Posts tagged: sip

Smartphone Trifecta: 2016’s Very Best Cellphones with Two Awesome Surprises


Every year we try to check out the latest and greatest smartphones with emphasis on finding those that are the best fit with Asterisk®. So this year is really special because our three favorite new phones all come with a couple of surprises. First, monthly cellular service can be FREE on all of them! Second, all of the phone numbers associated with the three phones can be used as free SIP trunks with Incredible PBX™ or your favorite Asterisk server.

If you’ve been following Nerd Vittles since early February of this year, then you already know that RingPlus, a Sprint MVNO, is the best bargain on the planet. Over the past six weeks of weekly specials from RingPlus, we’ve managed to update all of our free RingPlus accounts to either unlimited calling, texting, and 2GB of monthly data or 3,000 minutes of calling, 3,000 text messages, and 3GB of data. For anybody (except teenagers) that’s sufficient monthly capacity to do almost anything you’d like to do with a smartphone except stream movies all day.

We initially showed how to take dirt cheap Boost Mobile and Virgin Mobile prepaid phones and repurpose them for use with RingPlus. Sprint apparently read our article as well because that loophole is going away on April 17. However, you still have time to find one and activate it on RingPlus following our previous tutorial. The only catch is that, if you ever deactivate it, you will lose the ability to reactivate it without first using it with Boost or Virgin for a full year. The landfills will be so happy with all these cellphone bricks because of Sprint’s latest attempt to shoot itself in the foot. We think there also are some legal issues that the FCC needs to address. These phones are sold as “contract-free” when, in fact, there is a very specific and undisclosed contractual requirement. If you don’t keep service with the provider for a year, your phone becomes a brick. In antitrust terminology, it’s called tying. And some would argue that it also constitutes false advertising. We plan to file a complaint and would urge all of our readers to do the same. Here’s a link.

But enough about the Sprint mentality. It really is legendary, and it’s been the same for 20+ years. We doubt it will ever change unless the entire Sprint management team is replaced. So where do we go from here? Well we decided to upgrade most of our phones to the latest and greatest postpaid phones available, and we wanted to try out our 2016 favorites (pictured above). Here’s some really great news. Samsung’s new Galaxy S7 and S7 Edge as well as Apple’s new iPhone SE work swimmingly with RingPlus as long as you purchase the Sprint-branded models at full retail price from either Best Buy or an Apple Store. Sprint and Target will refuse to sell you one unless you activate it with Sprint in the store. You also can’t buy the Sprint-branded iPhones on line from Apple without activating Sprint service, but that restriction doesn’t apply if you visit an Apple Store.

It took a week to chase down a Galaxy S7 and almost two weeks to find a Galaxy S7 Edge at a Best Buy store. Don’t believe the store inventory on their web site. Neither of the phones we purchased was shown as available at the locations where we bought them. So you’ll need to call or visit a store at least while the new Galaxy phones remain a scarce commodity. As for the iPhone SE, it went on sale at Apple Stores this morning at 10 a.m. At the Charleston store, I was third in line and both of the people in front of me also were buying the Sprint-branded iPhone SE to use with RingPlus. The Apple sales folks said they had never before seen a run on Sprint phones. Guess why?

Here’s the drill. Purchase your favorite phone after you read our mini-reviews below. Don’t open the box just yet. Instead, look on the bottom of the box and decipher the IMEI/MEID of your phone. Immediately run that number through the RingPlus Device Checker to be sure it will work on the Sprint network using a RingPlus account. There shouldn’t be a problem with any of these three phones, and all of them come with a Sprint SIM card so you won’t have to worry about obtaining one from RingPlus. Some have reported that the Best Buy phones were locked. We can only surmise that the customer delayed activating the phone with RingPlus which gave Sprint time to block the serial number which Best Buy reported. If this happens to you, we are told that Sprint will unlock the phone once you provide proof that it was purchased at full retail price. If all else fails, Best Buy has a 14-day return policy. Remember, anything is possible when dealing with Sprint.

Once your phone passes the compatibility check, sign up for a new free RingPlus plan. These plans change weekly and sometimes are only offered for a couple of hours so you may want to hold off on signing up until a deal comes along that meets your requirements. Update: There are a number of excellent promotions at the moment which run through April 5. Our previous article explained in detail how these free plans work. Switching plans typically is limited to those that buy into the annual Member+ program. You can read all about the plans and programs on the RingPlus Community Forum. If you already have a RingPlus account with a registered phone, you can swap out the phone with one of these three new ones for a one-time charge of $1.99. All you’ll need is your new MEID and ICC ID numbers. The entire phone swap only takes a minute or two. Once it’s complete, turn on your phone. The rest is automagic!

Comparing the Phones. We don’t often glow about reviews, but the TechRadar review of the Galaxy S7 Edge is a must-read. There has never been a better phone than this one. And, only an inch behind it is the Galaxy S7 which bears an uncanny resemblance to the new iPhone SE except for its 50% larger screen size. We actually are more comfortable carrying the Galaxy S7 with its all-metal construction. For whatever reason, the S7 Edge always feels like its about a millisecond away from slipping out of your hand. You will most definitely want a case for the S7 Edge.

In terms of performance and camera quality, the new Galaxy phones are in a league of their own. Here’s a photo hurriedly snapped through a restaurant window with our Galaxy S7 earlier this week. If you’ve ever tried to take sunset pictures with an iPhone or cheapo Android device, you’ll appreciate what a challenge these shots can be. We’ll annotate this article with an iPhone SE photo if and when the opportunity presents itself. The other good news with the new Galaxy phones is they are at least waterproof for a few minutes. If you live near the water, that will come as a welcome addition as well. Finally, Samsung has closed the gap with Apple’s iPhones on backing up and restoring everything on your phone. For years, this has been Apple’s best feature in our humble opinion. Now Samsung goes Apple one better. If you happen to have two Samsung devices that you want clone, simply choose Backup and Reset from Settings. Then Open Smart Switch on both devices and hold the two phones back to back. It’s that easy. Or you can opt for the more traditional restore method that works precisely as it does with an iPhone using the Samsung Cloud. For some additional tips and tricks, visit the PCMag.com site and watch the video “Exploring the Galaxy S7” which includes a number of comparisons with Apple iPhone devices including the iPhone SE. Enjoy!

We previously covered the SIP setup for RingPlus devices using their WiFi Fluidcall feature. It provides a free SIP trunk for Asterisk at a cost of zero dollars. For the complete tutorial, take a look at the original article. Enjoy!

