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JUST RELEASED: Visit the Incredible PBX WikiBest of Both Worlds: Safely Marrying Asterisk to OpenSIPS
Last week we introduced OpenSIPS, the multi-functional, multi-purpose signaling SIP server which can fulfill almost any communications function one can dream up except the unified communications tasks typically performed with a PBX such as Asterisk®. Today we want to marry the two platforms to give you the best of both worlds. For Incredible PBX® users, the primary advantage of adding an OpenSIPS front end is the elimination of the complexities associated with interacting with your PBX from remote sites with ever changing, dynamic IP addresses coupled with NAT firewalls over which you have no control. While there are many approaches to interconnecting the two platforms, we’re not comfortable with the exposure that a simple registration passthrough design introduces for many Asterisk users. Instead we prefer a model that lets everybody contact you and your users without providing the world the access necessary to allow anonymous strangers a platform from which to launch endless attempts to compromise your Asterisk server and individual Asterisk accounts.
Not all users on an Asterisk PBX need anonymous lurkers to have worldwide, public access to their individual phones. For most, DIDs suffice for public access. For users that do need such access, we will begin by creating a SIP account on your OpenSIPS server that is separate and apart from your Asterisk user account or extension. Also keep in mind that anonymous SIP calls require a match on the SIP URI to reach the person or function desired. You can enable and disable these SIP URI-accessible functions on your OpenSIPS server as desired. And you can determine how obscure to make each of the SIP URIs. Security through obscurity works and deters many SIP attacks. Now let’s address what you can and cannot do with this setup.
Using a SIP phone from anywhere in the world, any SIP user CAN:
- Make SIP URI calls to authorized extensions and ring groups on your Asterisk PBX
- Make SIP URI calls to authorized clients registered to a 3CX PBX with a SIP UUID
- With a legitimate password, make DISA-like calls with Asterisk trunks, if enabled
- With an invalid DISA password, converse with Lenny
- With a legitimate password, check and manage authorized Asterisk voicemail accounts
- With a legitimate password, participate in authorized Asterisk conferences
- Access other authorized Asterisk applications available from Asterisk extensions
Using a SIP phone from anywhere in the world, an OpenSIPS-registered User also CAN:
- Make PSTN calls from OpenSIPS-registered SIP phones, if enabled
- Receive calls from Asterisk forwarded to any OpenSIPS-registered SIP phone
- Receive calls from Asterisk forwarded to any 3CX-registered client or SIP phone
Using any SIP phone registered to a SIP proxy, you CANNOT:
- Log into any Asterisk user account without whitelist permission and credentials
- Make 911 calls
Prerequisites: To complete today’s setup, we’re assuming you have (1) an Incredible PBX server running Asterisk 13, (2) an OpenSIPS server built with version 1.2.0 or later of the Incredible PBX for OpenSIPS installer, and (3) either a registered SIP account and SIP URI on your OpenSIPS PBX or a SIP account with a provider such as a free linphone.org account.
Running pbxstatus on your OpenSIPS server will tell you which version you have. If you don’t have pbxstatus or the version is below 1.2.0, please initialize your Debian 8 platform, download the latest release, and reinstall following the our OpenSIPS tutorial here. There were major changes in the OpenSIPS configuration to support Asterisk connectivity which made an in place upgrade too complex. Our apologies.
Before creating user accounts on your OpenSIPS server, give some thought to a numbering scheme that won’t conflict with extension registrations on your Asterisk server. For example, if your Asterisk server uses extensions 701 through 750, then you may wish to consider using 7701 through 7750 on your OpenSIPS server. The one-to-one match keeps things simple without running into conflicts between the Asterisk extension numbers and the OpenSIPS user accounts. We’ll use the 700 (Asterisk) and 7700 (OpenSIPS) extension ranges in our examples which follow. And we’ve reworked the original OpenSIPS tutorial in keeping with this design to simplify Asterisk integration for new readers just joining the party.
We want to express our sincere appreciation to Bill Simon for his patient tutelage in walking us through some of the potential landmines in marrying an OpenSIPS server with Asterisk. Should your organization ever need professional help with a SIP deployment, there is no finer SIP authority than Simon Telephonics.
1. Configuring Asterisk for Inbound OpenSIPS Calls
Assuming you have an Incredible PBX 13 platform, open the GUI as admin using a browser from your desktop. First, let’s create a Trunk for the OpenSIPS server. Choose Connectivity -> Trunks -> Add SIP (chan_sip) Trunk. For Trunk Name, use opensips. Next, click on the SIP Settings tab in the dialog. For Trunk Name, again use opensips. In PEER DETAILS, enter the following and replace xxx.xxx.xxx.xxx twice with the actual IP address of your OpenSIPS server. Then click Submit and Reload Dialplan when prompted.
type=peer host=xxx.xxx.xxx.xxx context=from-opensips insecure=port,invite disallow=all allow=ulaw deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.xxx/255.255.255.255
Next, using SSH or Putty, login to your Asterisk server as root and issue these commands replacing xxx.xxx.xxx.xxx with the IP address of your OpenSIPS server (choose option 0 when prompted for access type):
cd /root ./add-ip opensips xxx.xxx.xxx.xxx wget http://incrediblepbx.com/from-opensips.tar.gz tar zxvf from-opensips.tar.gz rm -f from-opensips.tar.gz nano -w from-opensips.txt
When the editor opens, scroll down to line 16 and enter a very secure PIN (up to 10 digits) for access to the DISA-like service to make outbound calls via SIP URI. It’s your phone bill so make it long (up to 10 digits) and something that is not easily guessed. On line 20, we have configured DISA for numbers up to 11 digits. If your dialplan requires international dialing support, you can adjust 11 to the desired number of digits. Then save the file and copy the dialplan code into extensions_custom.conf and reload your dialplan:
cd /etc/asterisk cat /root/from-opensips.txt >> extensions_custom.conf asterisk -rx "dialplan reload"
IMPORTANT NOTE: Just because you have configured this DISA option on your Asterisk server does not mean it is available via SIP URI. In fact, no SIP URI access to your Asterisk server is enabled at this juncture. You still must set up the SIP URI connections on your OpenSIPS server. Whether to do that and which features to activate are completely up to you.
2. Configuring OpenSIPS for Asterisk Connectivity
Beginning with version 1.2.0 of the Incredible PBX installer for OpenSIPS, the server itself is preconfigured to support Asterisk connectivity using AVPs. Implementation only requires command line execution of an AVP script to enable each feature you wish to activate. A similar script can be used to deactivate any AVP feature previously activated. To install the scripts on your OpenSIPS server, log in as root using SSH or Putty and issue these commands:
cd /root wget http://incrediblepbx.com/asterisk-features-for-opensips.tar.gz tar zxvf asterisk-features-for-opensips.tar.gz rm -f asterisk-features-for-opensips.tar.gz
The function of each of the Asterisk scripts is self-explanatory from the script names:
- asterisk-add-forward
- asterisk-delete-forward
- asterisk-list-forwards
Three pieces of information are required to add a SIP URI forward from OpenSIPS to your Asterisk server using the AVP asterisk-add-forward script:
- UUID of SIP URI (from any SIP phone, dial UUID@opensips.yourdomain.com to connect)
- Asterisk Extension (destination where incoming OpenSIPS call should be forwarded)
- Asterisk Public IP Address
To add a SIP URI for extension 701 on your Asterisk server at xx.xx.xx.xx reachable at 701@opensips.yourdomain.com, the command would look like this where xx.xx.xx.xx is the public IP address of your Asterisk server and opensips.yourdomain.com is the FQDN of your OpenSIPS server: /root/asterisk-add-forward 701 701 xx.xx.xx.xx
CAUTION: Other than for forwards like this, do NOT set up User accounts in the OpenSIPS Control Panel using the same numbers as existing extensions on your Asterisk server. Otherwise, if your SIP phone is registered to a 701 user account on your OpenSIPS server, you lose the ability to connect to any extension on your Asterisk server if a 701 account requiring registration also existed on the Asterisk platform.
To use a name in the SIP URI or enable a second SIP URI for the same Asterisk 701 extension (jdoe@opensips.yourdomain.com): /root/asterisk-add-forward jdoe 701 xx.xx.xx.xx
Simply repeat the steps above for every SIP URI you wish to enable for an Asterisk extension.
To enable DISA-like access via SIP URI using dial as UUID (dial@opensips.yourdomain.com): /root/asterisk-add-forward dial *1 xx.xx.xx.xx
Keep in mind that you need not use "dial" as the UUID. You can make up any name you like. So long *1 is the DISA extension, the UUID can be as obscure as desired e.g. disa5038now.
For voicemail access via SIP URI, you can do it in two ways. For generic access triggering prompts for both the voice mailbox number and the mailbox PIN, use the following: /root/asterisk-add-forward vm *98 xx.xx.xx.xx
For voicemail access to a specific mailbox (701) with only a prompt for the mail PIN, use: /root/asterisk-add-forward vm701 *98701 xx.xx.xx.xx
For access to a specified conference (2663) with a prompt for the conference PIN, use: /root/asterisk-add-forward conf2663 2663 xx.xx.xx.xx
For access to Weather Reports (947) with a prompt for the ZIP Code, use something like this: /root/asterisk-add-forward weather 947 xx.xx.xx.xx
For News Headlines (951), use: /root/asterisk-add-forward news 951 xx.xx.xx.xx
To delete any previously created UUID forward: /root/asterisk-delete-forward
To list existing UUID forwards for SIP URIs: /root/asterisk-list-forwards
Calling Tip: If your softphone is registered to an OpenSIPS User account, you can call any of the enabled forwarding entries by entering the UUID without @opensips.yourdomain.com, e.g. dialing vm would connect to the Asterisk voicemail system with a prompt for mailbox.
3. Enabling Inbound Calls from Asterisk to a SIP Phone
In today’s design, incoming calls to your Asterisk PBX can be forwarded to a user account on your OpenSIPS server or a free linphone.org user account by (1) creating a free User account in the OpenSIPS Control Panel or at linphone.org, (2) logging into that user account with a SIP phone or softphone, (3) creating a custom extension in the Incredible PBX GUI that points to the SIP URI of your user account on the OpenSIPS server or your free linphone.org SIP user account or a 3CX client, and (4) adding that custom extension to either a Ring Group that includes your Asterisk extension or enabling FindMe/FollowMe for your Asterisk extension and designating the custom extension as the No Answer Destination. Need support for multiple Asterisk users? Not a problem. Repeat the drill for each user.
The procedure for adding a User Account in the OpenSIPS Control Panel was covered in last week’s article. The procedure for creating a free Linphone User Account was covered in an earlier article so we won’t repeat it here. Another obvious SIP URI destination is any 3CX Client if you’ve previously set up a free 3CX server following our 3CX tutorial. Refer back to those articles if you need a refresher.
On the Asterisk side, login to the Incredible PBX GUI as admin with your favorite browser. Then choose Applications -> Extensions -> Add Custom Extension. For the User Extension and Display Name, we recommend using the 7701 numbering scheme for remote accounts. Then click on the Advanced tab and enter the SIP URI of your OpenSIPS, Linphone, or 3CX User account as the Dial option, e.g. SIP/yourname@sip.linphone.org or SIP/7701@opensips.yourdomain.com. Click Submit and Apply Config to reload dialplan.
To assure that incoming calls ring on both your Asterisk phone (701) and your registered SIP phone, we recommend setting up a Ring Group on the Asterisk side that includes both the 701 extension and the new 7701 custom extension. Then adjust your Inbound Routes to point to the number of this Ring Group instead of to 701. In this way, you can preserve the voicemail functionality associated with your 701 extension. FYI: None of these servers proxy audio and video of your calls. They provide a SIP registration service only.
The other alternative to a Ring Group is to enable FindMe/FollowMe in the 701 extension settings and then specify Extension:701 as the No Answer Destination. With this approach, voicemail will never be triggered on calls sent to extension 701 on your PBX. Since OpenSIPS lacks voicemail, you would lose calls not answered on your registered SIP phone or softphone.
TIP: We use 3CX clients exclusively for inbound calls on iPhones and Android devices because we have found they are far superior in dealing with both push notifications and NAT routing. 3CX clients actually ring when someone calls AND you can hear both sides of every call.
4. Outbound PSTN Calling from OpenSIPS
The DISA setup documented above allows your existing Trunks to continue to be managed and secured exclusively on your Asterisk server with no trunk exposure on the OpenSIPS platform at all. Thus, if either your public-facing OpenSIPS server or Linphone is ever compromised, nobody will be able to make any calls on your nickel because there will be no trunks available to process the outbound calls. Your DISA password is never exposed.
For some (like us), a two-step outbound calling procedure is just too painful. In that case, with providers such as Skyetel, you can deploy a PSTN calling platform on both your Asterisk server and on OpenSIPS. We documented the Skyetel trunk setup for OpenSIPS in our tutorial last week. The good news is nothing precludes deployment of Skyetel at multiple sites even if you only use Skyetel on the OpenSIPS platform for outbound calling. And this completely avoids implementing a DISA solution which has security implications of its own. Effective 10/1/2023, $25/month minimum spend at Skyetel is required.
Enabling direct PSTN calling with OpenSIPS means nobody can ever make PSTN calls merely by guessing a SIP URI. It requires an actual SIP registration to OpenSIPS, and you have Fail2Ban to assist with securing that process. So the outbound calling design is completely up to you. Direct PSTN calling from OpenSIPS is no less safe so long as none of your OpenSIPS User account passwords are compromised.
5. Enabling Calls from Asterisk to OpenSIPS Users
For OpenSIPS AVP forwards that have been enabled to Asterisk extensions, you probably will also want to provide a way for Asterisk users to return those calls directly to OpenSIPS users since that will be the CallerID that displays when an OpenSIPS user places a call directly to a forwarded Asterisk extension. Assuming a SIP phone has been registered to User account 7709, when that OpenSIPS user places a call to a forwarded Asterisk extension 701, it means the Asterisk user will see 7709 displayed as the CallerID for the incoming call even though the User of the OpenSIPS 7709 extension may also be associated with extension 709 on the Asterisk side. If the Asterisk callee attempts to return the call by dialing 7709 instead of 709, the call would fail. To avoid confusion by Asterisk users, the simple solution is to add an additional Custom SIP extension for every OpenSIPS User account.
