Posts tagged: sip

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! Note to Naysayers…

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.

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. If that’s too rich for your blood, you can roll the dice and try what we did (twice) by purchasing a brand new Sprint or Boost Mobile Prepaid LG phone in an unopened box starting at $35. NO GUARANTEES THIS WILL WORK FOR YOU! We’ve had good luck with LG phones at both ends of the price spectrum, and it MAY work with other brands; however, every manufacturer sets up their phones a little differently, and we happen to know that this worked on both a low end and high end LG prepaid phone.

Visit your favorite Sprint/Boost Mobile reseller including Radio Shack (with a Sprint store), Best Buy, and WalMart.3 If store employees will let you, find the Sprint or Boost Mobile Prepaid LG 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! If there is a problem, you still can return it for a full refund at this point.

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 the 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

2/9/2016 Update: You may wish to put off signing up until 3:00 PM Eastern time today (Tuesday, Feb. 9) so you can take advantage of the new Giuseppe Farina monthly plan with 1,200 free minutes, 1,200 free text messages, and 1.2GB of free data per month with a $20 deposit. Also provides tethering for an additional $0.49 per day.

2/11/2016 Update: Three more blockbuster free plans will be introduced for Valentine’s weekend starting at 3:00 PM Eastern time Friday, Feb. 12. You can read all about them here and decide whether it’s worth holding off for one more day. Keep in mind that RingPlus had a major voice outage earlier this week, and their SIP offering now has been dead in the water for over a week. But, if you’re a gambler, this may still be worth the very real risk.

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 at least once every 60 days to keep your account. The simple way to do this is to set up a monthly reminder using your RingPlus trunk. Schedule the reminder to call you once every month using Telephone Reminders in Incredible PBX. 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.


Some Recent Nerd Vittles Articles of Interest…

  1. Be advised that future availability of these “free” plans may go away after February 15 unless you join the Member+ program, the cost of which has not yet been announced. 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! []
  3. Problems have been reported with WalMart varieties. It may be that they have prelocked the phones to Sprint since they get a commission on the revenue. []

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.


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.


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.


Some Recent Nerd Vittles Articles of Interest…

The Gotcha-Free PBX: Simon Telephonics New SIP Gateway for Google Voice

We promised you that free Google Voice calling in the U.S. and Canada would soon be available on every Asterisk® platform whether the platform supported Asterisk Motif or not. And this week we’re covering the second SIP gateway offering for Google Voice. We introduced Bill Simon’s first Google Voice gateway back in June of 2012. This time around the latest iteration features secure OAUTH authentication so there’s no need to divulge your Google Voice credentials. Once you’ve set up your account on the Simonics Google Voice Gateway site,1 you simply create a standard SIP trunk on your Asterisk server or SIP device of choice, and PRESTO! You get secure authentication to Google Voice without worrying whether Google will drop support for insecure authentication methods such as Asterisk Motif down the road. And you can set all of it up for a one-time setup fee. For Nerd Vittles readers, you get $1 off the current $5.99 fee by using this link. Unlike last week’s GVsip offering, the new Simonics service includes free CallerID name lookups plus the ability to connect multiple devices at multiple sites and communicate between the devices using some clever SIP magic. You also can map incoming calls to any SIP URI rather than just the destination from which you register a Google Voice account. This new gateway is a real winner!

Why do this? There are several reasons aside from the free calls and free phone number. First, Google has warned for years that insecure authentication to Google Voice is going away. It hasn’t yet which is the reason Asterisk Motif logins still work. When Google finally pulls the plug (and they will), your Google Voice days are over using the Asterisk platform. Second, some of the Asterisk aggregations such as Elastix® never supported Google Motif. Hence, free Google Voice calling wasn’t available at all to those using the Elastix platform. That limitation is now a thing of the past. You can create a simple SIP trunk and begin enjoying free Google Voice calling in the U.S. and Canada just like some of the rest of us have been doing for years. Third, Google Voice support was the sole reason that many have stuck with the FreePBX® GUI despite the gotchas. Now you have a choice. Any Incredible PBX™ or Asterisk-GUI™ server now supports Google Voice without your having to worry about constant changes to the Asterisk Motif driver to support refinements at the Google Voice end. Now it’s a pure SIP trunk using pure SIP technology as far as Asterisk is concerned. The only limitation is the one imposed by Google. You need to reside in the United States to use Google Voice even though free calling is available to the U.S. and Canada.