Originally published: Thursday, March 31, 2016





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


 
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    The Sensational Raspberry Pi 3 Featuring Incredible PBX with Raspbian 8 Jessie

    Hard to believe it’s been four years since the introduction of the original Raspberry Pi®. Over eight million RasPi’s have been shipped. To celebrate its fourth birthday, Eben Upton has done it again. Meet the sensational Raspberry Pi 3 sporting a 1.2GHz 64-bit quad-core ARM Cortex-A53 CPU with ten times the performance of the original Raspberry Pi. Of particular interest to the VoIP community will be the RasPi 3’s integrated 802.11n wireless LAN and Bluetooth 4.1 hardware. And, of course, the RasPi 3 retains its compatibility with the Raspberry Pi 1 and 2. Did we mention it’s still just $35? Because we like to celebrate birthdays, too, we’re pleased to introduce a brand new Incredible PBX™ image for the Raspberry Pi 2 and 3 featuring Raspbian 8 and the latest release of Asterisk® 13. Unlike previous builds, this one installs in under a minute. Yes, it’s still FREE and features pure open source GPL code. No Gotchas!

    Raspberry Pi 3 Performance. Gone are the days of worrying about Raspberry Pi performance. Both the user interface and call quality now match what you’d expect to find on a $300-$500 VoIP server. Even with a Raspberry Pi 2, we have detected no performance degradation thanks to the latest Raspbian 8 OS and a virtually flawless Asterisk 13 platform. For best results, we recommend 32GB Class 10 microSD cards which now are plentiful at the $10 price point.1

    Raspberry Pi 3 Shopping List. Before you can install Incredible PBX, you’ll need a compatible Raspberry Pi 3 platform. Here’s the short list:

  • $35* Raspberry Pi 3 from MCM or Newark or Amazon
  • $10 Power Adapter (2.5 amps minimum!)
  • $9 32GB microSDHC Class 10 card
  • £12.95 Rainbow Pibow case or $9.50 Official RasPi case
  • About That Asterisk. We write about Asterisk® regularly, but the asterisk we’re talking about is the one accompanying the $35* price tag for the Raspberry Pi 3. Yes, that’s the advertised price. And, no, if you want one this year, you’re not going to pay that. There are the marked up shipping prices, the bundled add-on’s that you don’t need or want, and the must-have accessories like a power adapter. We’re assuming you already own a USB keyboard and an HDMI-compatible monitor. If so, just plan on $100 and consider yourself lucky if you get all the pieces for less. Our order from Pimoroni in the U.K. with a case and 3-day shipping was £59.36 or $82.95 U.S. Our order from MCM for just the RasPi 3 with shipping was $46.99.

    Incredible PBX Feature Set. Where to begin? Let’s start with the Alphabet Stew: IAX, SIP, GVSIP, SMS, and SRTP functionality. Voice Recognition and Text-to-Speech VoIP application support using FLITE, GoogleTTS, and PicoTTS. Free calling with Google Voice, Simonics SIP gateway, or RingPlus cellular service. And all of your Nerd Vittles favorites: Fax, AsteriDex, Click-to-Dial, News, Weather, Reminders, and Wakeup Calls. Plus hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and AutoAttendants, Email Delivery of Voicemail, Voicemail Blasting, and more…

    10-Layer Network Security Model. Most phone calls cost money. Unlike many of the other “free” VoIP solutions, our most important criteria for VoIP is rock-solid security. If your free server ends up costing you thousands of dollars in phone bills due to fraud, it isn’t free at all. Once you plug in that network cable, you’ve painted a bullseye on your checkbook.

    No single network security system can protect you against zero-day vulnerabilities that no one has ever seen. Deploying multiple layers of security is not only smart, it’s essential with today’s Internet topology. It works much like the Bundle of Sticks from Aesop’s Fables. The more sticks there are in your bundle, the more difficult it is to break them apart. If a vulnerability suddenly appears in the Linux kernel, or in Asterisk, or in Apache, or in your favorite web GUI, you can continue to sleep well knowing that other layers of security have your back. No one else in the telecommunications industry has anything close. Ours is all open source GPL code so we would encourage everyone to get on board and do their part to make the Internet a safer place!

    Do your part and do your homework. Comparison shop as if your phone bill matters! 😉 Incredible PBX provides:

    1. Preconfigured IPtables Linux Firewall
    2. Preconfigured Travelin’ Man 3 WhiteLists
    3. Randomized Port Knocker for Remote Access
    4. TM4 WhiteListing by Telephone (optional)
    5. Fail2Ban Log Monitoring for SSH, Apache, Asterisk
    6. Randomized Ultra-Secure Passwords
    7. Automatic Update Utility for Security & Bug Fixes
    8. Asterisk Manager Lockdown to localhost
    9. Apache htaccess Security for Vulnerable Web Apps
    10. Security Alerts via RSS Feeds in Kennonsoft and Incredible PBX GUIs

    Installation Tutorial. Here’s everything need to know about installation and setup. “Automatic” means you just watch.

    1. Download and unzip Incredible PBX image from SourceForge (with or without GV OAuth support)
    2. Transfer Incredible PBX image to microSD card
    3. Boot Raspberry Pi from new microSD card
    4. Login to RasPi console as pi:raspberry to initialize your server (Automatic)
    5. Reboot after writing down your server IP address (Automatic)
    6. Login via SSH as root:password to randomize passwords & configure firewall (Automatic)
    7. Optionally, install Incredible Fax: /root/incrediblefax13_raspi3.sh (Credentials: admin:password)
    8. Enjoy!

    Configuring Trunks with Incredible PBX

    Before you can actually make and receive calls, you’ll need to add one or more VoIP trunks with providers, create extensions for your phones, and add inbound and outbound routes that link your extensions to your trunks. Here’s how a PBX works. Phones connect to extensions. Extensions connect to outbound routes that direct calls to specific trunks, a.k.a. commercial providers that complete your outbound calls to any phone in the world. Coming the other way, incoming calls are directed to your phone number, otherwise known as a DID. DIDs are assigned by providers and you register your trunks using credentials handed out by these providers. Incoming calls are routed to your DIDs which use inbound routes telling the PBX how to direct the calls internally. A call could go to an extension to ring a phone, or it could go to a group of extensions known as a ring group to ring a group of phones. It could also go to a conference that joins multiple people into a single call. Finally, it could be routed to an IVR or AutoAttendant providing a list of options from which callers could choose by pressing various keys on their phone.

    We’ve done most of the prep work for you with Incredible PBX. We’ve set up an Extension to which you can connect a SIP phone or softphone. We’ve set up an Inbound Route that, by default, sends all incoming calls to a Demo IVR. And we’ve built a dozen trunks for some of the best providers in the business. Sign up with the ones you prefer, plug in your credentials, and you’re good to go.

    Unlike traditional telephone service, you need not and probably should not put all your eggs in one basket when it comes to telephone providers. In order to connect to Plain Old Telephones, you still need at least one provider. But there is nothing wrong with having several. And a provider that handles an outbound call (termination) need not be the same one that handles an incoming call (origination) and provides your phone number (DID). We cannot recommend Vitelity highly enough, and it’s not just because they have financially supported our projects for almost a decade. They’re as good as VoIP providers get, and we use lots of them. If you’re lucky enough to live in the U.S., you’d be crazy not to set up a Google Voice account. It’s free as are all phone calls to anywhere in the U.S. and Canada. The remaining preconfigured providers included in Incredible PBX are equally good, and we’ve used and continue to use almost all of them. So pick a few and sign up. You only pay for the calls you make with each provider so you have little to lose by choosing several. The PIAF Forum includes dozens of recommendations on VoIP providers if you want additional information.