For example, on the Asterisk side, login to the Incredible PBX GUI as admin with your favorite browser. Then choose Applications -> Extensions -> Add Custom Extension. For the User Extension, enter 7709. For the Display Name, enter the name of the person using that OpenSIPS user account. Next, click on the Advanced tab and enter the SIP URI for this OpenSIPS User account as the Dial option, e.g. SIP/7709@opensips.yourdomain.com. Click Submit and Apply Config to reload dialplan.
FYI: Matching Custom Extension numbers on the Asterisk platform to identical extensions on your OpenSIPS server does not create the registration problems we cautioned against earlier. Only Asterisk extensions requiring actual SIP registration need to remain unique from accounts on your OpenSIPS platform.
6. A Few Words About Security 
If you’ve been using Incredible PBX with its Travelin’ Man 3 firewall, it’s not unlike living in a gated community where most of the outside world doesn’t even know you exist. Adding a "second home" with OpenSIPS is not unlike buying a summer place next door to Fred Sanford in Watts. You might as well have set up shop in the middle of Russia because, for all intents and purposes, you have. Anybody in the world can guess your IP address and spend the day trying to break into your server. So the name of the game is vigilance. Especially for the first few weeks, you need to run iptables -nL regularly and see how quickly your Fail2Ban blacklist is filling up. If you heeded our advice and set up your OpenSIPS server on a KVM platform (instead of OpenVZ), we’ve got a handy little script that will let you move bad guys snagged by Fail2Ban to the permanent IPset blacklist. Just download the script and run it daily to move the Fail2Ban entries to permanent block status in the IPset blacklist:
cd /root wget http://incrediblepbx.com/move-fail2bans-to-ipset.tar.gz tar zxvf move-fail2bans-to-ipset.tar.gz rm -f move-fail2bans-to-ipset.tar.gz ./move-fail2bans-to-ipset
Once you have verified that the IP addresses actually are being populated in the IPset blacklist table (ipset list | sort), you can add the script to /etc/crontab to run automatically each night:
echo "2 4 * * * root /root/move-fail2bans-to-ipset > /dev/null 2>&1" >> /etc/crontab
If you’d like a head start on your IPset blacklist, simply download our latest list and then reboot your server:
cd /etc wget http://incrediblepbx.com/badguys.tar.gz tar zxvf badguys.tar.gz rm -f badguys.tar.gz
Another potential vulnerability is SSH. This command will tell you who has attempted to login to your server as root: cat /var/log/auth.log | grep password. If you ever see a failed login and it wasn’t a mistake on your part, change your SSH access port immediately if not sooner: nano -w /etc/ssh/sshd_config. Then restart SSH: /etc/init.d/ssh restart. Better yet, set up SSH public key authentication.
The other major consideration is the number of holes you punch into the security of your Asterisk server using the OpenSIPS asterisk-add-forward script. Every time you add an extension to this list, you open another (read-only) window into your Asterisk communications world. And anybody can connect to these extensions using either the FQDN of your OpenSIPS server or its IP address. Even though we don’t practice what we preach, we strongly recommend using alphanumeric UUIDs instead of numbers for these access points. That at least avoids random calls from bad guys that are accustomed to numeric numbers only in SIP URIs. Regularly review your OpenSIPS log for unusual strings of forwarded calls and adjust your forwarding UUIDs accordingly: cat /var/log/opensips.log | grep forwarded.
In our previous article, we’ve already addressed how important it is to limit User accounts to your FQDN and never the IP address of your OpenSIPS server. In this way, you limit OpenSIPS registration exposure to your FQDN and never the IP address of your server. Fail2Ban also assists here by blocking failed login attempts after a single failure unless you have whitelisted the IP address in Fail2ban’s ignoreip list in /etc/fail2ban/jail.conf and restarted Fail2Ban with this command: /etc/init.d/fail2ban restart. These are the only entry points that offer the ability to actually register to your server. AVPs never do. Obviously, a successful SIP registration is much more dangerous than a random phone call on a SIP URI set up using AVP extension forwarding.
Finally, passwords now matter on your Asterisk PBX for any port forward you’ve established with OpenSIPS. For example, if you’ve set up a generic forward to access voicemail, then it means anybody guessing the SIP URI you created can spend the day (at no cost) attempting to break into ANY voicemail account on your Asterisk server by guessing the PIN. Fail2Ban will not protect you here. If you’ve set up DISA-like access to your Asterisk server on OpenSIPS, then the same applies except now the attacker gets a blank check to make commercial calls if they can guess your access PIN. Worried yet? We hope so. Sure beats a $100,000 phone bill.
7. Taking OpenSIPS for a Test Drive 
We usually provide a Demo Line for readers to try out our latest creations. For obvious reasons, we prefer not to disclose our OpenSIPS FQDN to the general public. But we have set up a port forward from a DID that we temporarily configured on our OpenSIPS server. So, if you’d like to sample the voice quality of placing a call to a DID in Atlanta forwarded to an OpenSIPS server in New York forwarded to an Asterisk server in Miami and then back to you, try calling 843-606-0555 for a weather report in your favorite ZIP code. We’re betting you will be dumbfounded by the quality of the call. Enjoy!
Originally published: Monday, May 20, 2019
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The 5-Minute Wonder: OpenSIPS Server Takes the Cake
We covered Kamailio in our Part I article. And we’ve skipped writing about SIP server contestants two, three, and four because they each had a healthy dose of insurmountable problems… at least for us. So today we’re pleased to present Part V in our SIP server series. And, as the headline exclaims, with OpenSIPS we’ve found a platform that finally is worthy of your attention. Our requirements were fairly straightforward. We wanted an open source SIP server to which we could connect users to make and receive free as well as commercial calls worldwide. We also wanted a SIP server with good documentation that was simple to install and to integrate into our existing Asterisk platforms without hiring a consultant. And finally we were searching for a SIP server that could be secured easily without providing free phone service to every bad guy on the planet. OpenSIPS has it all.
OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS Platforms, Call Centers, and many others. Source: opensips.org
We’ve often complained that the problem with many open source projects is that the developers get so focused on making money that they skimp on the documentation to encourage consulting work or participation in expensive conferences. We have found just the opposite with OpenSIPS. In fact, much of today’s implementation is based upon an excellent tutorial by the folks at PowerPBX. Down the road, if you find yourself in need of a consultant, their services would be a good place to start. What we’ve added to the PowerPBX design is security, support for clients behind NAT-based routers, and an integration scheme for Asterisk®, FreePBX®, and Incredible PBX® platforms so that you get the best of both worlds, a public facing SIP server with the UC feature set that most organizations expect. Last but not least, our turnkey GPL installer will get you up and running in about 5 minutes.
Choosing a KVM/OVZ7 Platform for OpenSIPS
Let’s begin by addressing the appropriate platform for an OpenSIPS server. The server needs to have a public IP address that is static, and the server should not be situated behind a NAT-based router. It only complicates things and is beyond the scope of what we plan to address. For those that are frequent visitors, you already know that we’ve been pushing everyone to kiss their local hardware goodbye and join the cloud revolution. When it comes to public-facing VoIP platforms like OpenSIPS, most of us don’t have a choice. You need a static IP address on the open Internet. And, for the sake of security, a KVM or OVZ7 cloud platform is a must since older OpenVZ platforms don’t support the ipset component of IPtables which makes it easy to block hundreds of thousands of IP addresses without a performance hit on your server. While we previously have identified older OpenVZ providers for our Incredible PBX platforms protected by the Travelin’ Man 3 firewall, pure whitelist access simply isn’t an option if you wish to retain the functionality of a VoIP application such as OpenSIPS.
Ten to twenty gigabytes of disk space should be more than ample for OpenSIPS. The amount of RAM in your server depends upon the volume of calls your server will be handling. If it’s a dozen simultaneous calls then 1GB of RAM will suffice. If it’s 100,000 calls, then take a look at this article for tips on sizing your server. For today’s implementation, we’ll be using Debian 8 so any low-cost provider or KVMs at Digital Ocean, Vultr, and OVH should be fine.1
We recently went on the hunt to identify KVM or OVZ7 cloud providers around the world that could offer a KVM VPS with 1GB RAM, 20GB storage, and 1TB of monthly bandwidth for about $25 a year. No small feat! But our friends at LowEndTalk have come through. Read the message thread and find an offer with a site that best meets your requirements. Many of the KVM offers require you to open a ticket to get the special pricing and configuration outlined above. Here’s a short list of our favorites, but remember to only use the KVM or OVZ7 offerings below for OpenSIPS!
| Provider | RAM | Disk | Bandwidth | Performance as of 12/1/19 | Cost |
|---|---|---|---|---|---|
| CrownCloud KVM (LA) | 1GB | 20GB + Snapshot | 1TB/month | 598Mb/DN 281Mb/UP 2CPU Core | $25/year Best Buy! |
| Naranjatech KVM (The Netherlands) | 1GB | 20GB | 1TB/month | Hosting since 2005 VAT: EU res. | 20€/year w/code: SBF2019 |
| BudgetNode KVM (LA) | 1GB | 40GB RAID10 | 1TB/month | Also available in U.K PM @Ishaq on LET before payment | $24/year |
| FreeRangeCloud KVM (Ashburn VA, Winnipeg, Freemont CA) | 1GB | 20GB SSD | 3TB/month | Pick EGG loc'n Open ticket for last 5GB SSD | $30/year w/code: LEBEGG30 |
Choosing OpenSIPS Components to Deploy
We’ve divided up today’s tutorial into bite-sized pieces so that you can pick and choose where to stop implementing and start using. You do not need to have an Asterisk server to make and receive calls with OpenSIPS. However, OpenSIPS lacks voicemail and AutoAttendant/IVR components so, if those are a requirement, then you either need a VoIP service provider that offers them, or deploy a $50 Incredible PBX for the Raspberry Pi to add the missing pieces.
What OpenSIPS offers is a free server platform for worldwide SIP communications so that you, your friends, and business associates can call or connect from anywhere using freely available SIP softphones or any of dozens of SIP telephone instruments. We’ll stick with softphones for today, but hardware-based SIP telephones are equally simple to deploy.
This is not a criticism because it is one of the best tutorials we’ve ever used but, if you want to see how complex a typical OpenSIPS server deployment is, take a look at the PowerPBX tutorial we used as a starting point with OpenSIPS. We’ve compressed most of those procedures into a turnkey installer that only requires you to enter a MySQL root password of passw0rd (with a zero) once you have your Debian 8/64 platform up and running.
Deploying a Debian 8 Server Platform
Start by choosing a cloud provider that offers the 64-bit Debian 8 minimal platform as a deployment option. Most do. As noted, we recommend a KVM or OVZ7 platform, but older OpenVZ platforms perform equally well minus support for ipset which makes it easy to block entire countries overrun with bad guys. Choose offerings with at least 1GB RAM and a 10GB drive to get started. Configure your Debian 8 server with a fully-qualified domain name (FQDN). This is critically important with our security design because we will assign all OpenSIPS users/extensions to this FQDN and reserve your server’s IP address purely for connections from service providers and Asterisk servers. This makes it all but impossible for anyone to hack into your server since most script kiddies launch attacks on IP addresses, not FQDNs. Using an unusual FQDN adds an extra layer of security, but that’s your call. If you lack the ability to assign FQDN aliases to a domain which you own, you can obtain a free FQDN from numerous sources including ChangeIP and point it to the IP address of your OpenSIPS server.
Installing OpenSIPS on a Debian 8 Server
Now the fun begins. Log into your Debian 8 server as root and issue the following commands to prepare for the OpenSIPS install:
cd /root wget http://incrediblepbx.com/opensips.tar.gz tar zxvf opensips.tar.gz rm -f opensips.tar.gz
After untarring opensips.tar.gz above, there’s one extra step for those using KVM or OVZ7 platforms. Do NOT make this change if you’re on an older OpenVZ-based server (not recommended!) that shares its kernel with the host machine. Otherwise, the firewall startup will always fail. For KVM and OVZ7 platforms only, issue the following command: cp -p /root/kvm/* /root
Make sure you have logged into your Debian 8 server as root using SSH or Putty from a desktop PC that you will use to manage OpenSIPS with a browser. The reason is because this IP address automatically will be whitelisted in the OpenSIPS firewall as part of the install process. Otherwise, you will need to manually log into SSH and whitelist the IP address of your desktop PC using /root/add-ip each time you wish to access the OpenSIPS Control Panel since TCP port 80 (HTTP) is not exposed to the public Internet as a security precaution.
To begin the install, issue this command:
/root/install
As the install progresses, you’ll be prompted several times to assign and then to use the MySQL root password. Please use passw0rd (with a zero) as your MySQL password, or the install will fail. This is NOT a security risk unless your Debian 8 root user account is compromised. And, in that case, it won’t matter anyway since the MySQL password could easily be changed. The rest of the install is self-explanatory. There are a couple of steps where you will be prompted for input. Correct responses are indicated before the various prompts. Pay particular attention when you are prompted to change the SSH port from TCP 22 to a port number in the 1000-2020 range as a security precaution. We recommend using the year you were born because it will be easy for you to remember. When the install finishes and you log out of your server, the next SSH login will look like this where XXXX is the SSH port you chose and yyy.yyy.yyy.yyy is the OpenSIPS server address: ssh -p XXXX root@yyy.yyy.yyy.yyy
Although most of the configuration of your OpenSIPS server will be handled using a web browser and the OpenSIPS Control Panel GUI, we’ve included a few scripts in /root to assist with maintenance of your server platform. Here’s a brief summary of the script functions:
- pbxstatus – Status of your OpenSIPS server (image sample above)
- add-ip – Temporarily WhiteList IP address until next iptables-restart
- ban-ip – Permanently Ban an IP address
- unban-ip – Unban a previously banned IP address
- log-purge – Zero out all of the major Linux log files
- opensips-check – Assures OpenSIPS and RTPproxy are running (runs automatically)
- Fail2Ban BlackLists –
iptables -nL | grep -A100000 "opensips (" - IPset BlackList (KVM/OVZ7 platforms only) –
ipset list | sort
We secure your server in several ways: (1) by disguising the SSH port, (2) by locking down almost every port on your server with the IPtables firewall with the exception of the SIP ports, (3) by deploying Fail2Ban to scan your OpenSIPS log for errors and lock out attackers for an extended period of time, and (4) by deploying the IPset blacklist on KVM/OVZ7 platforms. With this design, there is a symbiotic relationship between IPtables, Fail2Ban, and IPset. Therefore, it is critically important that you only restart these services using the iptables-restart command. NEVER issue other IPtables commands to restart or save your firewall settings.
