Finally a 100% Portable PBX: Introducing GoIP, a SIP-GSM Gateway for Asterisk
How far we have come! The original Asterisk® claim to fame was its ability to interface with proprietary phone systems and legacy telephony hardware, the glue that literally kept companies stuck to their overpriced PBXs. And, just as wired phone systems began to lose their edge, along came the Bell Sisters to introduce cellular communications with billing that began when the phone started ringing and an end to toll-free calling and extra fees for text messaging on top of exorbitantly priced data service. The piece that traditionally has been missing from Asterisk deployments has been interconnectivity with cellular data services. Well, that was then, and this is now. Meet the GoIP GSM Gateway in one, four, eight, and 16-channel flavors to meet your every need. Our focus today will be the one-channel GoIP device, but the larger units work almost identically so, once you’ve mastered the device, it’s not rocket science to move to the 4-channel or 8-channel device (or even larger) if the extra GSM ports better meet your office’s requirements.1
Let’s begin with the basics. What does it do? What does it cost? Why do I need it? How steep is the learning curve?
What Does It Do? In a nutshell, GoIP is a SIP-talkin’ GSM gateway that sits on the same network as your Asterisk server. Once you configure a trunk and a few special Asterisk settings to support SMS messaging, you’ll have another full-featured provider for your PBX, only this one happens to be GSM cellular-based. The good news is GoIP brings to your PBX most of the same feature set that is available using your favorite GSM cellphone except now every extension on your PBX in a Flash™ server can share the cellular connection both for calls and messaging. That means inbound and outbound cell calls as well as inbound and outbound SMS messaging for every extension on your PBX.
With today’s Nerd Vittles additions, here’s the new feature set using a GoIP device from any extension on your PBX:
- Make outbound calls through the GoIP cellular trunk from any PBX extension
- Receive incoming cellular calls and redirect them to any number on your PBX
- Dictate text by phone and deliver SMS messages to any SMS-capable device
- Use a browser to create and deliver outbound SMS messages to any SMS device
- Receive incoming SMS messages and forward the messages to any email address
- Receive incoming SMS messages and forward the messages to any SMS number
- Send an SMS message with a password and receive a callback with DISA dialtone
What Does It Cost? As much as we love Amazon for its referral revenue support of our blog and open source projects, we couldn’t find a single-channel GoIP offering at a reasonable price. The Amazon links provided above for the larger units are competitive (about $100 per port). For the single-channel model, eBay® is your friend. You’ll find multiple providers in the $150 price range. All of the units we’ve found ship from China. We used this provider who got the GoIP device to us exactly 14 days after we ordered it. Ours shipped with the latest firmware, but firmware updates are available here. AliExpress also sells the devices for about the same price. We’ve had good luck with them in the past.
The other expense with the GoIP devices is cellular service. For each channel, you’ll need a GSM SIM card just like what your GSM, AT&T, or T-Mobile cell phone uses. The good news is there are lots of other choices now. See WalMart for some options. Another option for low frequency use would be T-Mobile’s pay-by-the day plans. The $1 (unlimited SMS messaging) or $2/day (unlimited calls and unlimited SMS messaging) plans are almost perfect since you don’t need data. Just be sure to choose a GSM carrier, AT&T or T-Mobile in the U.S. market. Both are supported by StraightTalk. Our favorite remains the (almost) unlimited calling, text, and data $45 plan from StraightTalk. With their AT&T-compatible SIM (don’t buy it in a StraightTalk-locked phone!), it’s a simple matter of moving the SIM card from your cellphone to the GoIP’s GSM slot (connectors facing down). The GoIP unit can spoof an IMEI for picky providers.
Why Do I Need It? The two major advantages of adding a cellular trunk to your PBX are redundancy and portability. Except in the Hurricane Katrina situation, chances are that your Internet service provider and your cellular provider won’t both be dead in the water2 at the same time. The good news is that even with a hurricane, you can pack up your PBX in a Flash server or Raspberry Pi together with your GoIP device and move to higher ground. As fast as you can say “George Bush is a compassionate conservative,” you’ll be back in business.