    With the preconfigured trunks in Incredible PBX, all you need are your credentials for each provider and the domain name of their server. Log into Incredible PBX GUI Administration as admin using a browser. From the System Status menu, click Connectivity -> Trunks. Click on each provider you have chosen and fill in your credentials including the host entry. Be sure to uncheck the Disable Trunk checkbox! Fill in the appropriate information for the Register String. Save your settings by clicking Submit Changes. Then click the red Apply Config button.

    Configuring a Softphone for Incredible PBX

    We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.

    We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. Choose Applications _> Extensions -> 701 and write down your SIP/IAX Password. You can also find it in /root/passwords.FAQ. Fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password you assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

    Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:

    DEMO - Apps Demo
    123 - Reminders
    947 - Weather by ZIP Code
    951 - Yahoo News
    *61 - Time of Day
    TODAY - Today in History

    If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store.

    Configuring Google Voice

    If you want to use Google Voice, you’ll need a dedicated Google Voice account to support Incredible PBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

    IMPORTANT: Do NOT under any circumstances take Google’s bait to switch from Google Chat to Hangouts, or you will forever lose the ability to use Google Chat with Incredible PBX. Also be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. Good News! You’re in luck. Google has apparently had a change of heart on discontinuing Google Chat support so it’s enabled by default in all new Google Voice accounts. Once you’ve created a Gmail and Google Voice account, go to Google Voice Settings and 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
    • Call Options (Enable Recording)OFF
    • Global Spam FilteringON

    Click Save Changes once you’ve adjusted 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.

    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!

    Now you’re ready to configure your Google Voice account in Incredible PBX. You can do it from within the Incredible PBX GUI by choosing Connectivity -> Google Voice. How you enter your credentials depends upon whether you have chosen the Incredible PBX image with OAuth 2 support. For a complete Google Voice OAuth tutorial, follow steps 8-10 in this Nerd Vittles tutorial. Once you’ve entered your credentials, you MUST restart Asterisk from the command line, or Google Voice calls will fail.

    If you have trouble getting Google Voice to work (especially if you have previously used your Google Voice account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.

    If you have difficulty finding the Google Chat option after setting up a new Google Voice account, follow this tutorial.

    Another option is to use an inexpensive SIP Gateway to Google Voice. The Simonics trunk in the Incredible PBX GUI is preconfigured for this purpose. All you’ll need is your Google Voice credentials. Get started with this tutorial.

    Adding Speech Recognition Support to Incredible PBX

    To support many of our applications, Incredible PBX has included Google’s speech recognition service for years. These applications include Weather Reports by City (949), AsteriDex Voice Dialing by Name (411), and Wolfram Alpha for Asterisk (4747), all of which use Lefteris Zafiris’ terrific speech-recog AGI script. Unfortunately (for some), Google now has tightened up the terms of use for their free speech recognition service. Now you can only use it for “personal and development use.” If you meet those criteria, keep reading. Here’s how to activate speech recognition on Incredible PBX. Don’t skip any steps!

    1. Using an existing Google/Gmail account to join the Chrome-Dev Group.

    2. Using the same account, create a new Speech Recognition Project.

    3. Click on your newly created project and choose APIs & auth.

    4. Turn ON Speech API by clicking on its Status button in the far right margin.

    5. Click on Credentials in APIs & auth and choose Create New Key -> Server key. Leave the IP address restriction blank!

    6. Write down your new API key or copy it to the clipboard.

    7. Log into your server as root and edit the speech recognition script:

    nano -w /var/lib/asterisk/agi-bin/speech-recog.agi
    

    8. When the nano editor opens, go to line 70: my $key = "". Insert your API key from Step #6 above between the quotation marks and save the file: Ctrl-X, Y, then Enter.

    9. To use Wolfram Alpha by phone, you first must obtain a free Wolfram Alpha APP-ID. Then issue the following command replacing APP-ID with your actual ID. Do NOT change the yourID portion of the command:

    sed -i "s|yourID|APP-ID|" /var/lib/asterisk/agi-bin/4747
    

    Now you’re ready to try out the speech recognition apps. Dial 949 and say the name of a city and state/province/country to get a current weather forecast from Yahoo. Dial 411 and say “American Airlines” to be connected to American.

    To access Wolfram Alpha by phone, dial 4747 and enter your query, e.g. “What planes are overhead.” Read the Nerd Vittles tutorial for additional examples and tips.

    Enabling WiFi on the Raspberry Pi

    With the Raspberry Pi 3, wi-fi hardware is included. With the Raspberry Pi 2, you’ll need to add an inexpensive wifi dongle. The next step is connecting to your WiFi router. Simply open /etc/wpa_supplicant/wpa_supplicant.conf with your favorite editor and insert the following code using the actual SSID name and password to access your local, password-protected WiFi router or any open WiFi network:

    network={
     ssid="YourSSID"
     psk="YourSSIDpassword"
     key_mgmt=WPA-PSK
     scan_ssid=1
     priority=5
    }
    
    network={
     key_mgmt=NONE
     priority=1
    }
    

    Finally, stop and restart the wlan0 interface, count to 15, and check the status of your server to decipher the new IP address for your WiFi connection:

    ifdown wlan0
    ifup wlan0
    pbxstatus
    

    If you want to run your Raspberry Pi exclusively off the WiFi connection, simply unplug the network cable from your RasPi and reboot your server.

    UPDATE: There still is a quirk with the wireless LAN driver on the Raspberry Pi 3. The problem has to do with the default power management of the wlan0 interface which results in it being powered off after very brief periods of inactivity. Special thanks to Matt Gemmell for this fix. Just cut-and-paste the lines below into a terminal window, and you’ll be good to go.

    WARNING: Run pbxstatus first. If the top line shows Raspberry Pi 3, the following WiFi patch is already installed.

    echo "options 8192cu rtw_power_mgnt=0 rtw_enusbss=0 rtw_ips_mode=1" > /etc/modprobe.d/8192cu.conf
    sed -i '/exit 0/d' /etc/rc.local
    echo "sleep 10" >> /etc/rc.local
    echo "iwconfig wlan0 power off" >> /etc/rc.local
    echo "exit 0" >> /etc/rc.local
    echo "[Unit]" > /etc/systemd/system/root-resume.service
    echo "Description=Turn off wlan power management" >> /etc/systemd/system/root-resume.service
    echo "After=suspend.target" >> /etc/systemd/system/root-resume.service
    echo "" >> /etc/systemd/system/root-resume.service
    echo "[Service]" >> /etc/systemd/system/root-resume.service
    echo "Type=simple" >> /etc/systemd/system/root-resume.service
    echo "ExecStartPre= /bin/sleep 10" >> /etc/systemd/system/root-resume.service
    echo "ExecStart= /sbin/iwconfig wlan0 power off" >> /etc/systemd/system/root-resume.service
    echo "" >> /etc/systemd/system/root-resume.service
    echo "[Install]" >> /etc/systemd/system/root-resume.service
    echo "WantedBy=suspend.target" >> /etc/systemd/system/root-resume.service
    systemctl enable root-resume
    reboot
    

    After rebooting, if you issue the iwconfig wlan0 command, it should show: Power Management:off.