Activating a SIP Server with OpenSIPS Control Panel
We don’t want to overload you on the first day with your new OpenSIPS platform so we’ll walk you through the preliminary setup steps to create your SIP Domain. Then we’ll show you how to set up user accounts (also known as extensions). Finally we’ll walk you through setting up a trunk to make and receive calls from a commercial SIP provider. When we’re finished today, you’ll be able to make and receive calls using SIP URIs or DIDs which you have purchased from a provider. Then next week we’ll focus on integration of OpenSIPS with an Asterisk platform of your choice using Incredible PBX and FreePBX as an example. Once we’re finished, you’ll be able to handle user account registrations exclusively on your OpenSIPS server while leaving your Asterisk platform completely hidden from public exposure.
Logging into the OpenSIPS Control Panel
As deployed, the OpenSIPS Control Panel is accessible via web browser. As noted previously, HTTP Port 80 access is blocked by default unless the IP address of your desktop PC has been whitelisted either as part of the initial install or using the add-ip script in /root. Once your desktop PC’s IP address is whitelisted, point your browser to http://xxx.xxx.xxx.xxx/cp
The default Username is admin, and the default password is opensips. Once you’re logged in, immediately click on the Users icon in the upper-right corner of the dashboard. Then click the Edit Info pencil icon for user Admin and change your password. Click Save when done.
Creating Domains with OpenSIPS Control Panel
In the Left column of the Dashboard, you’ll see two tabs: Users and System. Click on the System tab to expose the available choices. Then choose the Domains option.
Domains are the essential building blocks in OpenSIPS. You can manage one or a hundred domains on a single OpenSIPS server, and each domain can have its own set of Users, Trunks/Gateways, and Dialplan rules. We’re actually going to create two domains, one for the IP Address of your OpenSIPS server and a second one for the FQDN of your OpenSIPS server. For added security, we will create all User accounts under the FQDN Domain. And we’ll reserve the IP Address Domain for DID Trunks/Gateways from registered, commercial SIP providers. This design allows attackers to attempt to register to accounts on your IP Address Domain until the cows come home, and they will never be successful because there are no existing SIP user accounts there. Keep it that way! With our OpenSIPS design, Fail2Ban will block attackers after a single failed registration attempt. And OpenSIPS itself will identify and block all SIP flood attacks using either Fail2Ban or IPset (on KVM and OVZ7 platforms only).
Now that you understand the design, let’s set up your domains. After choosing System -> Domains, enter the IP Address of your OpenSIPS server at the SIP Domain prompt. Then click Add New Domain followed by Reload on Server. Repeat the same steps to enter the fully-qualified domain name (FQDN) of your OpenSIPS server. When finished, you should see:
Creating Users with OpenSIPS Control Panel
We’ve already explained the security implications and reason for creating User accounts with your FQDN Domain only. Click on Users -> User Management -> Add New to get started. You can use Numbers (what we call Extensions in Asterisk) or Names. Our preference is to use Numbers for the User accounts and then to create Alias Names (as desired) for each User account. You can’t dial names from most SIP telephones. This also keeps the design similar to what many are used to coming from the Asterisk environment. A completed dialog would look something like the following. Use the Domain pull-down to choose your FQDN. Obviously, the passwords must be secure and must match. Then the Register button will be enabled to save. The actual Numbers used for Usernames are completely up to you.
Create at least a couple User accounts so that you can set up two SIP phones to call yourself and verify that everything is working. These User accounts become an integral part of the SIP URI to receive calls from any SIP phone in the world:
7701@opensips.yourdomain.com
Before you can actually answer an incoming call to your SIP URI, you’ll need to register the User account using either a softphone or SIP phone. We’ll do that next. But, first, let’s create an Alias to 7701 User so that folks can reach you by calling joe@opensips.yourdomain.com
Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the example below. Make sure that you select your FQDN Domain using the pull-downs for BOTH the Domain and Alias Domain fields. Then click Add to save.
Registering a Softphone to an OpenSIPS User Account
There are literally dozens of free SIP soft phones from which to choose. We covered some of our favorites for every platform in previous articles. For our purposes today, we recommend you choose one of the Linphone softphones which are available for the PC, Mac, Linux, Android, and iOS platforms. We also recommend signing up for a free Linphone.org SIP account which doesn’t cost you anything. For today, we will be configuring the softphone to register to your new OpenSIPS server.
Once you have downloaded and installed the Linphone client, go into the Preferences menu and make the following changes. Some depend upon your calling platform.
- Audio Codecs: PCMU, G722, PCMA
- Video Codecs: VP8, H264
- Call Encryption: None
- DTMF: RFC2833 only
- Send InBand DTMF: OFF
- Send SIP INFO DTMF: OFF
- SIP UDP 5060: Enabled
- SIP TCP 5060: Enabled
- Allow IPv6: Disabled
Then set up a new SIP Proxy account: Username (7701), Password (as defined), Domain: your FQDN not IP address, Transport: UDP, Outbound Proxy: OFF, Stun Server: stun.linphone.org, ICE: ON, AVPF: OFF, Push Notification: ON, Country Code Prefix: 1 (if required by your commercial SIP provider), Register: YES, Account Enabled: YES. HINT: You can call Alias Names via SIP URI, but you can only register to a SIP account using its actual Username.
Avoiding Lockouts with NeoRouter VPN
By design, Fail2Ban is unforgiving when it comes to failed registrations. A single failed registration will get an IP address banned for a full week. The reason is because the new bad guy strategy is to hit your server once to determine whether anybody is home. Then the creep bombards you later with an endless stream of registration attempts. With our design, nobody will be home when they return. The bad news is a single failed registration attempt by you or your users will also trigger a ban. There are several workarounds. The easiest is to set up the NeoRouter client on each of your machines including your OpenSIPS server and use the 10.0.0.x private network for access. These IP addresses never get banned. Our previous tutorial will walk you through setting up a free NeoRouter server and installing the free NeoRouter clients on your machines. The client software already is installed and running on your OpenSIPS server. It only requires that you log in using nrclientcmd and register to your NeoRouter server to obtain a private IP address.
There are other options to unban an IP address which has accidentally been snagged. First, almost all of the cloud providers include a Console option in their web portals. Second, you can log into your server via SSH from any non-blacklisted IP address to remove the banned IP address. Once you’re logged in, simply run this command using the IP address you wish to unban: /root/unban-ip xxx.xxx.xxx.xxx
Choosing Commercial SIP Providers
Recall that you cannot register to a SIP alias on your OpenSIPS server. We’ll take advantage of this restriction in setting up incoming calls from commercial providers’ DIDs. To set up Trunks from commercial providers so that you can not only receive incoming calls but also make outbound calls over their PSTN network connections, you must use providers that support IP address authentication rather than a SIP registration. Many providers support this including our platinum sponsor, Skyetel, as well as providers such as VoIP.ms, Anveo Direct, V1VoIP, and many others. In our OpenSIPS design, you also can use DIDs from providers that support SIP URI forwarding such as CallCentric and LocalPhone; however, you are limited to receiving inbound calls only. VoIP communications really shines here because you don’t have to choose a single provider to meet all of your communications requirements.
Skyetel is by far the easiest provider to set up with OpenSIPS. See our earlier tutorial for a special offer that will get you half-price calling for up to $500. Effective 10/1/2023, $25/month minimum spend required. Once you’re registered on the Skyetel site, add a new EndPoint Group using the IP address of your OpenSIP server and designate UDP 5060 as the access port. Sign up for a DID and map it to the OpenSIPS Endpoint Group. Done. In the OpenSIPS Control Panel, navigate to System -> Dynamic Routing and click Add Gateway. Using the template below, create 5 Proxy gateways for the following Skyetel data centers:
- skyetel-NW 52.41.52.34
- skyetel-SW 52.8.201.128
- skyetel-NE 52.60.138.31
- skyetel-SE 50.17.48.216
- skyetel-EU 35.156.192.164
The latest installer will automatically whitelist the Skyetel IP addresses in /etc/iptables/rules.v4 just below the existing 10.8.0.0/24 rule. This will protect you in the event that one or more of the Skyetel IP addresses gets blacklisted inadvertently. You should also add the IP addresses of any other providers you need and then issue the command: iptables-restart
Next, we need to create what Asterisk users know as an Outbound Route. This tells OpenSIPS to send dialed numbers in 11-digit format to Skyetel for termination. We’ve already created the Dial Plan rule for calling out by dialing 1 plus a 10-digit number. So, while you’re still in the Dynamic Routing section of the OpenSIPS Control Panel, click on the Rules tab at the top of the template. Then click Add Rule. Begin by clicking Add ID button and choosing Group ID 0. In the Prefix field, type 1. Now click the Add GW button 3 times after choosing the Skyetel gateways in the following order from the GW pull-down list: skyetel-nw, skyetel-sw, and skyetel-se. Those are the three currently operational Skyetel gateways. When you’re finished, your template should look like the following. Then click the Add button to save the new rule. Click Reload Server to load the new rule into OpenSIPS. Then repeat this procedure leaving the Prefix field blank so that you can make 10-digit calls as well.
Finally, we need to create what Asterisk users know as an Inbound Route. This tells OpenSIPS where to send incoming calls from our Skyetel DID. OpenSIPS handles inbound routes by defining a User Alias for the Username to which you want to route the incoming DID calls. Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the following template and then click Add.
- Username: 7701 (the extension to which to route the incoming calls)
- Domain: opensips.xyz.com (the FQDN of your OpenSIPS server)
- Alias Username: 18435551212 (the 11-digit Skyetel DID)
- Alias Domain: 11.12.13.14 (the IP address of your OpenSIPS server)
- Alias Type: dbaliases
Introducing the VoIP Blacklist
We’ve always dreamed of an effective VoIP Blacklist, and many have tried. But the crowd-sourced VoIP Blacklist at voipbl.org is the real deal. Everybody can post entries (including the bad guys) and, magically, most of the illegitimate entries get sifted out before the next day’s list is released. We’ve made this easy in two ways. First, the list gets populated every night while you sleep. At last count, there were 84,504 IP addresses. And, second, to contribute to the blacklist, run iptables -nL weekly to see if Fail2Ban has snagged any bad guys. If so, simply run the new /root/blacklist utility which will move them into your local blacklist and also format the entries for easy submission to voip.bl whenever you feel the urge. Simply issue the command cat /root/blcklist.txt to display the entries you just blacklisted. Then cut-and-paste the results and post them to the VoIP Blacklist. The whole process takes less than a minute, and you’ll be contributing to a very valuable VoIP resource while also using it.
Congratulations! You now have a functioning OpenSIPS server that can process incoming calls from SIP URIs as well as DIDs. And you can make SIP URI and 11-digit PSTN calls using your SIP softphone that’s registered to your OpenSIPS server. See you next week. Enjoy!
Continue Reading: Best of Both Worlds: Safely Marrying Asterisk to OpenSIPS
Originally published: Monday, May 13, 2019 Updated: Monday, June 24, 2019
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
- Nerd Vittles receives referral fees from some VoIP service providers to help cover the costs of our blog. We never recommend particular companies solely to generate commissions. We also test all services that we recommend. [↩]
Interconnecting a Mobile PBX to the Asterisk Mothership
The Holy Grail for a mobile VoIP solution is a simple way to connect back to your primary Asterisk® PBX via Wi-Fi from anywhere in the world to make and receive calls as if you never left. Let’s tick off the potential problems. First, many home-based PBXs are sitting behind NAT-based routers. Second, almost all remote Wi-Fi connections are made through a NAT-based router. Third, chances are the remote hosting platform blocks outgoing email from downstream servers such as a mobile PBX. Fourth, deciphering the IP address of your remote connection can be problematic. Fifth, the chances of experiencing one-way audio or no audio on your VoIP calls is high because of NAT-based routers at both ends of your connection.
Last week we introduced OpenVPN as a solution for those with multiple VoIP sites to interconnect. But there’s a much simpler solution for those that travel regularly and want to avoid the complexity of configuring OpenVPN. Here is a quick thumbnail of the setup we recommend as your mobile companion, and you’ll never have a one-way audio problem again. In terms of hardware, you’ll need a Raspberry Pi 3B+ with its native WiFi support and a Windows or Mac notebook computer for traveling. You’ll also need a NeoRouter VPN server to make this process seamless. If you’ve already set up an OpenVPN server platform, it will work equally well. One advantage of NeoRouter is that clients can be added from the client side without having to create a config file on the VPN server. All you need is a username and password. But the choice of VPN platform is totally a matter of preference. The objective using either OpenVPN or NeoRouter is secure communications to your home base. We don’t want to have to reconfigure either your home PBX or your traveling PBX or your notebook PC based upon changes in your public and private IP addresses.
Today we’ll walk you through the easiest way to set up a (free) NeoRouter server on the Internet. It can be used to connect up to 254 devices on an encrypted private LAN. We’re delighted to have finally found a perfect use for the (free) Google Cloud instance.
Using a RaspberryPi 3B+, build an Incredible PBX 13-13.10 platform by following our previous tutorial. We’ll set this up on your home WiFi network so that you only have to throw the Raspberry Pi and its power supply in your suitcase when you travel. As part of the setup, we’ll download NeoRouter and activate private IP addresses for your notebook computer as well as both of your PBXs (using nrclientcmd). Next, we’ll interconnect the two PBXs using SIP trunks and the NeoRouter private LAN IP addresses. We’ll take advantage of a neat little Raspberry Pi trick by storing a wpa_supplicant.conf template on your PC for the remote WiFi setup even though we don’t yet know anything about the remote LAN. Once we know the SSID and password at the remote destination, we’ll use your notebook computer to edit the template and transfer the file to the /boot folder of your RasPi’s microSD card. When the card then is inserted and the RasPi is booted, it will automatically move the template to the proper /etc/wpa_supplicant folder to successfully activate your WiFi connection. We’ll also load links, a fast text-based browser, just in case you encounter a hotel that requires some sort of acknowledgement or password before establishing your WiFi connection to the Internet.