And then there are the mobile users such as construction site workers, mobile firefighters deployed to a site far from home and other first responders, or even the nomads that manage conventions in a different town every week. Think AstriCon! Rather than relying on crappy hotel WiFi service or paying an arm and a leg for installation of cable or DSL Internet service which often isn’t available anyway, now you have the flexibility to deploy a full-featured PBX at almost any temporary site with nothing more than a $30 Wi-Fi firewall/router, a PBX in a Flash Server or Raspberry Pi, and a GSM SIP trunk courtesy of GoIP. The only other ingredient you need is a little electricity. That could be a wall outlet, or a generator, or an inexpensive AC inverter for your vehicle. Did we mention it’ll work identically on the next site without spending an extra nickel. Hardware cost for the Mobile Communications Center (as shown below): about $250.
Last but not least are all of the organizations that could benefit from an SMS-based emergency messaging service. A dollar a day is a small price to pay to deploy a service that can alert the public, employees, or parents and students of emergency situations. Before you read about the next mass shooting or midnight tornado, give it some thought. We’ve already introduced SMS Blaster to make the job easy. Or you can roll your own by building a simple text file in /tmp/callees.txt with a 10-digit3 callee’s phone number on each line. Then add the following snippet to your Asterisk dialplan code and put your emergency message in line 2. You’ve just replaced a $100 a month message blasting service with a totally portable, self-managed solution. And you’ll recover your hardware costs in less than three months.
[goip-sms-blaster]
exten => s,1,Answer
exten => s,n,Set(SMSMSG="Here is where your emergency message goes.")
exten => s,n,ReadFile(callees=/tmp/callees.txt)
exten => s,n,Set(callees=${URIENCODE(${callees})})
exten => s,n,Set(callees=${REPLACE(callees,%0A,-)})
exten => s,n,Set(SMSNUM=${callees:0:10})
exten => s,n,While($[${LEN(${SMSNUM})}>9])
exten => s,n,NoOp(Here's where we send SMS message to: ${SMSNUM})
exten => s,n,Set(SMSOUT=${SMSNUM}%0A${SMSMSG})
exten => s,n,Set(SMSOUTRAW=${URIDECODE(${SMSOUT})})
exten => s,n,Set(MESSAGE(body)=${SMSOUTRAW})
exten => s,n,MessageSend(sip:goip_1)
exten => s,n,Set(callees=${callees:13})
exten => s,n,Set(SMSNUM=${callees:0:10})
exten => s,n,Set(SMSNUM=${REPLACE(SMSNUM,-,0)})
exten => s,n,EndWhile()
exten => s,n,Hangup()
How Steep Is the Learning Curve? Lucky for you, you’re not going to have to worry about the learning curve. After all, that’s why you come to Nerd Vittles, isn’t it? We’ve spent the better part of a week getting the GoIP to sit up and bark. If you’re a slow typist, it might take you 10 minutes to get everything set up and functional once you have your GoIP device and SIM card in hand. When we’re finished, you’ll have an easy way to make and receive calls through your GoIP device using any extension on your PBX. And you’ll have a simple utility to send and receive SMS messages. In fact, you’ll be able to dictate your SMS messages from any phone connected to your PBX and send them out to any number supported by SMS including the millions of Google Voice numbers. Last but not least, we’ll provide a utility to send password-protected SMS messages to GoIP and receive a return call with DISA dial tone to make outbound calls using any available trunks on your PBX.
A Word About Security. We’re a little paranoid when it comes to security so bear with us. Without impugning anyone’s integrity, suffice it to say this device is manufactured in China. Although the device reportedly runs Linux, none of its other firmware is open source, at least not that we could find. There also are three back doors into the system which can be triggered by SMS commands to the device itself. These are well documented in the GoIP User’s Manual. Whether there are other backdoors or whether the device “phones home” are questions we have neither the time nor the money to explore. Unless you do, you are well advised to treat the device in the same way you would treat a new employee on their first day at work. Don’t put the device on a private LAN in which other computers or devices on the LAN are not protected. Don’t use a SIM card with an automatic renewal feature or with authority to post charges against your credit or debit card. Change your Admin password to the device immediately. Don’t use a password you use elsewhere! Anyone can reset the device to factory defaults by knowing the default credentials and sending RESET admin in an SMS message to the device. We love the device, but be careful.