    Update: Lessons Learned for Raspberry Pi 3 Road Warriors

    As with all new devices, you learn some things as you go along. So we’re providing an update to our original article to offer a couple of additional tips and tricks for those that want to travel with a RasPi…

    Alternative Power Sources. If you’re like us, you have a number of devices around the house or office that all require 5V power adapters of various amperages. The Raspberry Pi has traditionally been one of the most temperamental when it came to power adapters and, with the Raspberry Pi 3, the developers specifically mention a 2.5 amp minimum. If you travel and want to take devices such as the Raspberry Pi with you, the last thing you want to do is approach airport security with a bunch of wires hanging out of your carry-on bag. Well, there’s good news. The Anker device shown in the Amazon ad in the right column of Nerd Vittles can supply power to 6 devices including a Raspberry Pi 3. And we’ve given the RasPi a healthy workout with no adverse effects.

    Deciphering the RasPi IP Address. As we mentioned, we travel a lot so obtaining a DHCP address for your RasPi in WiFi mode is not always the easiest thing to accomplish. If your smartphone supports tethering, that’s the easiest way to get connectivity on the road. A better way is to stick a WiFi HotSpot in your luggage and it, too, can be powered using the Anker device. See our recent article for WiFi HotSpot choices. Regardless of which option you choose, it will require some planning to use your RasPi sans monitor and keyboard. First, you need to preconfigure /etc/wpa_supplicant/wpa_supplicant.conf with the SSID of the device you’ll be using to hand out DHCP addresses. You’ll note from the discussion above that each entry in this file has a priority with higher numbers having higher priority. The way we typically do this is to assign our home network as the highest priority. Below that, we set up credentials for our MiFi Hotspot, then our smartphones, and finally open networks. So it looks like this:

    • Home Network – 6
    • MiFi HotSpot – 5
    • Android phone – 4
    • iPhone (AT&T) – 3
    • Open Network – 1

    Keep in mind that the Incredible PBX firewall probably will block you from accessing the RasPi from a computer on the public network. So you also must connect your computer to the same private WiFi network because private LAN addresses are whitelisted in the firewall by default.

    Once you have connectivity for your RasPi and your laptop, the other wrinkle is figuring out the IP address of the Raspberry Pi. Our recommended approach goes like this. First, configure SendMail on the RasPi to use a Gmail account that you own as an SMTP smarthost to send emails. That should work almost anywhere you go. Second, modify /etc/rc.local to automatically send you an email with the IP address and SSID of your wireless network whenever the RasPi boots. Again, this takes some advance planning because you need to set all of this up and test it before you go on the road.

    Here are the steps to modify SendMail to use an existing Gmail account as a SmartHost. Log into your RasPi as root and issue the following commands:

    cd /etc/mail
    hostname -f > genericsdomain
    touch genericstable
    makemap -r hash genericstable.db < genericstable
    mv sendmail.mc sendmail.mc.original
    wget http://nerdvittles.dreamhosters.com/pbxinaflash/source/sendmail/sendmail.mc.gmail
    cp sendmail.mc.gmail sendmail.mc
    mkdir -p auth
    chmod 700 auth
    cd auth
    echo AuthInfo:smtp.gmail.com "U:smmsp" "I:user_id" "P:password" "M:PLAIN" > client-info
    echo AuthInfo:smtp.gmail.com:587 "U:smmsp" "I:user_id" "P:password" "M:PLAIN" >> client-info
    echo AuthInfo:smtp.gmail.com:465 "U:smmsp" "I:user_id" "P:password" "M:PLAIN" >> client-info
    nano -w client-info
    

    When the nano editor opens the client-info file, change the 3 user_id entries to your Gmail account name without @gmail.com and change the 3 password entries to your actual Gmail password. Save the file: Ctrl-X, Y, then ENTER.

    Now issue the following commands. In the last step, press ENTER to accept all of the default prompts:

    chmod 600 client-info
    makemap -r hash client-info.db < client-info
    cd ..
    make
    sed -i 's|sendmail-cf|sendmail/cf' /etc/mail/sendmail.mc
    sed -i 's|sendmail-cf|sendmail/cf|' /etc/mail/sendmail.mc
    sed -i 's|sendmail-cf|sendmail/cf|' /etc/mail/Makefile
    sed -i 's|sendmail-cf|sendmail/cf|' /etc/mail/sendmail.cf
    sed -i 's|sendmail-cf|sendmail/cf|' /etc/mail/databases
    sed -i 's|sendmail-cf|sendmail/cf|' /etc/mail/sendmail.mc.gmail
    sed -i 's|sendmail-cf|sendmail/cf|' /etc/mail/sendmail.cf.errors
    sendmailconfig
    

    Next, edit /etc/hosts and /etc/hostname. Change the raspberypi3 entries to: raspberrypi3.incrediblepbx.com.

    Finally, stop and restart SendMail and then send yourself a test message. Be sure to check your spam folder!

    /etc/init.d/sendmail stop
    /etc/init.d/sendmail start
    apt-get install mailutils -y
    echo "test" | mail -s testmessage yourname@yourdomain.com
    

    The last step is to add these commands to /etc/rc.local to send you an email with your IP address and SSID whenever the RasPi is rebooted. Insert the following commands just above the exit 0 line at the end of the file. Use an email address to which you have access on the road!

    ESSID=`iwconfig | grep ESSID | tail -1 | cut -f 9 -d " "`
    echo "IP address: $(hostname -I) on $ESSID" | mail -s "RaspberryPi3 IP Address" yourname@yourdomain.com
    

    Enabling Bluetooth on the Raspberry Pi


    Incredible Fax Returns for the Raspberry Pi


    Mastering the Incredible PBX Feature Set

    Now would be a good time to explore the Incredible PBX applications. Continue reading there. If you have questions, join the PBX in a Flash Forums and take advantage of our awesome collection of gurus. There’s an expert available on virtually any topic, and the price is right. As with Incredible PBX, it’s absolutely free. Enjoy!

    Originally published: Monday, March 7, 2016  Updated: Saturday, March 26, 2016


    Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address 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 Forum. 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, the PIAF Forum 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.