Setting Up a (free) NeoRouter Server in the Cloud
Because NeoRouter uses a star-based VPN architecture, that means the NeoRouter Server must always be available at the same IP address for all of the NeoRouter Clients (aka Nodes) to talk to. If you already have a cloud-based server that has a static IP address and can handle the traffic cop duties of NeoRouter Server, then that’s an ideal place to install NeoRouter Server. Simply download the Free flavor of NeoRouter Server that matches your existing platform and install it. Add an FQDN for your server’s IP address, and you’re all set. A detailed summary of available management options is included in our previous NeoRouter v2 article.
We devoted a couple weeks to Google Cloud instances last month, and it turned out to be a pretty awful platform for hosting Asterisk. But the free offering looks to be a perfect fit as a hosting platform for NeoRouter Server. You also won’t have to worry about Google going out of business anytime soon. So let us walk you through an abbreviated setup process on the Google Cloud platform. If you’re just getting started with Google Cloud, read our previous article to take advantage of Google’s generous $300 offer to get you started and to generally familiarize yourself with the mechanics of setting up an instance in the Google Cloud.
For NeoRouter Server, navigate to https://console.cloud.google.com. Click the 3-bar image 
in the upper left corner of your Dashboard. This exposes the Navigation Menu. In the COMPUTE section of the Dashboard, click Compute Engine -> VM Instances. Then click CREATE PROJECT and name it. Now click CREATE INSTANCE and Name it nrserver. The instance name becomes the hostname for your virtual machine. If you want to remain in the Free Tier, choose f1-micro instance as the Machine Type and choose a U.S. Region (us-central1, us-east1 or us-west1). For the Boot Disk, choose CentOS 6 and expand the disk storage to at least 20GB (30GB is available with the Free Tier). For the Firewall setting, leave HTTP and HTTPS disabled. Check your entries carefully and then click the Create button.
When your virtual machine instance comes on line, jot down the assigned public IP address. We’ll need it in a minute. Now click on the SSH pull-down tab and choose Open in a Browser Window. Now we need to set a root password and adjust the SSH settings so that you can login from your desktop computer using SSH or Putty:
sudo passwd root su root nano -w /etc/ssh/sshd_config
When the editor opens the SSH config file, add the following entries. Then save the file and restart SSH: service sshd restart
PermitRootLogin yes PasswordAuthentication yes
You now should be able to log in to your instance as root from your desktop computer using SSH or Putty. Test it to be sure: ssh root@server-IP-address
Before we leave the Google Cloud Dashboard, let’s make the assigned public IP address permanent so that it doesn’t get changed down the road. Keep in mind that, if you ever delete your instance, you also need to remove the assigned static IP address so you don’t continue to get billed for it. From Home on the Dashboard, scroll down to the NETWORKING section and choose VPS Network -> External IP Addresses. Change the Type of your existing address to Static and Name it staticip. Next, choose Firewall Rules in the VPS Network section and click CREATE FIREWALL RULE. Fill in the template like the following leaving the other fields with their default entries. Then click CREATE.
- Name: neorouter
- Target Tags: neorouter
- Source IP Range: 0.0.0.0/0
- Protocols/Ports: check tcp: 32976
CAUTION: Before this firewall rule will be activated for your instance, it also must be specified in the Network Tags section for your instance. Shut down your instance and add the neorouter tag by editing your instance. Then restart your instance.
Now we’re ready to install NeoRouter Free v2 Server on your instance. Be sure to choose the Free v2 variety. Log back into your server as root using SSH/Putty and issue these commands:
yum -y update yum -y install nano wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/CentOS/nrserver-2.3.1.4360-free-centos-x86_64.rpm rpm -Uvh nrserver-2.3.1.4360-free-centos-x86_64.rpm /etc/rc.d/init.d/nrserver.sh restart nrserver -setdomain <DOMAINNAME> <DOMAINPASSWORD> nrserver -adduser <USERNAME> <PASSWORD> admin nrserver -enableuser <USERNAME> nrserver -showsettings
Finally, add the following command to /etc/rc.local so that NeoRouter Server gets started whenever your instance is rebooted:
echo "/etc/rc.d/init.d/nrserver.sh start" >> /etc/rc.local
Installing Incredible PBX 13-13.10 on a Raspberry Pi
Configuring NeoRouter Client on Your Computers
On Linux-based (non-GUI) platforms, setting up the NeoRouter Client is done by issuing the command: nrclientcmd. You’ll be prompted for your NeoRouter Server FQDN as well as your username and password credentials. Perform this procedure on both your home PBX and the Raspberry Pi.
To add your Windows or Mac notebook to the NeoRouter VPN, download the appropriate client and run the application which will prompt for your NeoRouter Server FQDN as well as your NeoRouter credentials. Once completed, you should see all three machines in your NeoRouter Free Client Dashboard: your PC as well as your home PBX and Raspberry Pi-based Incredible PBX. Make note of the private VPN addresses (10.0.0.X) of both your home PBX and your Raspberry Pi. These VPN addresses never change, and we’ll need them to interconnect your PBXs and to set up a softphone on your notebook computer.
Admininistrative Tools to Manage NeoRouter
Here are a few helpful commands for monitoring and managing your NeoRouter VPN.
To access your NeoRouter Linux client: nrclientcmd
To restart NeoRouter Linux client: /etc/rc.d/init.d/nrservice.sh restart
To restart NeoRouter Linux server: /etc/rc.d/init.d/nrserver.sh restart
To set domain: nrserver -setdomain YOUR-VPN-NAME domainpassword
For a list of client devices: nrserver -showcomputers
For a list of existing user accounts: nrserver -showusers
For the settings of your NeoRouter VPN: nrserver -showsettings
To add a user account: nrserver -adduser username password user
To add admin account: nrserver -adduser username password admin
For a complete list of commands: nrserver –help
Interconnecting Your Raspberry Pi and Home PBX
To keep things simple, our setup examples below assume the following NeoRouter VPN addresses: Home PBX (10.0.0.1) and Raspberry Pi (10.0.0.2). Using a browser, you’ll need to login to the GUI of your Home PBX and Raspberry Pi and add a Trunk to each PBX. Be sure to use the same secret on BOTH trunk setups. We don’t recommend forwarding incoming calls from your Home PBX to your Raspberry Pi because most folks won’t be sitting in their hotel room all day to answer incoming calls. Instead, add the number of your smartphone to a Ring Group on the Home PBX and don’t forget the # symbol at the end of the number. On the Raspberry Pi side, we are assuming that whenever a call is dialed from a registered softphone with the 9 prefix, the call will be sent to the Home PBX for call processing (without the 9). For example, 98005551212 would send 800-555-1212 to the Home PBX for outbound routing and 9701 would send 701 to the Home PBX for routing to the 701 extension. You can obviously adjust your dialplan to meet your own local requirements.
On the Home PBX, the chan_sip trunk entries should look like this:
Trunk Name: raspi-remote PEER DETAILS host=10.0.0.2 type=friend context=from-internal username=home-pbx fromuser=home-pbx secret=some-password canreinvite=no insecure=port,invite qualify=yes nat=yes
On the Raspberry Pi, the chan_sip trunk entries should look like this:
Trunk Name: home-pbx PEER DETAILS host=10.0.0.1 type=friend context=from-internal username=raspi-remote fromuser=raspi-remote secret=some-password canreinvite=no insecure=port,invite qualify=yes nat=yes
On the Raspberry Pi, add an Outbound Route named Out9-home-pbx pointed to home-pbx Trunk with the following Dial Patterns. For each Dial Pattern, prepend=blank and prefix=9:
dial string: 1NXXNXXXXXX dial string: NXXNXXXXXX dial string: *98X. dial string: XXX dial string: XXXX dial string: XXXXX
Tweaking Your Raspberry Pi for WiFi Mobility
wpa_supplicant.conf config file to the /boot directory on the card once you arrive at your destination and know the SSID and password of the local WiFi network. When the Raspberry Pi is subsequently booted, the operating system will move the config file to the /etc/wpa_supplicant directory so that your WiFi network will come on line. Here’s what a typical wpa_supplicant.conf file should look like using your actual credentials. The last network section handles open WiFi network connections (think: McDonald’s) if you want to enable them:
country=US
update_config=1
network={
ssid="your-SSID"
psk="your-SSID-password"
key_mgmt=WPA-PSK
scan_ssid=1
priority=5
}
network={
key_mgmt=NONE
priority=1
}
The other gotcha is that some public WiFi networks require some type of web login procedure before you can actually access the Internet even though an IP address may have been assigned to your Raspberry Pi. To handle this situation, you’ll need a text-based web browser on the Raspberry Pi that can be accessed through your notebook PC using SSH and your Raspberry Pi’s VPN address. Our favorite is links which can be installed on your Raspberry Pi before you pack up.
apt-get install links -y
Once you arrive at your destination, connect both your notebook PC and Raspberry Pi to the same WiFi network, login to the RasPi with SSH at the VPN address assigned to your RasPi, and run links to start the browser. Press <esc> to access the links menu options. If you can’t access your RasPi at the VPN IP address, try its WiFi-assigned local IP address.
Adding a Softphone to Your Notebook PC
We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for an extension on your Raspberry Pi. Then enter the VPN IP address of your server plus your extension’s password. Click OK to save your entries.
If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store.
Adding a Softphone to Your Smartphone
Enjoy your pain-free traveling!
Originally published: Monday, April 22, 2019
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
A New VPN for All Seasons: Introducing OpenVPN for Asterisk
This month marks our twentieth anniversary wrestling with virtual private networks. Here’s a quick walk down memory lane. Our adventure began with the Altiga 3000 series VPN concentrators which we introduced in the federal courts in 1999. It was a near perfect plug-and-play hardware solution for secure communications between remote sites using less than secure Windows PCs. Cisco quickly saw the potential, gobbled up the company, and promptly doubled the price of the rebranded concentrators. About 10 years ago, we introduced Hamachi® VPNs to interconnect Asterisk® and PBX in a Flash servers. At the time, Hamachi was free, but that was short-lived when they were subsequently acquired by LogMeIn®. What followed was a short stint with PPTP VPNs which worked great with Macs, Windows PCs, and many phones but suffered from an endless stream of security vulnerabilities. Finally, in April 2012, we introduced the free NeoRouter® VPN. Version 2 still is an integral component in every Incredible PBX® platform today, and PPTP still is available as well. While easy to set up and integrate into multi-site Asterisk deployments, the Achilles’ Heel of NeoRouter remains its inability to directly interconnect many smartphones and stand-alone SIP phones, some of which support the OpenVPN platform and nothing else.
The main reason we avoided OpenVPN® over the years was its complexity to configure and deploy.1 In addition, it was difficult to use with clients whose IP addresses were frequently changing. Thanks to the terrific work of Nyr, Stanislas Angristan, and more than a dozen contributors, OpenVPN now has been tamed. And the new server-based, star topology design makes it easy to deploy for those with changing or dynamic IP addresses. Today we’ll walk you through building an OpenVPN server as well as the one-minute client setup for almost any Asterisk deployment and most PCs, routers, smartphones, and VPN-compatible soft phones and SIP phones including Yealink, Grandstream, Snom, and many more. And the really great news is that OpenVPN clients can coexist with your current NeoRouter VPN.
Finally, a word about the OpenVPN Client installations below. We’ve tested all of these with current versions of Incredible PBX 13-13, 16-15, and Incredible PBX 2020. They should work equally well with other server platforms which have been properly configured. However, missing dependencies on other platforms are, of course, your responsibility.
Building an OpenVPN Server Platform
There are many ways to create an OpenVPN server platform. The major prerequisites are a supported operating system, a static IP address for your server, and a platform that is extremely reliable and always available. If the server is off line, all client connections will also fail. While we obviously have not tested all the permutations and combinations, we have identified a platform that just works™. It’s the CentOS 7, 64-bit cloud offering from Vultr. If you use our referral link at Vultr, you not only will be supporting Nerd Vittles through referral revenue, but you also will be able to take advantage of their $50 free credit for new customers. For home and small business deployments, we have found the $5/month platform more than adequate, and you can add automatic backups for an additional $1 a month. Cheap insurance!
To get started, create your CentOS 7 Vultr instance and login as root using SSH or Putty. Immediately change your password and update and install the necessary CentOS 7 packages:
passwd yum -y update yum -y install net-tools nano wget tar iptables-services systemctl stop firewalld systemctl disable firewalld systemctl enable iptables
We recommend keeping your OpenVPN server platform as barebones as possible to reduce the vulnerability risk. By default, this installer routes all client traffic through the VPN server which wastes considerable bandwidth. The sed commands below modify this design to only route client VPN traffic through the OpenVPN server.
cd /root curl -O https://raw.githubusercontent.com/Angristan/openvpn-install/master/openvpn-install.sh chmod +x openvpn-install.sh sed -i "s|\\techo 'push \\"redirect-gateway|#\\techo 'push \\"redirect-gateway|" openvpn-install.sh sed -i "s|push \\"redirect-gateway|#push \\"redirect-gateway|" openvpn-install.sh sed -i 's|tls-client|tls-client\\npull-filter ignore "redirect-gateway"|' openvpn-install.sh ./openvpn-install.sh
Here are the recommended entries in running the OpenVPN installer:
- Server IP Address: using FQDN strongly recommended to ease migration issues
- Enabled IPv6 (no): accept default
- Port (1194): accept default
- Protocol (UDP): accept default
- DNS (3): change to 9 (Google)
- Compression (no): accept default
- Custom encrypt(no): accept default
- Generate Server
- Client name: firstclient
- Passwordless (1): accept default
In the following steps, we will use IPtables to block all server access except via SSH or the VPN tunnel. Then we’ll start your OpenVPN server:
cd /etc/sysconfig wget http://incrediblepbx.com/iptables-openvpn.tar.gz tar zxvf iptables-openvpn.tar.gz rm -f iptables-openvpn.tar.gz echo "net.ipv4.ip_forward = 1" >> /etc/sysctl.conf sysctl -p systemctl -f enable openvpn@server.service systemctl start openvpn@server.service systemctl status openvpn@server.service systemctl enable openvpn@server.service systemctl restart iptables
Once OpenVPN is enabled, the server can be reached through the VPN at 10.8.0.1. OpenVPN clients will be assigned by DHCP in the range of 10.8.0.2 through 10.8.0.254. You can list your VPN clients like this: cat /etc/openvpn/ipp.txt. You can list active VPN clients like this: cat /var/log/openvpn/status.log | grep 10.8. And you can add new clients or delete old ones by rerunning /root/openvpn-install.sh.