Initial Setup of the GoIP Device
To begin, you’ll need cellphone coverage in the place where you intend to connect your GoIP device. Verify this while the SIM card you plan to use is still installed in a working cellphone. Make a call and send an SMS message to verify that the site is appropriate. Next, verify that you have a place to connect your GoIP device to your LAN in the same location. Both of these are important first steps, or you’ll be wasting your time continuing on. Once the connectivity issues are out of the way, turn off your cell phone, remove the GSM SIM card, and insert it into the GoIP device with the connectors pointing downward. You should hear a click when the SIM card is properly seated. Now connect the device behind a hardware-based firewall/router that provides DHCP service. Plug an Ethernet cable into the LAN port of the GoIP device and connect it to your network. Finally, using the power adapter provided, apply power to the device. Watch the blinking lights. While booting the RUN light will flash on and off every 100 milliseconds. Once the RUN and CHANNEL lights flash GREEN once per second, you’re in business. Now use another cellphone to send a text message with the word INFO to the phone number associated with the SIM card you plugged into the GoIP Device. You should receive a return message telling you the DHCP LAN address associated with the GoIP CHANNEL port where you plugged in the SIM card. Write it down! We’re not going to use the PC port so you can ignore its IP address for now.
Asterisk Prerequisites for Today’s GoIP Project
We’ll be using PIAF-Green with Asterisk 11 and FreePBX 2.11 today so you’ll have to read between the lines if you’re using a prehistoric release or a non-FreePBX system. We’re also assuming you’ve installed Incredible PBX™ 11 which provides the necessary components to get Google’s text-to-speech and speech-to-text features working. If you’d prefer to roll your own, then start by installing Lefteris Zafiris’ GoogleTTS and Speech Recognition components for Asterisk. For PBX in a Flash users that aren’t using Incredible PBX, you can follow this tutorial to install all of the necessary components in one click.
Initial Setup of FreePBX for the GoIP Device
We’ve found that it’s easier to configure the FreePBX® side to support the GoIP, and then configure the GoIP unit. There are seven simple steps. If you don’t want SMS DISA callback support in your setup, skip the last two steps.
- Add GoIP SIP Trunk
- Add Custom SIP Settings
- Add GoIP Outbound Route
- Add GoIP Custom Destination
- Add GoIP Misc Application
- Add GoIP DISA Context
- Add GoIP DISA Misc Application
1. Start by adding a new SIP Trunk to support the GoIP device. Be sure to match the device names we’ve shown exactly, or nothing will work. Our special thanks to samyantoun for his initial work on this. Replace 192.168.0.107 with the IP address of your GoIP. Replace 77 with whatever dialing prefix you want to use to make calls through the GoIP trunk. And add the phone number associated with your GoIP in the Outbound CallerID field. If you’re using the GoIP device behind a hardware-based firewall with no Internet port exposure, then you can leave password as the secret. Otherwise, you would want something very secure!4
2. Add a couple of custom SIP entries at the bottom of Asterisk SIP Settings to support SMS messaging with Asterisk. Set accept_outofcall_messages=yes and outofcall_message_context=sms_message. Then Submit Changes.
3. Add an Outbound Route to make calls using your GoIP device using the dial prefix you chose for the trunk:
4. Next we need to add a FreePBX Custom Destination to support the Nerd Vittles speech-to-text module which we’ll be using to dictate and send SMS messages using any telephone on your PBX. Under Admin -> Custom Destination, add an entry that looks like this:
5. Then we need to associate an extension number with the custom destination we just added. We’ve chosen 4647 which spells GoIP. Choose Applications -> Misc Application and enter the following:
6. DISA is an Asterisk function that lets someone call into your PBX and obtain dial tone to place an outbound call using the available trunks on your PBX. In the case of the GoIP device, this gets a little fancier. We’ll actually be sending an SMS message with a custom password to the GoIP device, and it will in turn call the SMS sender’s number and provide DISA dialtone after the user enters a special DISA PIN. Make the PIN and password very secure. We’ll get to the password in a minute. On the FreePBX side, add a DISA context in FreePBX under Applications -> DISA that looks something like the following with a secure PIN (not the one in the example):
7. In order to use DISA with GoIP, we’ll need an extension associated with the DISA function. We add this number using FreePBX Misc Application. You can use any available extension number you like. Just remember what you chose when we configure the GoIP side to support SMS DISA access. Here’s what we use:
Configuration of the GoIP Device
All of the GoIP device configuration is handled using a browser pointed to the internal IP address of the GoIP. If you haven’t already done so, send an SMS message with the word INFO to the phone number associated with your GoIP device. You will get a return message with the private IP address of the unit. Using a browser, point it to the IP address and login with username admin and password admin. It’s probably a good idea to reset your unit to factory defaults before beginning the setup just to make sure you’re starting with a clean slate. Send an SMS message to the device with the words RESET admin to initialize the hardware.