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    1. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. []

    I Have A Dream: Free Cellular Service with Integrated Remote SIP Connectivity

    As part of our Mobile Internet adventure this year, we’ve been scouring the countryside with two requirements in mind. First, we wanted a smartphone on which we could activate some type of free cellular service for making calls and sending text messages. Second, we wanted to integrate remote SIP connectivity using the same provider and phone number so that we could make and receive calls transparently using any SIP phone or Asterisk® server anywhere in the world. Sounds like a tall order, you say? Well, if you’ve enjoyed your Cloud@Cost Sandbox, you’re gonna love RingPlus!

    Yes, you’ll have to buy a compatible cellphone, but there are thousands to choose from. And, yes, you’ll need Sprint service in your neighborhood. Then you’ll have to cough up $10 to activate your cellular account. RingPlus offers dozens of plans.1 We recommend the Michelangelo plan which best meets what we’re trying to accomplish today, but the choice is all yours.2 With the Michelangelo plan, you can make and receive 1,000 minutes of free calls a month to anywhere in the U.S. (calls to Canada are 3¢ a minute), you can send and receive 1,000 free text messages a month, and you can use 500MB of free data service every month. You also can use your same account credentials with any SIP phone, softphone, or Asterisk server anywhere in the world to make and receive phone calls transparently using the same phone number as your smartphone. In other words, you can travel anywhere and make and receive phone calls just as if you were sitting in Atlanta, Georgia dialing from your smartphone. The SIP calls are deducted from your free minutes. No cellular service required at all. Meet RingPlus!


    So what’s the catch? How does RingPlus make money? Well, of course, they would prefer that you sign up for a plan with monthly fees. For those on the free plans, the only difference you will notice is an occasional ad which plays instead of a ring tone when you place outbound calls. This only occurs until the other party answers the call, and it can be all but eliminated by choosing a music selection in the RingPlus Radio feature in your RingPlus Dashboard.

    Who are the ones most likely to use something like this? Well, for openers, all of your kids unless you like springing for a $500 phone and spending $40+ dollars a month for cellular service for each of them. One of the other real beauties of RingPlus is you can set up a whitelist of numbers that can be called from the phone. Blacklists are supported as well. It’s perfect for kids just getting started with a cellphone. A second potential user group would be those who travel outside the United States and prefer not to pay exorbitant roaming rates for calls. Using a SIP phone connected to your RingPlus account, all of the international calls suddenly are free. And the calls are delivered with the same CallerID number as calls placed from your actual smartphone. In fact, your smartphone doesn’t have to be in service at all. A third and perhaps most important use for us was to serve as a failover trunk on one or more Asterisk servers. When all else fails, you can route outbound calls to your RingPlus SIP trunk for free calling using your RingPlus account. Doesn’t get any better than that.

    Official RingPlus WARNING: Starting April 17, 2016, per our carrier partner Sprint, Members and potential Members will no longer be able to activate prepaid devices which are not eligible under Sprint’s FED policies [Requires activation of prepaid phone on original Sprint MVNO network for at least one year!]. Such prepaid devices will no longer pass FED until actual eligibility date is met.

    There are probably numerous ways to put all these pieces in place so that things function just as we’ve described. Today we’ll share with you the solution that actually worked for us. You can take it from there and avoid the thousands of horror stories about incompatible smartphones. Be advised that acquiring used cellphones or even incompatible cellphones is a very dangerous and expensive business. If you buy one that happens to be stolen, or that has a balance due on the account, or that is incompatible with RingPlus, then you’ve bought a tiny boat anchor and not much else. So, our best advice is buy one from the provider. That’s the one and only RingPlus, and the smartphones start at just under $100. Many Sprint post-paid phones also work, such as the new iPhone SE (Sprint Model) from any Apple Store.

    If store employees will let you, find the Sprint postpaid phone that you like and look on the bottom of the box. There you will find the decimal value of the MEID. Log into http://nerd.bz/nvringplus and plug in the MEID to see if it is RingPlus compatible. If it passes, buy it. If it flunks, try another one. Whatever you do, DON’T BUY A PHONE IN AN OPENED BOX, AND DON’T OPEN THE BOX YET! Make certain there is a return policy in case things don’t work out as expected!

    Funny story. The Radio Shack employees at our local store were very savvy and refused to let me look at the MEID claiming it was a security issue. Fair enough. Of course, they were also curious why I wanted a phone without letting them configure it. Once I told them the deal, they all wanted one, too. They asked for the link to the MEID verification site and said they’d do it for me. Once it worked, excitement broke out in the room with all the staff reading an early copy of this article. While Radio Shack typically charges a $35 restocking fee on cell phones, that fee is waived if you return the phone in an unopened box. So the only thing you’re wasting if they insist that you purchase the phone is a little bit of your time and a lot of Radio Shack employee time if, in fact, the MEID flunks the verification test.

    Configuring Your Phone for RingPlus Service

    Now sign up for a RingPlus free plan using the MEID and ICC ID you previously verified. Michelangelo is probably the best bet if you missed our Twitter tip this past weekend. Deposit $10 in your new account, and activate it. Log into your RingPlus Dashboard, click on your phone in the upper right frame, and choose Manage Device. Write down your MSID, your phone number, and MSL. Once your account is active, then and only then unbox and turn on your phone. Go through the minimal setup steps by choosing your Language and choosing an available WiFi network. During this setup, RingPlus should push a PRL update to your new phone, and it will reboot. Check in Settings -> General -> About Phone -> Status and see if you have a phone number. If so, you’re good to go. If not, open the Phone Dialer application and dial ##72786# which should force another PRL update to your phone with another reboot. When it finishes, check again for a phone number and place an outbound call.

    Using a browser on your desktop computer, go back into the RingPlus Dashboard and sign in. Your phone device should show Active in the upper right corner of the screen. Click there and you’ll get a display like this:

    While still in the Device Settings Menu, click on the WiFi FluidCall option to decipher your SIP credentials. You’ll need these to set up your SIP phone or a SIP trunk on your Asterisk server. Your username is your 10-digit phone number, the domain name is sip.ringplus.net, and the password is a system-generated entry which you can recreate whenever you like. That’s probably a very good idea whenever you use public WiFi services to make calls with your SIP phone or a softphone.

    By the way, this isn’t some kludgy SIP-GSM gateway where the calls actually are routed out through your cellphone device. The RingPlus SIP gateway connects your SIP device directly to the Internet and simply uses your existing RingPlus CallerID to identify the calls. In short, you get the best of both worlds: a dirt cheap or free cellphone service plus a dirt cheap or free SIP trunk for use anywhere in the world.

    Configuring a RingPlus SIP Trunk with Asterisk

    If you’d like to set up your RingPlus number as a failover trunk on your Asterisk server, here is the setup that worked for us with Incredible PBX using your assigned 10-digit phone number for your username and fromuser settings and your assigned password for your secret. If you include a registration string and configure an inbound route using your RingPlus DID, then inbound calling will work as well. If you skip the registration step, then you can use the same RingPlus trunk on multiple Asterisk servers for emergency outbound calling. No firewall adjustments should be necessary.