For better security, change the SSH access port replacing 1234 with desired port number:
PORT=1234 sed -i "s|#Port 22|Port $PORT|" /etc/ssh/sshd_config systemctl restart sshd sed -i "s|dport 22|dport $PORT|" /etc/sysconfig/iptables systemctl restart iptables
04/16 UPDATE: We’ve made changes in the Angristan script to adjust client routing. By default, all packets from every client flowed through the OpenVPN server which wasted considerable bandwidth. Our preference is to route client packets destined for the Internet directly to their destination rather than through the OpenVPN server. The sed commands added to the base install above do this; however, if you’ve already installed and run the original Angristan script, your existing clients will be configured differently. Our recommendation is to remove the existing clients, make the change below, and then recreate the clients again by rerunning the script. In the alternative, you can execute the command below to correct future client creations and then run it again on each existing client platform substituting the name of the /root/.ovpn client file for client-template.txt and then restart each OpenVPN client.
cd /etc/openvpn sed -i 's|tls-client|tls-client\\npull-filter ignore "redirect-gateway"|' client-template.txt
Creating OpenVPN Client Templates
In order to assign different private IP addresses to each of your OpenVPN client machines, you’ll need to create a separate client template for each computer. You do this by running /root/openvpn-install.sh again on the OpenVPN server. Choose option 1 to create a new .ovpn template. Give each client machine template a unique name and do NOT require a password for the template. Unless the client machine is running Windows, edit the new .ovpn template and comment out the setenv line: #setenv. Save the file and copy it to the /root folder of the client machine. Follow the instructions below to set up OpenVPN on the client machine and before starting up OpenVPN replace firstclient.ovpn in the command line with the name of .ovpn you created for the individual machine.
Renewing OpenVPN Server’s Expired Certificate
The server certificate will expire after 1080 days, and clients will no longer be able to connect. Here’s what to do next:
systemctl stop openvpn@server.service cd /etc/openvpn/easy-rsa ./easyrsa gen-crl cp /etc/openvpn/easy-rsa/pki/crl.pem /etc/openvpn/crl.pem systemctl start openvpn@server.service
Installing an OpenVPN Client on CentOS/RHEL
cd /root yum -y install epel-release yum --enablerepo=epel install openvpn -y # copy /root/firstclient.ovpn from server to client /root # and then start up the VPN client openvpn --config /root/firstclient.ovpn --daemon # adjust Incredible PBX 13-13 firewall below iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT cd /usr/local/sbin echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom
Running ifconfig should now show the VPN client in the list of network ports:
tun0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:10.8.0.2 P-t-P:10.8.0.2 Mask:255.255.255.0
UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1500 Metric:1
RX packets:9 errors:0 dropped:0 overruns:0 frame:0
TX packets:39 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:100
RX bytes:855 (855.0 b) TX bytes:17254 (16.8 KiB)
And you should be able to login to the VPN server using its VPN IP address:
# enter actual SSH port replacing 1234 PORT=1234 ssh -p $PORT root@10.8.0.1
Installing an OpenVPN Client on Ubuntu 18.04.2
cd /root apt-get update apt-get install openvpn unzip dpkg-reconfigure tzdata # copy /root/firstclient.ovpn from server to client /root # and then start up the VPN client openvpn --config /root/firstclient.ovpn --daemon # adjust Incredible PBX 13-13 firewall below iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT cd /usr/local/sbin echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom
Running ifconfig should now show the VPN client in the list of network ports:
tun0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:10.8.0.2 P-t-P:10.8.0.2 Mask:255.255.255.0
UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1500 Metric:1
RX packets:9 errors:0 dropped:0 overruns:0 frame:0
TX packets:39 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:100
RX bytes:855 (855.0 b) TX bytes:17254 (16.8 KiB)
And you should be able to login to the VPN server using its VPN IP address:
# enter actual SSH port replacing 1234 PORT=1234 ssh -p $PORT root@10.8.0.1
Installing an OpenVPN Client on Raspbian
Good news and bad news. First the bad news. Today’s OpenVPN server won’t work because of numerous unavailable encryption modules on the Raspberry Pi side. The good news is that NeoRouter is a perfect fit with Raspbian, and our upcoming article will show you how to securely interconnect a Raspberry Pi with any Asterisk server in the world… at no cost.
04/16 Update: We now have OpenVPN working with Incredible PBX for the Raspberry Pi. The trick is that you’ll need to build the latest version of OpenVPN from source before beginning the client install. Here’s how. Login to your Raspberry Pi as root and issue these commands:
apt-get remove openvpn apt-get update apt-get install libssl-dev liblzo2-dev libpam0g-dev build-essential -y cd /usr/src wget https://swupdate.openvpn.org/community/releases/openvpn-2.4.7.tar.gz tar zxvf openvpn-2.4.7.tar.gz cd openvpn-2.4.7 ./configure --prefix=/usr make make install openvpn --version
Now you should be ready to install a client config file, start up OpenVPN, and adjust firewall:
cd /root dpkg-reconfigure tzdata # copy /root/firstclient.ovpn from server to client /root # and then start up the VPN client openvpn --config /root/firstclient.ovpn --daemon # adjust Incredible PBX 13-13 firewall below iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT cd /usr/local/sbin echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom
Installing an OpenVPN Client on a Mac
While there are numerous OpenVPN clients for Mac OS X, none hold a candle to Tunnelblick in terms of ease of installation and use. First, create a new client config on your server and copy it (/root/*.ovpn) to a folder on your Mac where you can find it. Download Tunnelblick and install it. Run Tunnelblick and then open Finder. Click and drag your client config file to the Tunnelblick icon in the top toolbar. Choose Connect when prompted. Done.
Installing an OpenVPN Client for Windows 10
The installation procedure for Windows is similar to the Mac procedure above. Download the OpenVPN Client for Windows. Double-click on the downloaded file to install it. Create a new client config on your server and copy it (/root/*.ovpn) to a folder on your PC where you can find it. Start up the OpenVPN client and click on the OpenVPN client in the activity tray. Choose Import File and select the config file you downloaded from your OpenVPN Server. Right-click on the OpenVPN icon again and choose Connect. Done.
Installing an OpenVPN Client for Android
Our favorite OpenVPN client for Android is called OpenVPN for Android and is available in the Google Play Store. Download and install it as you would any other Android app. Upload a client config file from your OpenVPN server to your Google Drive. Run the app and click + to install a new profile. Navigate to your Google Drive and select the config file you uploaded.
Installing an OpenVPN Client for iOS Devices
The OpenVPN Connect client for iOS is available in the App Store. Download and install it as you would any other iOS app. Before uploading a client config file, open the OpenVPN Connect app and click the 4-bar Settings icon in the upper left corner of the screen. Click Settings and change the VPN Protocol to UDP and IPv6 to IPV4-ONLY Tunnel. Accept remaining defaults.
To upload a client config file, the easiest way is to use Gmail to send yourself an email with the config file as an attachment. Open the message with the Gmail app on your iPhone or iPad and click on the attachment. Then choose the Upload icon in the upper right corner of the dialog. Next, choose Copy to OpenVPN in the list of apps displayed. When the import listing displays in OpenVPN Connect, click Add to import the new profile. Click ADD again when the Profile has been successfully imported. You’ll be prompted for permission to Add VPN Configurations. Click Allow. Enter your iOS passcode when prompted. To connect, tap once on the OpenVPN Profile. To disconnect, tap on the Connected slider. When you reopen the OpenVPN Connect app, the OVPN Profiles menu will display by default. Simply tap once on your profile to connect thereafter.
Installing a Web Interface to Display Available Clients
One advantage of NeoRouter is a simple way for any VPN client to display a listing of all VPN clients that are online at any given time. While that’s not possible with OpenVPN, we can do the next best thing and create a simple web page that can be accessed using a browser but only from a connected OpenVPN client pointing to http://10.8.0.1.
To set this up, log in to your OpenVPN server as root and issue the following commands:
yum --enablerepo=epel install lighttpd -y systemctl start lighttpd.service systemctl enable lighttpd.service chown root:lighttpd /var/log/openvpn/status.log chmod 640 /var/log/openvpn/status.log cd /var/www rm -rf lighttpd wget http://incrediblepbx.com/lighttpd.tar.gz tar zxvf lighttpd.tar.gz ln -s /var/log/openvpn/status.log /var/www/lighttpd/status.log sed -i 's|#server.bind = "localhost"|server.bind = "10.8.0.1"|' /etc/lighttpd/lighttpd.conf systemctl restart lighttpd.service
Latest VPN Security Alerts
https://nakedsecurity.sophos.com/2019/04/16/security-weakness-in-popular-vpn-clients/
Originally published: Monday, April 15, 2019 Updated: Saturday, February 29, 2020
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
- Our discussion today is focused on the free, MIT-licensed version of OpenVPN. For details on their commercial offerings, follow this link. [↩]
Finding Utopia: In Search of the Perfect VoIP Server Platform
Over the past decade, there is no subject that we have devoted more resources to than searching for the best platform on which to run a VoIP server. While our experience primarily has focused on finding the perfect fit for Incredible PBX®, much of what follows applies equally to any other Linux-based VoIP server including Wazo, Issabel, VitalPBX, and 3CX. Today we’d like to share what we’ve learned. Incredible PBX is a complex application. With close to a thousand moving parts, it requires major computing resources to support not only Asterisk® and FreePBX® but also an Apache web server, a MySQL database server, a SendMail server, a HylaFax server, and a Linux firewall with both IPtables and Fail2Ban.
Let’s begin by ticking off the platforms that Incredible PBX currently supports. These include stand-alone dedicated hardware from beefy Dell servers to the Intel NUC and Raspberry Pi. Then there are the virtual machine platforms including VirtualBox, VMware ESXi, and Proxmox. In the Cloud space we’ve covered the stratosphere from the high end with $25/month Google Cloud and Amazon EC2 instances to the dedicated $15/month VoIP platform with RentPBX to the $5/month KVM platforms including Digital Ocean and Vultr to the $2.25/month OVH KVM offering to the $1/month OpenVZ providers including HostedSimply, HostFlyte, Hosting73, HostBRZ, SnowVPS, and AlphaRacks. Have there been some train wrecks along the way? Absolutely. Just search the PIAF Forum for the threads on CloudAtCost, WootHosting, and HiFormance for the war stories and our battle scars. We would be remiss if we didn’t thank the dozens of PIAF Forum volunteers who have endured years of suffering at the hands of some of these providers to make today’s article possible.
So what have we learned? Unless you’re building a VoIP platform as a tinkerer to support just your family, there is zero reason to choose dedicated hardware. And, for home use, with the availability of the $35 Raspberry Pi 3B+, buying a beefier piece of hardware to host your VoIP platform makes no sense. Not only will it be considerably more expensive both to purchase and to operate, but the performance of your VoIP server will be indistinguishable from what you’d see using a Raspberry Pi 3B+. Exhibit A is our $125 RasPi WiFi setup for traveling.
The downsides of dedicated hardware are numerous. In addition to the expense of the platform itself and the monthly cost of electricity, there also are other challenges. First, outages from most Internet service providers are frequent occurrences of unpredictable duration. Second, ISPs typically provide a dynamic IP address which is not a good fit for VoIP platforms that rely upon your IP address to reliably make and receive VoIP calls. Third, making backups using dedicated hardware is typically more expensive and less frequent than performing similar tasks with a cloud-based server. Recovery is easy with a spare SD card.
The virtual machine platforms certainly have their place in the corporate world. And, if your company already has a server farm full of VMware servers, then taking advantage of that platform to host your PBX makes perfect sense. Performance will probably never be an issue, and you’ll avoid the task of babysitting the hardware leaving that to a staff of dedicated employees. And, hopefully, someone else is making frequent backups of your VoIP server. For home users that already have a beefy desktop machine, a VirtualBox-based PBX is certainly an option worth considering although it again puts you in the driver’s seat of dealing with backups, Internet outages, and performance hiccups when your desktop machine is being used for tasks that consume substantial computing resources.
If you haven’t already guessed, our recommended VoIP platform will almost always be cloud-based. Not only does it offload most server and network management headaches, but more often than not, it’s a more dependable platform with better performance at a comparable or less expensive cost than using your own hardware. So here’s the Golden Nugget of our findings. When it comes to cloud providers, you can forget the old adage that you get what you pay for. You don’t. Our experience suggests it’s just the opposite when it comes to running a VoIP server. With cloud providers, what you typically get by paying more is an improvement in the odds that your provider will still be around when next year rolls around. Getting over that hurdle is simple. Make frequent backups. If there are a multitude of available providers offering similar services, backups are the best insurance you can have, and they cost you almost nothing. In fact, Incredible Backup handles the task with ease AND reliability. Once you get past the vendor longevity issue, the only things that really matter with a cloud platform are stability and performance. While the high-end providers certainly deliver stability, our experience suggests their performance is nothing short of abysmal unless you’re willing to pay through the nose. By way of example, our experimental Google Cloud server running as a $25/month Standard VPS instance with zero daily calls still receives regular alerts from Google recommending that the instance be upgraded to the next pricing tier which starts at $48.95/month. Performance-wise, our subjective comparison of the $25/month Google Cloud instance is virtually identical to what we are seeing on a stand-alone $35 Raspberry Pi. As a VoIP server platform, the so-called free tier with Google Cloud that provides 600K of RAM and a shared virtual CPU is laughable, and that’s being charitable.
We haven’t spent a lot of time using Amazon EC2 in the past couple years primarily because their platform was even more expensive than Google’s. But, if money is no object, it’s certainly a hosting platform worth exploring. For most VoIP applications, it doesn’t make good financial sense.
That narrows our search for the perfect VoIP platform down to two categories: the KVM and OpenVZ platforms. As a general rule of thumb, with a given provider’s offerings you can expect performance to be comparable but you typically will pay at least double for a KVM platform as opposed to an OpenVZ platform with similar RAM, storage, and bandwidth. In a nutshell, KVM servers provide your virtual machine with its own Linux kernel while OpenVZ servers share a kernel over which you have no control. If you run a VoIP application that requires kernel access, this matters. If you plan to expose your server to the public Internet, the KVM option also is desirable because it allows you to run ipset in conjunction with the Linux firewall to block entire countries from accessing your server. In the case of Incredible PBX servers which rely upon a firewall limiting access to whitelisted IP addresses, there is little reason to choose the KVM platform based solely upon performance or security.