As we’ve mentioned, sending the admin password to the device with the RESET keyword forces a total reset of the device so you obviously want to change this admin password immediately unless you want to risk a total stranger sending a reset command to your device. Do it now under Tools -> Change Password -> Administration Level. It’s probably a good idea to change the other passwords as well.
Next, click Configurations. This is the screen on which you set everything. The Preference pane has the country-specific settings for both the network and your cellphone carrier so set them carefully. The IMEI will default to the actual IMEI of your unit. If your cellphone carrier requires registration of a specific IMEI before your SIM card will work, then you can spoof the IMEI using the IMEI of the cell phone that was previously used with this SIM card. For the East Coast of the United States, our setup looks like this:
If you’re using DHCP for the GoIP, the Network Configuration pane shouldn’t require any changes. We do recommend that you lock the DHCP address to the GoIP in your router so that it doesn’t inadvertently change down the road. You will note that a PPTP VPN tunnel for the device is supported although we haven’t yet played with it.
The Call Settings pane has all of your SIP settings for the GoIP. These have got to be right or nothing will work. Our setup (that works) is shown below. Start by clicking on each of the Settings and Preferences links to open up the sub-menus. Both 192.168.0.180 entries should be replaced with the IP address of your Asterisk server. The Phone Number and Authentication ID both need to be goip_1 as shown. The password is password unless you changed your secret in the FreePBX trunk setup. DTMF Signaling should be changed to Outband and DTMF Type should be RFC2833. Ours still doesn’t work reliably, but that may be the lousy cellphone signal in our office. We recommend ULAW and ALAW exclusively for the Audio Codecs. You don’t want the overhead of codec translation particularly if you’re using a Raspberry Pi. On a normal server, G.729 would obviously reduce the bandwidth of GoIP voice calls. Get it working first and then experiment! The RTP port range should be 10000-20000 to match your Asterisk default setup.
The Call Divert pane is where we configure all of the Nerd Vittles magic. Forward Number(PSTN To VoIP) should be the number on your PBX to which you want inbound GoIP calls forwarded when someone calls the cellphone number associated with your GoIP device. This could be an extension, ring group, IVR, or even the DISA number we set up above. Just be sure you have a verrrrrry secure DISA PIN if you go this route! It’s your phone bill. The SMS Mode must be changed to Relay, and SMS Forward SIP Number must be s to work with the Nerd Vittles apps.
Once you have all of your settings entered, click the Save Settings link under Configurations. The unit will reload its SIP setup. It usually takes about 30 seconds. We recommend you now test the setup to make sure you can make a call to the GoIP number and have it forwarded to an extension on your Asterisk server. Then use an extension on your PBX to place an outbound call using the GoIP dial prefix you assigned above. If either call fails, check your settings for typos in both the FreePBX and GoIP configurations.
Adding the Nerd Vittles Apps to Support the GoIP Device
Now for the fun stuff. We’ve built a little shell script that sets up all of the Nerd Vittles applications we outlined above. It’s licensed as GPL2 code so you are more than welcome to make any changes or additions which you believe would be useful. We hope you’ll share them with the rest of us. The script puts everything in the proper place on Incredible PBX systems to support SMS messaging with Asterisk. You’ll be prompted for the following information:
- Email address to which to forward incoming SMS messages
- SMS number to which to forward incoming SMS messages
- Very secure password to trigger PBX callbacks
- Extension number to ring on callbacks
1. When incoming SMS messages are received by the GoIP unit, Asterisk will forward them to this email address.
2. When incoming SMS messages are received by the GoIP unit, Asterisk will forward them to this SMS number. You can disable either the forwarding email address or the forwarding SMS number (not both!) by editing the [sms_message] context in extensions_custom.conf and commenting out either of these lines with a semicolon:
exten => s,n,system(echo "SMS Message From ${SMSDID}: ${SMSMSG}"...exten => s,n,MessageSend(sip:goip_1)
3. This password is what must be sent as an SMS message to the GoIP device to trigger a return call from Asterisk. Do NOT include any spaces in the password and make it very secure!