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

1. Using your favorite browser, log in to the Google Voice account you wish to associate with the Simonics SIP gateway. Be sure that you’ve enabled Google Chat in your Google Voice setup.

2. Using a separate tab of your browser, connect to the Simonics Google Voice Gateway site.

3. Go through the steps to register your Google Voice account with the Simonics Google Voice gateway and obtain your credentials.

4a. For those using FreePBX or Elastix, use another tab of your browser to open the GUI interface and create a new SIP trunk using your new SIP login credentials. Replace 8005551212 with your actual Google Voice number and YOUR-SIP-PW with your actual Simonics SIP password in BOTH the PEER Details and Registration String. Add your Google Voice number to the end of the Registration String like this: GV18005551212:YOUR-SIP-PW@gvgw.simonics.com/8005551212

4b. For those using Incredible PBX for Asterisk-GUI, simply download and run our One-Click Installer. You’ll need your Simonics SIP account name and password plus a two-digit dialing prefix to use for outbound calls. It’s that simple!

cd /root
wget http://incrediblepbx.com/simonics-addon.tar.gz
tar zxvf simonics-addon.tar.gz
rm -f simonics-addon.tar.gz
./simonics-addon.sh

Once you’ve finished running the script, your trunk will be up and running. There’s no requirement for steps #5 and #6 with Asterisk-GUI. If desired, jump to Step #7 to set up a SIP URI for your incoming calls.

5. Create an Inbound Route for your incoming calls using the 10-digit number you entered at the end of the Registration String in step #4a.

6. Create an Outbound Route for outgoing calls that should be handled by your Google Voice trunk. The CallerID number will be your Google Voice number. You cannot change it.

7. If you’d prefer to send incoming calls to a designated SIP URI instead of the server that registered with the Simonics gateway, enter the address in the format: pbx@myserver.xyz. For additional details, read our previous article on SIP URIs.

8. Repeat this setup procedure for as many Google Voice accounts as you wish to activate using the steps above. If you’re using Incredible PBX for Asterisk-GUI, remember to edit the script and change the TRUNK=simonics entry to something like TRUNK=simonics2. Also use a unique two-digit dialing prefix for each trunk. Be sure to logout of your previous Google account before repeating the drill. Enjoy!


Don’t forget to List Yourself in Directory Assistance with your new IPkall PSTN number so everyone can find you by dialing 411. And be sure to add your new number to the Do Not Call Registry to block telemarketing calls.

Originally published: Monday, April 13, 2015


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



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


 
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.


Some Recent Nerd Vittles Articles of Interest…

  1. In addition to substantial technical assistance, Simon Telephonics is also a financial contributor to the Nerd Vittles project. []

The Gotcha-Free PBX: GVsip Gateway Service for Google Voice

We promised you that free Google Voice calling in the U.S. and Canada would soon be available on every Asterisk® platform whether the platform supported Asterisk Motif or not. And today the first of two SIP gateway offerings has arrived. With this new service, you simply create a standard SIP trunk on your Asterisk server of choice, associate your Google Voice account with the GVsip gateway service, and PRESTO! You get secure OAUTH authentication to Google Voice without worrying whether Google will drop support for insecure authentication methods such as Asterisk Motif down the road. And you can set all of it up without compromising your credentials for a one-time setup fee of $20 (yes, the price quadrupled!).