The elephant in the room with providers below the Google and Amazon tier is reliability. In the case of Digital Ocean and Vultr, they both have been around for many years now with excellent ratings in virtually every category. Both provide financial support for our open source projects through referral revenue, but we’d use them anyway. The virtual machine pricing from the two companies is almost identical. Except for extremely busy VoIP implementations, their 1GB RAM offering has proven to be a perfect choice at $5 a month. If you don’t mind paying by the year, you can’t beat OVH’s current $2.25/month KVM offering with 2GB RAM and 20GB SSD. They, too, have been around for years. At one time or another, OVH hosted much of 3CX’s cloud infrastructure. All offer scaling options to meet even the most demanding requirements. On the D.O. and Vultr platforms, you can add automatic backups for an additional $1 a month (20% surcharge) which is dirt cheap insurance. We have run both Incredible PBX and 3CX servers on all of these platforms with no outages or other issues… and weekly backups. Both Digital Ocean and Vultr also provide excellent web tools to manage your server, and the chance of any of these providers going out of business is extremely remote. We highly recommend all of them.
FULL DISCLOSURE: We have no business relationship with OVH or any of the following VPS providers and receive no referral commissions of any kind from any of them.
For some users and especially those that just want to learn about VoIP and tinker, there is yet another tier of providers. At roughly $1/month, their VPS services are a fraction of the cost of Digital Ocean and Vultr, but backups become your responsibility and at least one previous provider that many of us used went out of business. Those without a backup lost everything.
Choosing one of these providers comes down to balancing the risks versus the financial savings. We have nearly a dozen of these $1/month servers in operation all across the United States. While the VPS providers are different, almost all of the servers are hosted by ColoCrossing in Los Angeles, New York, Chicago, Dallas, or Atlanta. These VPS providers typically rent machines directly from ColoCrossing, and the performance of their VPS offerings varies depending upon the number of users each provider authorizes on each server. Some are obviously more greedy than others. And we’ve actually done the hard work of finding the reliable ones while rejecting at least as many that proved to be pretty awful.
Server locations and special signup details for these VPS providers are documented in our previous article. Average cost is about $1/month on an annual contract with a 1Gbit port or *free 1Gbit port upgrade on request based upon LowEndBox offer. All offer money-back guarantees for at least 24 hours so you can do your own testing if you hurry. Protect yourself by paying with PayPal which gives you 6 months to dispute a charge if the provider happens to go belly up. NOTE: The sort order below reflects our subjective performance evaluation.
| Provider | RAM | Disk | Bandwidth | Performance as of 12/1/19 | Cost |
|---|---|---|---|---|---|
| CrownCloud KVM (LA) | 1GB | 20GB + Snapshot | 1TB/month | 598Mb/DN 281Mb/UP 2CPU Core | $25/year Best Buy! |
| Naranjatech KVM (The Netherlands) | 1GB | 20GB | 1TB/month | Hosting since 2005 VAT: EU res. | 20€/year w/code: SBF2019 |
| BudgetNode KVM (LA) | 1GB | 40GB RAID10 | 1TB/month | Also available in U.K PM @Ishaq on LET before payment | $24/year |
| FreeRangeCloud KVM (Ashburn VA, Winnipeg, Freemont CA) | 1GB | 20GB SSD | 3TB/month | Pick EGG loc'n Open ticket for last 5GB SSD | $30/year w/code: LEBEGG30 |
Do we recommend these providers? Absolutely, with a couple of caveats. First, there is no guarantee that one or more of them may not go out of business at some point. The odds of several of them going under at the same time are fairly slim since none are related that we’re aware of. Second, make frequent backups when you make changes to your PBX and copy the Incredible Backups to a different location. Third, bring up a second VPS platform in a different location and keep it current with your latest backup. You could bring up all six of these platforms for roughly the same monthly cost as one Digital Ocean or Vultr virtual machine that’s running with automatic backups. If you can’t afford a second $1/month VPS platform, then at least create a matching VirtualBox platform, restore your backup, and make sure it is functional before deploying your VPS in the Cloud. It’s in your hands now. Enjoy!
Originally published: Monday, April 8, 2019
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
A Better Way to Deploy Incredible PBX in the Google Cloud
Last week we introduced you to Incredible PBX 13-13.10 for the Google Cloud. This week we’ll take off the training wheels and show you how to deploy Incredible PBX in the Google Cloud in exactly 3 minutes using Google Cloud’s Image repository. And you can repeat the drill to deploy as many PBXs as you like at 3 minutes a pop. If you’re still cruising along on your $300 credit from Google, then your PBX should be cost-free for the entire first year. After that, you can decide which Google Cloud Machine Type best meets your requirements and those of your wallet. The free tier is an option, but don’t expect much more performance-wise than what you’d get with the original Raspberry Pi. As the saying goes, "It ain’t pretty, but it works." We would encourage you to move up to the Standard machine type for consistent performance.
Before we get started, let us just offer a little constructive criticism regarding Google’s methodology. If a developer builds an application as we have and wants to make it publicly available at no cost, wouldn’t it make sense to allow the developer to host the image in the Google Cloud (for a fee) so that other users could quickly deploy it on their own Google Cloud platforms? That apparently makes too much sense so Google requires you to jump through all sorts of hoops to use free software unless we’re willing to type in the Google email address of every user authorized to deploy the software. Sorry but we’ve got better ways to waste our time. This is the corporate mentality run amuck. Don’t Be Evil, Google. Remember?
So here’s the drill to get you to the place that Google already should have provided. Download the 3GB tarball image to your desktop from SourceForge. After you’ve created your Google Cloud account, create a Bucket (storage locker) on the platform to house your files and upload the tarball into your own Bucket. Next, transform the tarball into what Google calls an Image that can be used to quickly build VM Instances (5 minutes). Finally, start up the instance. The Incredible PBX installer will work its magic letting you set your passwords, and then your PBX platform is ready for use (3 minutes). The real install time is under 10 minutes, but Google has managed to turn it into a project of an hour or more depending upon the speed of your Internet connection. Our apologies, but it beats the tedium of last week’s methodology.
Downloading Incredible PBX for the Google Cloud
Unlike Google and to its eternal credit, SourceForge still hosts open source projects with tarballs of enormous size which can be downloaded at no cost other than what your Internet service provider may charge for bandwidth. Begin your Incredible PBX adventure by downloading the tarball image (3GB) which was designed specifically for the Google Cloud. Depending upon the speed of your Internet connection, this takes some time. Here’s the link.
Creating a Google Cloud Account
If you haven’t already done so, hop over to https://cloud.google.com/free and claim your $300 credit by signing up for a Google Cloud account.
Creating a Bucket in the Google Cloud
To begin, log in to your Google Cloud Console using your Google credentials. If you haven’t already done so, Create a Project from your Dashboard. This Project will house your Compute Engine VM Instances. In Plain English, a Google Cloud VM Instance is nothing more than an application that happens to run in the Google Cloud.
Next, click on the 3-bar image in the upper left corner of your Dashboard. This exposes the Navigation Menu. Scroll down to the STORAGE section and choose Storage -> Browser.
Click on the CREATE BUCKET button. When the dialog window opens, Name your bucket something unique and creative in lower case letters. Fill in the rest of the form as shown and choose the Region in which you want to store your stuff. Then click Create.
Uploading Incredible PBX into Google Cloud Bucket
Once you have created your Bucket, the Bucket Details dialog will open. Click on the Upload Files tab and choose the Incredible PBX tarball that you downloaded from SourceForge. Or you can simply drag the file to the area reserved for uploads in the dialog window.
Once the file upload completes, the Browser window will appear displaying your Bucket. You can click on the Bucket name to display the files in your Bucket which should now include the uploaded Incredible PBX tarball:
Transforming Incredible PBX Tarball into an Image
Google Cloud can create Instances from Images, but not from tarballs in your Bucket. So the next step is to create an Image from the Incredible PBX tarball. Once that is done, you can delete the tarball and bucket from your Google Cloud platform so you don’t have to pay monthly storage fees. Up to this step is where Google could have handled setup transparently by simply allowing us to share our bucket with anonymous users without this knuckle drill, but…
So now we need to create an Image which will transform the Incredible PBX tarball into a format that can be used to create Instances.
Click on the Navigation Menu (the 3-bar image in the upper left corner of your Dashboard). Navigate to COMPUTE -> Compute Engine -> Images. Click CREATE IMAGE.
When the Create Image dialog opens, fill in the form as shown below and click on the Browse button to choose the Incredible PBX tarball from your Bucket. Then click Create.
Creating an Instance from a Cloud Image
It takes about 5 minutes for Google Cloud to transmogrify the Incredible PBX tarball into an Image that can actually be used to create Instances. So be patient. Once your image has been created, it will appear at the top of the Images listing.
Click on the checkbox 
to the left of the Image to select it as shown above. Then click CREATE INSTANCE at the top of the form.
The Create Instance dialog window will appear. Fill in the form as shown above using a unique Name for your Instance. Adjust the Region to match your closest location. This choice may also affect the performance of your instance so picking the default is not a good idea if you want to stick with the freebie platform. Note that the Standard Machine Type (1vCPU) is selected by default. If you still have remaining credits, this won’t be a problem. Otherwise, you’ll have to pay about $25/month for this Machine Type level once your credits expire. We’ve had fair to good results using the Small Machine Type which costs under $15/month.
HINT #1: Never use the default zone for your PBX if you plan to use one of the shared vCPU machine types (micro or small). If you prefer the freebie which we strongly discourage because of performance issues, change the Machine Type to micro in the pull-down. Also note that the Boot Disk defaults to 10GB in size. This won’t work for long, and we’d recommend upping it to at least 20GB. Up to 30GB is provided at no cost using the micro Machine Type. Simply click the Change button to adjust the disk size. Once you’ve made your desired changes, click Create to build the Incredible PBX instance and bring it on line.
HINT #2: If you’re not going to move up to at least the small Machine Type, we would strongly urge you try one of our recommended $1/month VPS providers, all of whom offer considerably better performance at much less cost. In fact, you can bring up a redundant platform with a second VPS provider and still spend about the same money for a year that you would spend with a Google Cloud Standard VPS for one month.
While your Instance is being created and activated, navigate to COMPUTE -> Compute Engine -> VM Instances to display the status of your instances and to decipher the public IP address of your server. After you complete the next section, we’ll make a couple additional modifications using the Google Cloud Console by changing your public IP address from ephemeral (dynamic) to static and and adjusting the Google Cloud firewall. Delay making these changes at this time for the reason covered in the Word of Caution which follows.
A Word of Caution: Incredible PBX for the Google Cloud installs with a default root password of 
That obviously makes your running instance susceptible to compromise if someone else reads this article. So IMMEDIATELY after creating and activating a new Incredible PBX instance, make sure you complete the setup process in the next step during which you will be prompted to reset all of your passwords including the root password.
Completing the Incredible PBX Setup Process
Login to Incredible PBX as root using the default password at the public IP address of your instance using SSH or Putty. The Incredible PBX license agreement should display. If not, your server may have already been compromised. Accept the license agreement and enter very secure passwords for your server when prompted. Once the setup process finishes, reboot your server and wait about a minute for the reboot process to finish. Then log back into your server and allow the Automatic Update Utility to bring your server up to current specs. Once the pbxstatus screen displays, make sure everything is up and running. If not, wait another minute and rerun pbxstatus. Now issue the command user and make certain that you are the only root user on your server. If not, or if you didn’t see the license agreement when you first logged in, or if you couldn’t log in with the default root password, immediately shut down and destroy your instance and create a new one from your Google Cloud Image as documented in the previous section. TIP: If you see connection refused when you first attempt to log in, don’t be alarmed. Just count to 60 and try to log in again. The instance has to have time to boot up after activation before you can log in.
Finalizing Your Google Cloud Setup
Now that you’ve completed the Incredible PBX setup process, here are a couple of changes that need to be made using the Google Cloud GUI. First, you’ll need to permanently assign your IP address to your instance so that you don’t risk having Google change it when your server is rebooted. We also need to make a couple adjustments in the Google Cloud Firewall. Login again to the Google Cloud Console using your Google credentials.
From the Navigation Menu scroll down to the NETWORKING section and choose VPS Network -> External IP Addresses. Change the Type of your existing address to Static and Name it staticip. Next, choose Firewall Rules in the VPS Network section and click CREATE FIREWALL RULE. Fill in the template like the following leaving the other fields with their default entries. Then click CREATE.
Name: incrediblepbx-udp Target Tags: udp-in Source IP Range: 0.0.0.0/0 Protocols/Ports:
If you plan to use HTTPS with your server, you’ll also need to add another firewall rule similar to the existing default-allow-http rule. Simply change the Port to tcp:443 and Name it default-allow-https with a Target Tag of https-in.
CAUTION: Before these firewall rules will be activated for your instance, they also must be specified in the Network Tags section for your instance by adding the udp-in and https-in tags and restarting your instance.
It should be noted that Incredible PBX includes its own Travelin’ Man 3 firewall that manages a whitelist of IP addresses that are allowed ANY access to your server. So we will primarily use the firewall component of the Google Cloud instance to allow sufficient access to Incredible PBX to allow it to actually control server access.
Once you’ve verified that your instance is functioning properly, it’s safe to go back to your Bucket and delete it together with its contents. This will save you having to pay monthly storage fees even though they are quite reasonable.
Getting Started with Incredible PBX
Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX® 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured above. HINT: You can always change it if you happen to forget it: /root/admin-pw-change
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. Some require trunk registration using credentials handed out by these providers. Others including Skyetel use the IP address of your PBX to make connections. 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 from registered trunks to a Demo IVR. And we’ve built dozens of trunks for some of the best providers in the business. Sign up with the ones you prefer, plug in your credentials, and you’re done.
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). Keep in mind that you only pay for the calls you make with each provider so you have little to lose by choosing several. The PIAF Forum also has dozens of recommendations on VoIP providers.
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.