    There are all sorts of other magic tricks you can implement using the RingPlus API, but you probably won’t need any of the features in light of the robust SIP connectivity RingPlus provides to an existing Asterisk server where the feature set is virtually unlimited. Be advised that you must make a call out at least once every 60 days to keep your account active. The simple way to do this is to set up a monthly reminder using your RingPlus trunk. Schedule the reminder to call out once every month using Telephone Reminders in Incredible PBX.

    RingPlus Gotcha Checklist

    Free service wouldn’t be free without a few land mines. So here’s a checklist to keep things running smoothly without any problems down the road. First, link your account to one of the social media options (Twitter, Facebook, or LinkedIn) when you sign up for service. You’ll find the link on your Dashboard under the Your Social Networks icon. Second, make at least one outbound call a month on every line you activate. As noted, this can be accomplished automatically using the Telephone Reminders application in Incredible PBX. Third, keep a valid credit card on file in your account at all times. Fourth, keep a positive balance in your account for each phone that you activate to avoid automatic replenishment at the original rate when you signed up for your plan. Fifth, be mindful of the Domino Effect. With some plans, if you allow a related plan to end (for example, Queen of Hearts when you also have an Ace of Hearts plan), then your better plan will be demoted in its feature set. Enjoy the Free Ride!

    Originally published: Monday, February 8, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    1. Be advised that future upgrades of these “free” plans may go away after February 15 unless you join the Member+ program, the cost of which changes almost weekly. This will not affect those that already are participating in the program according to RingPlus. []
    2. In case you’re curious, a plan equivalent to the free Michelangelo plan at RingPlus would run you $41.00 per month at Ting. Ouch! []

    Just in Time for Santa: Return of The Glory Days with Skype Connect for Asterisk?

    You’ve been good boys and girls all year, and today we have some great news for Asterisk® lovers. Skype is back! Oh, if it were only that simple. But let’s revel in the good news for a bit. Microsoft introduced Skype Connect™ about 5 years ago. Now it’s a SIP interface to Skype. And today we’ll take a fresh look at whether it’s a good fit with Asterisk. Skype Connect is part of Skype Manager™, a carefully considered and beautiful product offering that showcases Microsoft’s UI design skills. After shelling out our weekly allowance to join the party, we were ready to go. Here’s a quick overview from Microsoft:

    Skype Connect provides connectivity between your business and the Skype community. By adding Skype Connect to your existing SIP-enabled PBX, your business can save on communication costs with little or no additional upgrades required.

    With Skype Connect, your business can make great value Skype calls and receive calls from your customers using your desk phones. Customers can also contact your business for free by using Skype from a browser with Skype buttons, by calling [not for free] the Skype business accounts associated with your SIP-enabled PBX, or [by placing PSTN calls to Skype Numbers you may have purchased].

    In addition to an Asterisk server, here’s what you’ll need to get started. First things first, sign up for a Skype Manager account if you don’t already have one. It’s easy and it’s free. Once you’re signed up and logged in, you’re going to need a little cash in your Skype credit account to get things going. $30 will get you started but finish reading the article before you invest.

    Configuring Skype Connect for Asterisk

    To get started, click Features in the toolbar, choose Skype Connect and click Set up a SIP Profile. Give the profile a name “SOHO Inc.” and click Next. Next, choose the number of Channels you need at $6.95 per month. A channel gets you one simultaneous call in or out of Skype. Two channels gets you one call in and one call out simultaneously for $13.90 per month. You can take it from there but, sorry, you can only buy 300 channels at this time. You can add the U.S. Minute Bundles, and we’ll explain that in a minute.

    Don’t buy your channels just yet. For now, cancel out of the dialog by clicking Back. Microsoft will set up your profile anyway:

    The money deposited into your Skype Manager account will be needed to fund Skype Connect in three separate ways: (1) monthly payments for Channels at $6.95 each, (2) monthly payments for Phone Numbers associated with those Channels at $6.30 each, and (3) allocation of funds in advance to pay for outbound calls from each profile you create. You’ll need at least one phone number (a.k.a. DID) to receive any inbound calls from POTS phones to the Skype Connect SIP account on your Asterisk server. You’ll also need at least one phone number before you can assign a CallerID to your outbound calls.1 Otherwise, they go out as Anonymous calls. Outgoing and incoming calls using traditional Skype Names are not supported.

    Once you get your finances in order, it’s time to set up your SIP credentials for your new profile. Click on Authentication Details to display the dialog. Leave the Registration tab highlighted, and click on Generate a New Password, and a new SIP password will be sent to the email address you used to register when you set up your Skype Manager account.

    Configuring Asterisk for Skype Connect

    On your Asterisk server using your favorite GUI, create a new SkypeConnect SIP trunk with your CallerID and the number of channels you’ve paid for. For the Dialed Number Rule: Prepend: +1, Prefix: 759, Match Pattern: NXXNXXXXXX. Insert the following OutGoing Settings in PEER Details. Use skypeconnect for the peer name and your actual SIP user number and password from Microsoft:

    username=990xxxxxxxxxxx
    secret=YourRealPassword
    type=peer
    qualify=yes
    insecure=invite
    host=sip.skype.com
    fromdomain=sip.skype.com
    disallow=all
    allow=ulaw
    context=from-trunk
    fromuser=990xxxxxxxxxxx
    

    For the Register String, it’s your SIPusernumber:password@sip.skype.com/SIPusernumber

    Finally, create an Incoming Route for your SIPusernumber and tell the GUI where to route the incoming calls. Create an Outbound Route for SkypeOut with a pattern of 759NXXNXXXXXX that points to your skypeconnect trunk. Calls can be placed by dialing the 759 prefix plus a 10-digit number. Adjust as necessary to meet your international requirements.

    A Cost-Benefit Analysis of Skype Connect

    If you’ve followed along so far and done the math for yourself, you’ve quickly discovered that Skype Connect’s beauty may only be skin deep depending upon your calling patterns. Let’s give Microsoft the benefit of the doubt and assume that they’re using first rate SIP trunks to carry your calls. Here’s our review of how Skype Connect stacks up to the competition.

    Vitelity is one of our corporate sponsors. Their SIP trunking services are by no means the cheapest on the planet, but you get what you pay for so we’ll use them to compare prices against Skype Connect. For openers, if you haven’t figured it out already, Skype Connect doesn’t bear much resemblance to the Skype of yesteryear. It is essentially a pay-as-you-go SIP trunking service with very few of the historical benefits of Skype. None of the benefits are documented! According to Microsoft, no free calls except with Skype Buttons. This requires a web development effort and limits callers to browser-based phone calls, not exactly ideal. There’s another wrinkle. It doesn’t work. Skype URIs might, but we didn’t test it. No ability to call existing Skype users is supported except those that have purchased a $6.30/month telephone number to associate with their Skype account. And then you pay for the call… by the minute. There is a silver lining, however. By examining the Skype Connect logs, we discovered that Microsoft internally forwards incoming calls to DIDs back into Skype Connect account numbers before processing the calls. That suggested that Microsoft was using these account numbers for internal call routing. And, sure enough, that is the case. Although undocumented, existing Skype users can dial your Skype Connect account number with a + prefix, and the call will be connected to Skype Connect at no cost (see below). If your Skype Connect SIP trunk is registered to an Asterisk server, then the calls will flow directly into Asterisk.