4. This is the extension number that will be used to place the return call from Asterisk. For DISA service, it would be 3172 in today’s setup. It could also be a regular extension on your PBX if you simply want to trigger a return call from your home or office extension when you send this password via SMS to the GoIP device. Note that the home or office extension must answer the call before the return call will be placed to your SMS device or phone.
Installation. To install the components (a one-minute job!), log into your server as root and issue the following commands:
cd /root
rm GoIP-install.sh
wget http://incrediblepbx.com/GoIP-install.sh
chmod +x GoIP-install.sh
./GoIP-install.sh
If you ever need to make changes to your setup, just run the script again and answer the prompts.
Kicking the Tires. To make sure everything is working, try sending an SMS message to the GoIP with your secret password from #3 above. You should get a return call within 30 seconds. Next, from an extension on your PBX, dial 4647. Dictate a brief message and then enter a phone number for delivery of the message via GoIP to some SMS device (not your GoIP unit!). Finally, send a “Hello World” SMS message to your GoIP device. It should be forwarded to both your email address (#1) and SMS number (#2) within a few seconds. Enjoy!
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Originally published: Monday, September 30, 2013
Need help with Asterisk? Visit the PBX in a Flash Forum.
We are pleased to once again be able to offer Nerd Vittles’ readers a 20% discount on registration to attend this year’s 10th Anniversary AstriCon in Atlanta. And, if you hurry, you also can take advantage of the early bird registration discount. Here’s the Nerd Vittles Discount Code: AC13NERD.
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.Some Recent Nerd Vittles Articles of Interest…
- Some 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. [↩]
- With apologies for the tasteless photo and pun. [↩]
- The length of the phone numbers obviously can be adjusted to meet your local requirements. Just replace the 10′s with the length of the phone numbers you wish to use. Then replace 13 with 3 more than the phone number length you chose. [↩]
- We have engineered today’s GoIP solution for users in the U.S. and Canada. It obviously will support international deployment as well by making adjustments to the dial strings and cellphone settings in both the FreePBX and GoIP configurations. [↩]
The Incredible PBX 11 Inventory. For those that have never heard of The Incredible PBX, here’s the current 11.0 feature set in addition to the base install of PBX in a Flash with the CentOS 6.4, Asterisk 11, FreePBX 2.11, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Incredible Fax, NeoRouter and PPTP VPNs, and all sorts of backup solutions are still just one command away and may be installed using the scripts included with Incredible PBX 11 and PBX in a Flash. Type help-pbx and browse /root for dozens of one-click install scripts. 
By default, most Asterisk systems including those relying upon FreePBX® are configured to deny anonymous SIP calls. If your server has a fully-qualified domain name, it means SIP calls to 201@myserver.com will fail. Since SIP URI calls are free from anywhere in the world, that’s a big deal. The million dollar question is why not just enable anonymous SIP calling on your Asterisk server and call it a day. Then anybody can call any extension on your PBX. That’s half of the answer actually: “Then anybody can call any extension on your PBX.” If that were the only exposure by opening up SIP to anonymous callers, many of us could probably live with that. After all, that’s how POTS phones worked for almost 100 years. The difference, of course, is anonymous SIP calls are free and often undetectable regardless of where the calling party happens to actually be. Unlike HTTP requests which preclude users from spoofing the IP address, SIP requests have no such limitation. That means a SIP packet can knock on your door masquerading as a SIP packet initiated from your own server.
Don’t forget to 
It’s been over 7 years since we first wrote about streaming music on hold with Asterisk®. While we’re energized with Back to School Fever, we decided it was about time for a refresher. And, in honor of 
Cloak & Dapper. 
Introducing SIP.US. We’re delighted to introduce a new SIP trunking provider and supporter for the PBX in a Flash project. While Vitelity
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