Introducing Skyetel SIP Trunking for Incredible PBX
As frequent visitors already know, Skyetel is a Platinum Sponsor of Nerd Vittles and our open source projects including Incredible PBX. Their financial support keeps the lights on while all of our software remains free for the taking. Today we’re pleased to introduce a special new Skyetel offering for Nerd Vittles readers. If you loved BOGO deals at your favorite grocery store, then you’re going to love this new Skyetel offer which starts today. By signing up through this Nerd Vittles link, Skyetel will match any deposit originally made to your new account up to $250. For example, if you deposit $50, you’ll get $100 of SIP trunking service credit. Deposit $250, and you’ll get $500 of SIP trunking service credit. Basically, it’s half price service, and you get to choose how much you’d like. Skyetel also offers free porting of your DIDs for the first 60 days after you open your account plus a 10% reduction in your current origination rate and DID costs by presenting your last month’s bill.1 Complete details and configuration instructions on the Skyetel service are available in this tutorial. It only takes a minute or two to get up and running. Effective 10/1/2023, $25/month minimum spend at Skyetel is required.
Adding Skyetel Trunks to Incredible PBX
The Skyetel trunks were configured as part of the default install of Incredible PBX. All that’s required on your part is to sign up for Skyetel service to take advantage of the Nerd Vittles special offer. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the BOGO credit for your account by referencing the Nerd Vittles special offer. Greed will get you nowhere. Credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, open another ticket and attach a copy of your last month’s bill. See footnote 1 for the fine print. If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. Easy Peasy!
Unlike many VoIP providers, Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. A typical setup for use with Incredible PBX®, Asterisk®, or FreePBX® would look like the following:
- Name: MyPBX
- Priority: 1
- IP Address: PBX-Public-IP-Address
- Port: 5060
- Protocol: UDP
- Description: server1.incrediblepbx.com
To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.
Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.
Configuring a Skyetel Inbound Route
Because there is no SIP registration with Skyetel, incoming calls to Skyetel trunks will NOT be sent to the Default Inbound Route configured on your PBX because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes -> Add Inbound Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired and click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs after configuring the Inbound Routes.
With the included Incredible Fax add-on, you can enable Fax Detection under the Fax tab. And, if you’d like CallerID Name lookups using CallerID Superfecta, you can enable it under the Other tab before saving your setup and reloading your dialplan.
Configuring a Skyetel Outbound Route
If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. It’s preconfigured to support Skyetel in Connectivity -> Outbound Routes -> Add Outbound Route. The recommended setup is shown below. Just add the CallerID Number you wish to associate with your outbound calls through Skyetel:
Under the Dial Patterns tab, you’ll find the default rules as shown below. Adjust them to meet your own requirements.
There are a million ways to design outbound calling schemes on PBXs with multiple trunks. One of the simplest ways is to use no dial prefix for the primary trunk and then use dialing prefixes for the remaining trunks.
Another outbound calling scheme would be to assign specific DIDs to individual extensions on your PBX. Here you could use NXXNXXXXXX with the 1 Prepend as the Dial Pattern with every Outbound Route and change the Extension Number in the CallerID field of the Dial Pattern. With this setup, you’d need a separate Outbound Route for each group of extensions using a specific trunk on your PBX. Additional dial patterns can be added for each extension designated for a particular trunk. A lower priority Outbound Route then could be added without a CallerID entry to cover extensions that weren’t restricted or specified.
HINT: Keep in mind that Outbound Routes are processed by FreePBX in top-down order. The first route with a matching dial pattern is the trunk that is selected to place the outbound call. No other outbound routes are ever used even if the call fails or the trunk is unavailable. To avoid failed calls, consider adding additional trunks to the Trunk Sequence in every outbound route. In summary, if you have multiple routes with the exact same dial pattern, then the match nearest to the top of the Outbound Route list wins. You can rearrange the order of the outbound routes by dragging them into any sequence desired.
Audio Issues with Skyetel
If you experience one-way or no audio on some calls, make sure you have filled in the NAT Settings section in the GUI under Settings -> Asterisk SIP Settings -> General. In addition to adding your external and internal IP addresses there, be sure to add your external IP address in /etc/asterisk/sip_general_custom.conf like the following example and restart Asterisk:
externip=xxx.xxx.xxx.xxx
If you’re using PJSIP trunks or extensions on your PBX, implement this fix as well.
Receiving SMS Messages Through Skyetel
Most Skyetel DIDs support SMS messaging. Once you have purchased one or more DIDs, you can edit each number and, under the SMS & MMS tab, you can redirect incoming SMS messages to an email or SMS destination of your choice using the following example:
Sending SMS Messages Through Skyetel
We’ve created a simple script that will let you send SMS messages from the Linux CLI using your Skyetel DIDs. In order to send SMS messages, you first will need to create an SID key and password in the Skyetel portal. From the Settings icon, choose API Keys -> Create. Once the credentials appear, copy both your SID and Password. Then click SAVE.
Next, from the Linux CLI, issue the following commands to download the sms-skyetel script into your /root folder. Then edit the file and insert your SID, secret, and DID credentials in the fields at the top of the script. Save the file, and you’re all set.
cd /root wget http://incrediblepbx.com/sms-skyetel chmod +x sms-skyetel nano -w sms-skyetel
To send an SMS message, use the following syntax where 18005551212 is the 11-digit SMS destination: sms-skyetel 18005551212 "Some message"
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). With a cloud-based PBX, you need 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.
One cautionary note if you have multiple SIP softphones behind the same NAT-based router. Getting SIP packets routed back to the appropriate desktop machine can be problematic and typically results in missing audio on calls. The easy workaround is to set up the NeoRouter VPN on both your instance and each of your desktop computers. Then register the softphones to the NeoRouter private IP address of your instance. The NeoRouter client already is installed on your server, but you’ll need to set up a NeoRouter server somewhere and connect to it by running nrclientcmd.
Introducing the Incredible PBX Security Model
Incredible PBX includes one of the most secure turnkey PBX implementations on the planet. As configured, it is protected by both Fail2Ban and a hardened configuration of the IPtables Linux firewall. This release also includes Port Knocker for simple, secure access from any remote computer or smartphone. You can get up to speed on how the technology works by reading the Nerd Vittles tutorial. Your Port Knocker credentials are stored in /root/knock.FAQ together with activation instructions for your server and mobile devices. The NeoRouter VPN client also is included for rock-solid, secure connectivity to remote users. Read our previous tutorial for setup instructions. As configured, nobody can access your PBX without 
your credentials AND 
an IP address that matches the IP address of your server or the PC from which you installed Incredible PBX. You can whitelist additional IP addresses by running the command-line utility /root/add-ip. You can remove whitelisted IP addresses by running /root/del-acct. Incredible PBX is preconfigured to let you connect to many of the leading SIP hosting providers without additional firewall tweaking. The Google Cloud firewall adds an extra layer of protection.
The IPtables firewall is a complex piece of software. If you need assistance with configuring it, visit the PIAF Forum for some friendly assistance.
Incredible Backup and Restore
We’re pleased to introduce our latest backup and restore utilities for Incredible PBX. Running /root/incrediblebackup13 will create a backup image of your server in /tmp. This backup image then can be copied to any other medium desired for storage. To restore it to another Incredible PBX server, simply copy the image to a server running Asterisk 13 and the same version of the Incredible PBX GUI. Then run /root/incrediblerestore13. Doesn’t get much simpler than that.
Incredible PBX Automatic Update Utility
Every time you log into your server as root, Incredible PBX will ping the IncrediblePBX.com web site to determine whether one or more updates are available to bring your server up to current specs. We recommend you log in at least once a week just in case some new security vulnerability should come along.
In the meantime, we encourage you to sign up for an account on the PIAF Forum and join the discussion. In addition to providing first-class, free support, we think you’ll enjoy the camaraderie.
Upgrading to IBM Speech Engines
If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. With IBM’s services, you have a choice of free or commercial tiers. Let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada.
Getting Started with IBM Watson TTS Service
We’ve created a separate tutorial to walk you through obtaining and configuring your IBM Watson credentials. Start there.
Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:
cd /var/lib/asterisk/agi-bin ./install-ibmtts-dialplan.sh nano -w ibmtts.php
Insert your credentials in $IBM_username and $IBM_password. For new users, your $IBM_username will be apikey. Your $IBM_password will be the TTS APIkey you obtained from IBM. Next, verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.
Getting Started with IBM Watson STT Service
Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the Speech to Text app. Choose a Region to deploy in, choose your Organization from the pull-down menu, and select STT as your Space. Choose the Standard Pricing Plan. Then click Create. When Speech to Text Portal opens, click the Service Credentials tab. In the Actions column, click View Credentials and copy down your STT username and password.
Finally, login to your Incredible PBX server and issue these commands to edit getnumber.sh:
cd /var/lib/asterisk/agi-bin nano -w getnumber.sh
Insert apikey as your API_USERNAME and your actual STT APIkey API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:
sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf asterisk -rx "dialplan reload"
Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines."
Using Gmail as a SmartHost for SendMail
Many Internet service providers including Google block email transmissions from downstream servers (that’s you) to reduce spam. The simple solution is to use your Gmail account as a smarthost for SendMail. Here’s how. Log into your server 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://incrediblepbx.com/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:
chmod 600 client-info makemap -r hash client-info.db < client-info cd .. make service sendmail restart
Finally, send yourself a test message. Be sure to check your spam folder!
echo "test" | mail -s testmessage yourname@yourdomain.com
Check mail success with: tail /var/log/mail.log. If you have trouble getting a successful Gmail registration (especially if you have previously used this Google 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.
Originally published: Monday, April 1, 2019
Continue Reading: Configuring Extensions, Trunks & Routes
Don't Miss: Incredible PBX Application User's Guide covering the 31 Whole Enchilada apps
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 VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
- In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. [↩]
Spring Is Sprung: Taking Incredible PBX to the Google Cloud
Let’s chat about Google. We’ve had a love-hate relationship with Google for the past decade. For every good deed they do, they always manage to follow it up with a swift kick in the teeth… or lower. Business Insider recently catalogued all of the Google carnage over the years. And that ignores the train wreck that many VoIP users experienced with Google Voice. So we have mixed emotions about writing this column at all. But, suffice it to say, you won’t have to worry about Google’s Cloud Platform disappearing. It’s as well entrenched in the Google profit center as their advertising juggernaut.
I’ll begin with a story about a former neighbor of ours that was the IT Director at a major university. Maintaining their server farm with staff, labor, and hardware had simply become too expensive and too painful for the university to absorb so he made what at the time appeared to be a very brave decision. He decided to move all of the computing resources of the university to the Google Cloud. I haven’t spoken to him recently, but I can tell you the day it was completed was one of the happiest days of his life. Taking hardware acquisition, hardware maintenance, and facilities management out of the IT equation is great for your blood pressure. And the university actually has saved boatloads of money.
Is the Google Cloud right for everyone? Of course not. But you’ve got nothing to lose by trying it because Google is going to spot you $300 for the first year to get started. So we’d recommend you make the decision whether to continue AFTER you’ve spent the $300 you found lying on the sidewalk. Today we’ll show you how to build the always-free platform which probably will suffice for home users and small businesses in perpetuity. After your first year, the only charge would be a little chump change for bandwidth each month. If you decide not to use it as your PBX platform, it still would come in handy as a VPN server platform for an application such as NeoRouter. Pricing details here.
We want to start today by thanking Stewart Nelson on the DSLR Forum for his pioneering work on this beginning over a year ago. To start, hop over to https://cloud.google.com/free and claim your $300 by signing up for a Google Cloud account.
CAUTION: Before you embark on this adventure, we would encourage you to read through this article AND read our followup article which documented a much easier and simpler implementation scheme.
Creating a Google Cloud Instance for Incredible PBX
Once you have your account set up, it’s time to create your first project. Navigate to https://console.cloud.google.com. In the COMPUTE section of the dashboard, click Compute Engine -> VM Instances. Then click CREATE PROJECT and name it. Now click CREATE INSTANCE and Name it incrediblepbx. The instance name becomes the hostname for your virtual machine. If you want to remain in the Free Tier, choose f1-micro instance as the Machine Type and choose a U.S. Region (us-central1, us-east1 or us-west1). We strongly recommend installing your VPS using the N1-standard-1 as the Machine Type. It costs about 3 cents an hour and will save you several hours of tedious waiting. Once you complete the install, you then can shut down the server, downgrade to the f1-micro Machine Type, and restart your instance. For the Boot Disk, choose CentOS 6 and expand the disk storage to at least 20GB (30GB is available with the Free Tier). For the Firewall setting, enable HTTP and optionally HTTPS, if desired. Check your entries carefully and then click the Create button.
When your virtual machine instance comes on line, jot down the assigned public IP address. We’ll need it in a minute. Now click on the SSH pull-down tab and choose Open in a Browser Window. Now we need to set a root password and adjust the SSH settings so that you can login from your desktop computer using SSH or Putty. This is important since the Incredible PBX installer will whitelist the IP address of your desktop PC as part of the setup process. You don’t want to lock yourself out of your virtual machine.
sudo passwd root su root nano -w /etc/ssh/sshd_config
When the editor opens the SSH config file, add the following entries. Then save the file and restart SSH: service sshd restart
PermitRootLogin yes PasswordAuthentication yes
You now should be able to log in to your instance as root from your desktop computer using SSH or Putty. Test it to be sure: ssh root@server-IP-address
Before we leave the Google Cloud Dashboard, let’s make the assigned public IP address permanent so that it doesn’t get changed down the road. Keep in mind that, if you ever delete your instance, you also need to remove the assigned static IP address so you don’t continue to get billed for it. From Home on the Dashboard, scroll down to the NETWORKING section and choose VPS Network -> External IP Addresses. Change the Type of your existing address to Static and Name it staticip. Next, choose Firewall Rules in the VPS Network section and click CREATE FIREWALL RULE. Fill in the template like the following leaving the other fields with their default entries. Then click CREATE.
- Name: incrediblepbx-udp
- Target Tags: udp-in
- Source IP Range: 0.0.0.0/0
- Protocols/Ports: check udp: all
If you plan to use HTTPS with your server, you’ll also need to add another firewall rule similar to the existing default-allow-http rule. Simply change the Port to tcp:443 and Name it default-allow-https with a Target Tag of https-in.
CAUTION: Before these firewall rules will be activated for your instance, they also must be specified in the Network Tags section for your instance by adding the udp-in and https-in tags and restarting your instance.
It should be noted that Incredible PBX includes its own Travelin’ Man 3 firewall that manages a whitelist of IP addresses that are allowed ANY access to your server. So we will primarily use the firewall component of the Google Cloud instance to allow sufficient access to Incredible PBX to allow it to actually control server access.
