts at $429 for the WiFi only model. The comparable Acer machine is $80 cheaper. We opted for the Samsung WiFi machine which is well made, has an incredible battery life, and just works. For 95% of what we do, it’s a perfect device. There’s a short list of gotcha’s. First, you’ve got to have network connectivity since everything is cloud-based. Second, if your requirements include a lot of graphics manipulation and editing, this probably is not the machine for you quite yet. Finally, if movies (NetFlix) and music (Spotify) are must-have’s, you’d better wait a month or two until those products are available for the Chromebook. Google Music, which allows you to put your own music collection in the cloud, works fine today! There’s an add-on extension to Chrome for Google Voice. As of yesterday, it works flawlessly to make and receive calls. In summary, if your computing requirements primarily involve surfing the web, email, and SSH, then you’re going to be very happy with the Chromebook.

In our case, we’re trying to alternate our use between a Chromebook and the new MacBook Air. So far, we’ve been very satisfied with both. And the Chromebook is 1/4 the cost! Pioneers Forever! Enjoy!

Originally published: Tuesday, July 26, 2011




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…

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