    Our attempts to apply a similar methodology using a remote SIP client, however, failed.2 Others have claimed it works or at least did at one time. Both direct calling approaches eliminate the need for Skype users on BOTH ends of a call to purchase dedicated phone numbers from Microsoft and to pay for long distance calls. The fact that Microsoft has chosen not to document this suggests that free Skype calling to Skype Connect using Skype clients may be short-lived. For today using Skype clients (only), calls will connect using our documented methodology.

    Using the Nerd Vittles special Vitelity signup link below, $3.99 a month buys you a DID in your choice of area codes, unlimited incoming calls, and four channels. This means you can receive four simultaneous incoming calls without any caller receiving a busy signal. Now for the math. Identical service with Microsoft’s SIP trunking service and four channels would run you $34.10 per month, nearly 10 times the cost of Vitelity for comparable SIP service. That’s before you place your first outbound call.

    Let’s consider some examples that factor into the outbound calling equation. For outbound calls, Microsoft wins if you only make tons of calls within the continental United States only. A U.S. bundle of 5,000 minutes runs $30 with Microsoft.3 That is a bargain at .6¢/min. if you use all 5,000 minutes every month. You can buy one bundle for each channel purchased. Vitelity’s rate to the continental U.S., Hawaii, and Canada is 1.44¢ per minute which works out to $72 for the same 5,000 minutes. Change the call mix to Canada only, and the Microsoft rate skyrockets to $115 while the Vitelity rate stays the same.

    Using a more typical SOHO or home calling pattern of 2,000 outbound minutes a month, the Vitelity rate is $28.80 while Microsoft’s rate is $16. Combining the trunk charges, the Vitelity total comes to $32.75 while Microsoft comes in at $50.10. Translation: With the same trunks, channels, a single DID, and 2,000 minutes of outbound U.S. only calls, Vitelity saves you about a third of the monthly cost of the identical Microsoft configuration. For inbound only calling without factoring in free inbound Skype calls, Vitelity saves you 88%. For Canada calling with 2,000 minutes a month, Vitelity saves you about half.

    Your actual costs obviously will vary depending upon the mix and number of simultaneous inbound/outbound calls as well as the origin and destination of the calls. For home and SOHO organizations, Skype Connect rarely will be your best choice unless you get a lot of calls from Skype users around the world. In that case, $6.95 a month for a Skype Connect channel (and nothing else) would be a bargain. For the most part, Microsoft’s focus seems to be larger organizations. For U.S.-based organizations that make substantial numbers of outbound calls to U.S. destinations, Skype Connect also could be financially attractive because of the U.S. calling bundles.

    For an interesting look at Microsoft’s future in the telecom space, read this article.

    Q: Is Skype Connect a good value?

    A: It depends! Do the math. YMMV!

    Originally published: Monday, December 21, 2015





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    1. According to this article, phone numbers registered to your company can also be used as a CallerID number. []
    2. Attempts to register using a free DID at IPkall also were unsuccessful. []
    3. In the fine print, Microsoft discloses that nearly 100 six-digit dialing prefixes in the U.S. are excluded from the bundle pricing. Download the User Guide here. []

    VoIP’s Dirty Little Secret: Why ‘Unlimited’ SIP Trunks Are a Very Bad Deal


    The snazzy ads and free sign-up offers make so-called Unlimited SIP Trunks sound appealing. Let’s take a careful look at what a service such as SIPStation™ would actually provide and compare prices with what’s offered by providers such as Vitelity. Vitelity’s rates are competitive with those offered by many SIP providers as detailed in this PIAF Forum thread.

    Full Disclosure: Vitelity is a Platinum Sponsor of Nerd Vittles™ and our open source projects including PBX in a Flash™ and Incredible PBX™. We also happen to like their business practices and recommend them without hesitation.

    First, a couple of upfront gotcha’s to keep in mind. SIPStation trunks are touted as unlimited. Realistically, they’re limited in a number of ways. For openers, you can only make or receive ONE call at a time unlike trunks provided by most SIP providers that typically offer multiple channels for simultaneous calls. Second, you usually can’t spoof the CallerID number on all-you-can-eat trunks unlike the trunks offered by many providers. We’ll explain why that matters in a minute. Third, if you believe these one-call-at-a-time unlimited trunks provide truly unlimited calling, we’ve got some swamp land in Florida that may be of interest. Leave your trunks off-hook for 2 weeks playing music on hold and see how long your account lasts.

    One of the real beauties of VoIP technology and Asterisk® is that you can choose different providers to handle your incoming and outgoing calls. And you can choose still other providers to handle outbound calls in specific countries to take advantage of better calling rates. With a service such as SIPStation, you’re back to the old Ma Bell days, only worse. One incoming call means nobody else can receive an incoming call until the first caller hangs up or until you buy another $25 trunk. 1 It also means that no one else in the organization can make a simultaneous outbound call without buying additional trunks. At least with Ma Bell, you got call waiting. No such luck here. Another similarity to Ma Bell: the price tag.

    Now let’s suppose that your hardware store or restaurant needs four lines and 90% of the call traffic is incoming calls. With SIPStation, the monthly cost will be over $100. With a single Vitelity trunk and the PBX in a Flash special pricing, your cost for the phone number and four incoming calls at a time is $3.99 a month including 911 emergency service. That’s a 2500% price difference. And while you’d have to pay by the minute for the outgoing calls at a little less than a penny and a half a minute, in most businesses it amounts to chump change. So, unless your organization happens to make substantially more outgoing calls and makes several thousand minutes of outbound calls on every trunk every month, the business case simply isn’t there to justify any unlimited SIP trunking service. And, it gets worse.

    Most of these providers won’t let you spoof your CallerID number on the outbound calls so you are forced to use their trunks for all of your outgoing and incoming calls. If your business depends upon a readily identifiable phone number to transact business over the phone, that means you don’t have the option of using a trunk such as Vitelity’s for incoming calls while reserving SIPStation trunks for outgoing-only calls because the phone number of your business won’t match up. In case you didn’t know, inbound calls are less costly to providers than most outbound calls, hence the reason they prefer to bundle the two in all-you-can-eat plans.