Installing Incredible PBX in the Google Cloud
If you’ve installed previous iterations of Incredible PBX, here is a thumbnail sketch of the install procedure. After logging into your server as root from a desktop PC using SSH or Putty, issue the following commands:
yum -y update yum -y install net-tools nano wget tar wget http://incrediblepbx.com/incrediblepbx-13-13-LEAN.tar.gz tar zxvf incrediblepbx-13-13-LEAN.tar.gz rm -f incrediblepbx-13-13-LEAN.tar.gz # add swap file to your instance ./create-swapfile-DO # kick off Phase I install ./IncrediblePBX-13-13.sh # after reboot, kick off Phase II install ./IncrediblePBX-13-13.sh # adjust TM3 firewall to block Google Cloud locals sed -i 's|10.0.0.0/8|10.0.0.0/24|' /usr/local/sbin/iptables-custom iptables-restart # add Full Enchilada apps (see below) ./Enchilada-upgrade.sh # add HylaFax/AvantFax (see below) ./incrediblefax13.sh # after reboot, set passwords ./update-passwords # set desired timezone ./timezone-setup # fix permissions clobbered by Google Cloud install chown -R asterisk:asterisk /var/lib/asterisk amportal restart # set up NeoRouter client, if desired nrclientcmd # check network speed wget -O speedtest-cli https://raw.githubusercontent.com/sivel/speedtest-cli/master/speedtest.py chmod +x speedtest-cli ./speedtest-cli
WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration.
Using the Incredible PBX 13-13 Web GUI
NOTE: If you plan to upgrade to the Whole Enchilada, you can skip this section. It’s for those that wish to roll their own PBX from the ground up.
Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it.
To get a basic system set up so that you can make and receive calls, you’ll need to add a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You’ll also need a SIP phone or softphone to use as an extension on your PBX. Our previous tutorial will walk you through this setup procedure. Over the years, we’ve built a number of command line utilities including a script to preconfigure SIP trunks for more than a dozen providers in seconds. You’ll find links to all of them here.
Continue Reading: Configuring Extensions, Trunks & Routes
Upgrading to Incredible PBX Whole Enchilada
There now are two more pieces to put in place. The sequence matters! Be sure to upgrade to the Whole Enchilada before you install Incredible Fax. If you perform the steps backwards, you may irreparably damage your fax setup by overwriting parts of it.
The Whole Enchilada upgrade script now is included in the Incredible PBX LEAN tarball. If you have an earlier release, you may need to download the Whole Enchilada tarball as documented below. Upgrading to the Whole Enchilada is simple. Log into your server as root and issue the following commands. Try issuing just the last command first to see if the enchilada upgrade script already is in place. Otherwise, execute all of the commands below. Be advised that the upgrade will overwrite all of your existing Incredible PBX setup including any extensions, trunks, and routes you may have created previously. You also will be prompted to reset all of your passwords as part of the upgrade.
cd /root ./Enchilada*
If you accidentally installed Incredible Fax before upgrading to the Whole Enchilada, you may be able to recover your Incredible Fax setup by executing the following commands. It’s worth a try anyway.
amportal a ma install avantfax amportal a r
Installing Incredible Fax with HylaFax/AvantFax
You don’t need to upgrade to the Whole Enchilada in order to use Incredible Fax; however, you may forfeit the opportunity to later upgrade to the Whole Enchilada if you install Incredible Fax first. But the choice is completely up to you. To install Incredible Fax, log into your server as root and issue the following commands:
cd /root ./incrediblefax13.sh
After entering your email address to receive incoming faxes, you’ll be prompted about two dozen times to choose options as part of the install. Simple press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password; however, you first will be prompted for your Apache admin credentials which were set when you installed Incredible PBX 13-13 LEAN or the Whole Enchilada. Then you’ll be asked to change your AvantFax password.
Upgrading to IBM Speech Engines
If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. With IBM’s services, you have a choice of free or commercial tiers. Let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada.
Getting Started with IBM Watson TTS Service
We’ve created a separate tutorial to walk you through obtaining and configuring your IBM Watson credentials. Start there.
Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:
cd /var/lib/asterisk/agi-bin ./install-ibmtts-dialplan.sh nano -w ibmtts.php
Insert your credentials in $IBM_username and $IBM_password. For new users, your $IBM_username will be apikey. Your $IBM_password will be the TTS APIkey you obtained from IBM. Next, verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.
Getting Started with IBM Watson STT Service
Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the Speech to Text app. Choose a Region to deploy in, choose your Organization from the pull-down menu, and select STT as your Space. Choose the Standard Pricing Plan. Then click Create. When Speech to Text Portal opens, click the Service Credentials tab. In the Actions column, click View Credentials and copy down your STT username and password.
Finally, login to your Incredible PBX server and issue these commands to edit getnumber.sh:
cd /var/lib/asterisk/agi-bin nano -w getnumber.sh
Insert apikey as your API_USERNAME and your actual STT APIkey API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:
sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf asterisk -rx "dialplan reload"
Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines." Check back next week for the Whole Enchilada apps tutorial.
Adding Skyetel Trunks to Incredible PBX
The Skyetel trunks were configured as part of the default install of Incredible PBX. All that’s required on your part is to sign up for Skyetel service and take advantage of the exclusive Nerd Vittles BOGO offer beginning April 1. Skyetel will match your original deposit of up to $250 which translates into as much as $500 of half-price SIP trunking service. Effective 10/1/2023, $25/month minimum spend required. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the BOGO credit for your account by referencing this Nerd Vittles special offer. Greed will get you nowhere. Credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, open another ticket and attach a copy of your last month’s bill. See footnote 1 for the fine print.1 If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. Easy Peasy!
Unlike many VoIP providers, Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. A typical setup for use with Incredible PBX®, Asterisk®, or FreePBX® would look like the following:
- Name: MyPBX
- Priority: 1
- IP Address: PBX-Public-IP-Address
- Port: 5060
- Protocol: UDP
- Description: server1.incrediblepbx.com
To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.
Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.
Configuring a Skyetel Inbound Route
Because there is no SIP registration with Skyetel, incoming calls to Skyetel trunks will NOT be sent to the Default Inbound Route configured on your PBX because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes -> Add Inbound Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired and click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs after configuring the Inbound Routes.
If you have installed the Incredible Fax add-on, you can enable Fax Detection under the Fax tab. And, if you’d like CallerID Name lookups using CallerID Superfecta, you can enable it under the Other tab before saving your setup and reloading your dialplan.
Configuring a Skyetel Outbound Route
If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. From the GUI, choose Connectivity -> Outbound Routes -> Add Outbound Route. For the setup, we recommend the following using the CallerID Number you wish to associate with your outbound calls through Skyetel:
Enter the Dial Patterns under the Dial Patterns tab before saving your outbound route. Here’s what you would enter for 10-digit and 11-digit dialing. If you want to require a dialing prefix to use the Skyetel Outbound Route, enter it in the Prefix field for both dial strings.
There are a million ways to design outbound calling schemes on PBXs with multiple trunks. One of the simplest ways is to use no dial prefix for the primary trunk and then use dialing prefixes for the remaining trunks.
Another outbound calling scheme would be to assign specific DIDs to individual extensions on your PBX. Here you could use NXXNXXXXXX with the 1 Prepend as the Dial Pattern with every Outbound Route and change the Extension Number in the CallerID field of the Dial Pattern. With this setup, you’d need a separate Outbound Route for each group of extensions using a specific trunk on your PBX. Additional dial patterns can be added for each extension designated for a particular trunk. A lower priority Outbound Route then could be added without a CallerID entry to cover extensions that weren’t restricted or specified.
HINT: Keep in mind that Outbound Routes are processed by FreePBX in top-down order. The first route with a matching dial pattern is the trunk that is selected to place the outbound call. No other outbound routes are ever used even if the call fails or the trunk is unavailable. To avoid failed calls, consider adding additional trunks to the Trunk Sequence in every outbound route. In summary, if you have multiple routes with the exact same dial pattern, then the match nearest to the top of the Outbound Route list wins. You can rearrange the order of the outbound routes by dragging them into any sequence desired.
Audio Issues with Skyetel
If you experience one-way or no audio on some calls, make sure you have filled in the NAT Settings section in the GUI under Settings -> Asterisk SIP Settings -> General. In addition to adding your external and internal IP addresses there, be sure to add your external IP address in /etc/asterisk/sip_general_custom.conf like the following example and restart Asterisk:
externip=xxx.xxx.xxx.xxx
If you’re using PJSIP trunks or extensions on your PBX, implement this fix as well.
Receiving SMS Messages Through Skyetel
Most Skyetel DIDs support SMS messaging. Once you have purchased one or more DIDs, you can edit each number and, under the SMS & MMS tab, you can redirect incoming SMS messages to an email or SMS destination of your choice using the following example:
Sending SMS Messages Through Skyetel
We’ve created a simple script that will let you send SMS messages from the Linux CLI using your Skyetel DIDs. In order to send SMS messages, you first will need to create an SID key and password in the Skyetel portal. From the Settings icon, choose API Keys -> Create. Once the credentials appear, copy both your SID and Password. Then click SAVE.
Next, from the Linux CLI, issue the following commands to download the sms-skyetel script into your /root folder. Then edit the file and insert your SID, secret, and DID credentials in the fields at the top of the script. Save the file, and you’re all set.
cd /root wget http://incrediblepbx.com/sms-skyetel chmod +x sms-skyetel nano -w sms-skyetel
To send an SMS message, use the following syntax where 18005551212 is the 11-digit SMS destination: sms-skyetel 18005551212 "Some message"
Using Gmail as a SmartHost for SendMail
Many Internet service providers including Google block email transmissions from downstream servers (that’s you) to reduce spam. The simple solution is to use your Gmail account as a smarthost for SendMail. Here’s how. Log into your server 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://incrediblepbx.com/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:
chmod 600 client-info makemap -r hash client-info.db < client-info cd .. make service sendmail restart
Finally, send yourself a test message. Be sure to check your spam folder!
echo "test" | mail -s testmessage yourname@yourdomain.com
Check mail success with: tail /var/log/mail.log. If you have trouble getting a successful Gmail registration (especially if you have previously used this Google 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.
Finally, let's clean up Fail2Ban so you don't get bombarded by useless email messages. Issue the following commands and, when the editor opens, search for the nobody: line and change the destination from root to devnull. Save the file and then reload your aliases with the last command below:
sed -i 's|you@example.com|nobody@localhost|' /etc/fail2ban/jail.conf nano -w /etc/aliases newaliases
Continue Reading: A Better Way to Deploy Incredible PBX in the Google Cloud
Originally published: Tuesday, March 26, 2019
News Flash: Turn Incredible PBX into a Fault-Tolerant HA Platform for $1/Month
Continue Reading: Configuring Extensions, Trunks & Routes
Don't Miss: Incredible PBX Application User's Guide covering the 31 Whole Enchilada apps
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 VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
- In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. [↩]
Cell Phone Tips for Spring Break and International Travel
With Spring Break upon us and Summer Vacations just around the corner, we wanted to briefly review some of your cellphone and data options for those that might be planning a trip outside the United States. If you’re only going as far as Mexico, Canada, or the U.S. Virgin Islands, then your existing cellular provider in the U.S. may have you covered at no additional cost. If your provider is not AT&T, then check with your carrier. And speaking of AT&T, if you’ll only be outside these covered areas for a brief time, then you may wish to consider enabling AT&T’s International Day Pass which costs you nothing until you use it. On days that you use it in over 100 countries, it’s $10/day with the same talk, text, and data options you currently have in the U.S. No, it’s not a bargain for a 60-day vacation, but it’s a pretty good deal for a week or so when you only need cell service for a few days. No changes in your current AT&T plan are necessary other than enabling the International Day Pass feature. Click on the Get Started Link to enable the service. Be sure to read the fine print.
Once you get past the options in the first paragraph, most of the other economical choices for cell phone and data coverage internationally involve swapping out the SIM card in your phone with a country-specific SIM card from a local provider. The first step is to make absolutely certain that your cell phone is unlocked before you hop on a plane. The procedure varies with different providers so you’ll need to investigate what steps are required. Be sure to also decipher how to verify that your phone is unlocked. Again, with AT&T, it’s a simple matter of visiting their web site and filling out a form. Within 24 hours, you should be good to go.
Now comes the hard part, choosing an alternate provider meeting your travel requirements. This turns on a number of factors such as whether callers in the U.S. need to contact you using a U.S. phone number. If so, then the first paragraph is your best bet if you need to be reached on your existing phone number. If any U.S. phone number will suffice (and you can always forward your cellphone number to this new number), then using an Android phone or iPhone, there’s an easy solution if you have Wi-Fi access or some cellphone data to burn. Simply use a Google Voice phone number and associate it with the new Android or iPhone Google Voice app on your phone. Be sure to enable WiFi/Mobile Data calling in GV Settings, and you’re good to go with almost any smart phone with 4G service. With Wi-Fi, no SIM card is required. Just put your phone in Airplane Mode and enjoy free calling back to the U.S. and Canada.
The next issue to consider is whether you need to make frequent calls or send frequent text messages to those in the U.S. while you are away. If so, then the best choice we’ve found without Wi-Fi access is Orange Holiday Europe.1 Simply buy the $50 card and put it in your cellphone on the day you wish to begin your service. It buys you 10GB of data, 2 hours of calls, and 1000 text messages to almost any phone in the world from 30 European countries including the U.K. Once activated, the card is good for 14 days and includes tethering. It can be renewed for an additional 21.70€ (about $25) which adds another 14 days with an additional 10GB of data, 120 minutes of calls, and another 1,000 SMS messages.
If the phone number of your calls doesn’t matter and you can also take advantage of Google Voice for free calling to and from the U.S., then all you really need is the cheapest SIM card you can find in the country you’re visiting. One word of advice from our frequent traveler friends is don’t buy the SIM card in the airport where they typically are two to five times as expensive. To give you an example, a SIM card with 5GB of data in Madrid can be had for about 10€ per week. For iPhone users, a more flexible SIM card that looked appealing to us was Gigsky which offers regional SIM cards for anywhere in the world. Enjoy your vacation!
Originally published: Monday, March 25, 2019
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
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The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
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Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
- 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. [↩]
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