UPDATE: Due to frequent and lengthy outages, we no longer can recommend GVsip as a dependable choice for VoIP service. A second SIP Gateway for Google Voice is now available from Simon Telephonics. Our review is available here.

Why do this? There are several reasons aside from the free calls and free phone number. First, Google has warned for years that insecure authentication to Google Voice is going away. It hasn’t yet which is the reason Asterisk Motif logins still work. When Google finally pulls the plug (and they will), your Google Voice days are over using the Asterisk platform. Second, some of the Asterisk aggregations such as Elastix® never supported Google Motif. Hence, free Google Voice calling wasn’t available at all to those using the Elastix platform. That limitation is now a thing of the past. You can create a simple SIP trunk to the GVsip gateway and begin enjoying free Google Voice calling in the U.S. and Canada just like some of the rest of us have been doing for years. Third, Google Voice support was the sole reason that many have stuck with the FreePBX® GUI despite the gotchas. Now you have a choice. Any Incredible PBX™ or Asterisk-GUI™ server now supports Google Voice without your having to worry about constant changes to the Asterisk Motif driver to support refinements at the Google Voice end. Now it’s a pure SIP trunk using pure SIP technology as far as Asterisk is concerned.

Do us the favor of using our signup link for the new GVsip gateway service so that Nerd Vittles gets a piece of the action to keep the lights on. If you’re one that never trusts too-good-to-be-true offers, then take advantage of the free trial without ever pulling out your credit card. So here’s how to get started.

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

1. Using your favorite browser, log in to the Google Voice account you wish to associate with the GVsip gateway. Be sure that you’ve enabled Google Chat in your Google Voice setup.

2. Using a separate tab of your browser, click on the Get Access / Login with Google button on the GVsip site.

3. Go through the steps to associate your Google Voice account with the GVsip gateway and obtain credentials.

4a. For those using FreePBX or Elastix, use another tab of your browser to open the GUI interface and create a new SIP trunk using your actual GVsip credentials (replace ACCTNO and ACCTPW) and Google Voice number (replace 8005551212):

4b. For those using Incredible PBX for Asterisk-GUI, simply download and run our One-Click Installer. You’ll need your ACCTNO and ACCTPW from GVsip plus a two-digit dialing prefix to use for outbound calls. It’s that simple!

cd /root
wget http://incrediblepbx.com/gvsip-addon.tar.gz
tar zxvf gvsip-addon.tar.gz
rm -f gvsip-addon.tar.gz
./gvsip-addon.sh

Once your trunk is up and running, skip sections 5 and 6 below and jump to Step #7 to complete the install.

5. Create an Inbound Route for your incoming calls using the 10-digit number you entered in the previous step.

6. Create an Outbound Route for outgoing calls that should be handled by your Google Voice trunk. The CallerID number will be your Google Voice number.

7. Finally, go back to the GVsip site and login again if your original login expired. Then associate your registered GVsip trunk with your Google Voice account after accepting the Terms of Service agreement.

8. Repeat this setup procedure for as many Google Voice accounts as you wish to activate using the steps above. If you’re using Incredible PBX for Asterisk-GUI, remember to edit the script and change the TRUNK=GVsip entry to something like TRUNK=GVsip2. Also use a unique two-digit dialing prefix for each trunk. Be sure to logout of your previous Google account before repeating the drill. Enjoy!


Don’t forget to List Yourself in Directory Assistance with your new IPkall PSTN number so everyone can find you by dialing 411. And be sure to add your new number to the Do Not Call Registry to block telemarketing calls.

Originally published: Friday, April 3, 2015


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


NEWS FLASH: The Grandstream HT701 Handytone 701 ATA Analog Telephone Adapter with Lifetime Subscription to GVsip has just been released. For those with standard POTS phones, this ATA at $29.99 is a terrific Google Voice solution. Using our Amazon referral link helps keep the Nerd Vittles lights burning brightly.



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.


Some Recent Nerd Vittles Articles of Interest…