    Let’s do the math for a typical business with support for 4 simultaneous calls. The cost from SIPStation would be $24.99/mo. x 4 channels plus $1/mo. for a single DID. That works out to $100.96 per month. Comparable service from Vitelity would run $3.99/mo. for the unlimited incoming calls with four simultaneous channels leaving a balance of $96.97 for pay-by-the-minute outbound calls. With Vitelity, that works out to 7,211 outbound calling minutes to break even. Anything less than 7,211 minutes of outbound calls a month saves you $14 a month per thousand minutes compared to SIPStation pricing for four ‘unlimited’ trunks. For a business that makes less than an hour of outbound calls a day, the savings would be over $70 a month!

    The math only tells half the story. There are at least a couple other major issues. With SIPStation, if 75% of your calls are incoming and your call volume is substantial, it means that much of the time you’ll only have one trunk available for outgoing calls. That limitation wouldn’t apply with Vitelity since incoming and outgoing calls are managed separately. In effect, you’d be getting the flexibility to make 4 outbound calls at a time using any providers you choose. Not only could you spoof your outbound calls with the CallerID of your incoming DID, but you also could still have 4 available channels for simultaneous incoming calls. Thus, you’ve effectively doubled the call capacity provided by SIPStation for the same money. These numbers obviously reflect substantial savings even for a small business. When you scale up to hundreds of trunks, the effect on your telecom budget will be downright staggering.

    Finally, there’s the SIPstation design and forced integration into FreePBX®. As we’ve mentioned previously, it’s the only non-essential component in FreePBX that cannot be easily removed from within the FreePBX GUI. While you’re not forced to sign up, it does mark a new low by introducing NagWare into an open source product. Yesterday, that lock-in bit everyone in the butt. Because of one or more bugs in some FreePBX updates that were pushed out, entire systems were blown out of the water when attempting a generic FreePBX update of modules from within the GUI using Module Admin. One of the affected modules reportedly was SIPStation which could not be removed. The dilemma was that FreePBX functionality could not be restored without first removing the SIPStation module. For the benefit of those still struggling, here’s how to permanently remove it from your server “the old-fashioned way.” Log into your server as root and issue the following commands:

    amportal a ma uninstall sipstation
    rm -rf /var/www/html/admin/modules/sipstation
    

    Here’s Our Recommendation. Start with a service such as Vitelity and take advantage of the discount coupon below. Then monitor your incoming and outgoing call volume in your business for several months. Next, do the math and see if you don’t save hundreds, if not thousands, of dollars a year by using a provider such as Vitelity rather than an ‘unlimited’ SIP trunking service. Let us know your type of business and post the results of your testing for everyone else to see. Enjoy!

    Originally published: Wednesday, April 29, 2015



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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    1. To be fair, the trunks cost $24.99 per month. []

    Firewalls and Internet Security: Separating FUD and Fiction in the VoIP World

    Some of us have spent years developing secure VoIP solutions for Asterisk® that protect your phone bill while bringing Cloud-based solutions within reach of virtually anyone. So it’s particularly disappointing when a hardware manufacturer spreads fear, uncertainty, and doubt in order to peddle their hardware. In this case, it happens to be Session Border Controllers (SBCs). We want you to watch this latest “infomercial” for yourself:



    To hear Sangoma tell it, every VoIP server protected by merely a firewall is vulnerable to endless SIP attacks unless, of course, you purchase an SBC. And since implementation of Cloud-based servers traditionally limits the ability to deploy an SBC, most Cloud-based VoIP solutions would become vulnerable to SIP attacks. In the words of Sangoma:

    And with telecom fraud and PBX hacking on the rise, it’s important to keep your network secure. For most enterprises, it’s not a matter of if-but-when their [sic] network experiences an attack, potentially costing you valuable time and money.

    Now Sangoma is touting an article in a blog from the U.K. that begins with the headline “Why Firewalls are not Enough.” The purported author is Jack Eagle, who is otherwise unidentified. Not surprisingly, the owner of the blog happens to be a reseller of Sangoma hardware. Here’s what Jack Eagle suggests:

    In addition, the inherent function of firewalls is to deny all unsolicited traffic. Whereby, the act of making a phone call is an unsolicited event, thus, firewalls can be counterproductive to an effective VoIP deployment by denying VoIP traffic.

    For the benefit of those of you considering a VoIP deployment either locally or in the Cloud using Asterisk, let’s cut to the chase and directly address some of the FUD that’s been thrown out there.

    FUD #1: Internet SIP Access Exposes Asterisk to Attack

    False. What is true is that unrestricted SIP access to your server from the Internet without a properly secured firewall may expose Asterisk to attack. Perhaps it’s mere coincidence but the only major Asterisk aggregation that still installs Asterisk with an unsecured firewall and no accompanying script, tutorial, or even recommendation to properly lock it down and protect against SIP attacks happens to be from the same company that now wants you to buy a session border controller.

    FUD #2: Firewalls Aren’t Designed to Protect Asterisk from SIP Attacks

    False. What is true is that the base firewall installation provided in the FreePBX® Distro does not protect against any attacks. In a Cloud-based environment or with local deployments directly exposed to the Internet, that could very well spell disaster. And it has on a number of occasions. The Linux IPtables firewall is perfectly capable of insulating your Asterisk server from SIP attacks when properly configured. With PBX in a Flash and its open source Travelin’ Man 3 script, anonymous SIP access is completely eliminated. The same is true using the tools provided in the latest Elastix servers. And, Incredible PBX servers have always included a secured firewall with simple tools to manage it. Of course, with local VoIP hardware and a hardware-based firewall, any Asterisk server can be totally insulated from SIP attacks whether IPtables is deployed or not. Just don’t open any ports in your firewall and register your trunks with your SIP providers. Simple as that.

    FUD #3: SIP Provider Access to Asterisk Compromises Your Firewall

    False. Registering a server with SIP or IAX trunk providers is all that is required to provide secure VoIP communications. Calls can flow in and out of your Asterisk PBX without compromising your server or communications in any way. Contrary to what is depicted in the infomercial, there is no need to poke a hole in your firewall to expose SIP traffic. In fact, we know of only one SIP provider that requires firewall changes in order to use their services. Simple answer: use a different provider. Consider how you access Internet sites with a browser from behind a firewall. The connection from your browser to web sites on the Internet can be totally secure without any port exposure in your firewall configuration. Registering a SIP trunk with a SIP provider accomplishes much the same thing. All modern firewalls and routers will automatically handle the opening and closing of ports to accommodate the SIP or IAX communications traffic.

    FUD #4: Remote Users Can’t Access Asterisk Without SIP Exposure

    False. Over the past several years, we have written about a number of methodologies which allow remote users to securely access an Asterisk server. That’s what Virtual Private Networks and Port Knocking and Remote Firewall Management are all about. All of these solutions provide access without exposing your server to any SIP vulnerabilities! We hope the authors of this infomercial will give these open source tools a careful look before tarnishing the VoIP brand by suggesting vulnerabilities which any prudent VoIP deployment can easily avoid without additional cost. Just use the right products!

    Originally published: Thursday, April 23, 2015



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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…