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Category Archives: Ubuntu/Debian
Big Kahuna: 70 New FreePBX GPL Modules for Incredible PBX
We don’t change the mix of FreePBX® GPL modules in Incredible PBX® 13-13 often although you can easily add or update any of particular interest at any time using the gpl-install scripts included in the distribution. So we’re excited to introduce the 2019 collection of 70 FreePBX GPL modules for those that want to keep their Asterisk® PBX platform loaded with the latest and greatest. We’ve included a batch installer which means ALL of the existing modules get updated with the latest releases from GitHub. Depending upon the speed of your Internet connection, it’s a 5 or 10-minute procedure. Schedule it for a time when the PBX is idle.
Upgrading the FreePBX GPL Modules
The upgrade procedure couldn’t be easier. Log into your server as root. We recommend you make a backup first using the incrediblebackup script in /root. Next, make sure you have at least 50MB of free disk space: df -h
. Then issue these commands and have a cup of coffee:
cd /tmp wget http://incrediblepbx.com/modules13.tar.gz tar zxvf modules13.tar.gz rm -f modules13.tar.gz cd modules13 ./update-modules.sh
When you return, your Incredible PBX 13-13 server will be all shiny and new. You can review the license terms for each module by referencing the table below and calling up the GPL license provisions with a browser pointed to http://server-IP/admin/licenses.
FreePBX GPL Modules Documentation
The FreePBX Dev Team has generously provided excellent documentation for all of the modules. We have arranged them in the same order as the GUI’s menus for ease of use.
Admin Modules
Administrators Module
Asterisk CLI Module
Asterisk Phonebook Module
Backup and Restore Module
Blacklist Module
Bulk Handler
CID Superfecta
CallerID Lookup Sources
Certificate Management Module
Config Edit
Custom Destinations Module
Custom Extensions Module
Feature Codes Module
Module Admin Module
Phone Restart Module
Presence State Module
REST API
Sound Languages
System Recordings Module
User Management Module
Applications Modules
Announcements Module
Call Flow Control Module
Call Recording Module
Callback Module
Conferences Module
DISA Module
Directory Module
Extensions Module
Follow Me Module
IVR Module
Languages Module
Misc Applications Module
Misc Destinations Module
Paging and Intercom Module
Parking Module
Queue Priorities Module
Queues Module
Ring Groups Module
Set CallerID Module
Text to Speech Module
Time Conditions Module
Time Groups Module
Voicemail Blasting Module
Wakeup Calls Module
Connectivity Modules
DAHDI Channel DIDs
Inbound Routes Module
OSS End Point Manager (Disabled)
Outbound Routes Module
Trunks Module
Dashboard
Reports Modules
Asterisk Info Module
Asterisk Logfiles
CDR Reports Module
Call Event Logging (CEL) Module
Print Extensions
Rest API Report
Weak Password Detection
Settings Modules
Advanced Settings
Asterisk IAX Settings
Asterisk Logfile Settings
Asterisk Manager Interface
Asterisk SIP Settings
Extension Settings
Fax Configuration
Music on Hold Module
Pin Sets
Route Congestion Messages
Text to Speech Engines Module
Voicemail Admin
Third Party Addons
UCP
Installing OSS Endpoint Manager
If you have dozens of SIP phones to configure, then you’ll appreciate Andrew Nagy’s terrific OSS Endpoint Manager Module. Here’s how to install it once your Incredible PBX 13-13 server is updated with the new modules above:
cd / wget http://incrediblepbx.com/epm.tar.gz tar zxvf epm.tar.gz ./install-epm.sh
You will also need to install and configure a TFTP server. Here’s the CentOS procedure:
cd /root wget http://incrediblepbx.com/setup-tftp chmod +x setup-tftp ./setup-tftp
Pay particular attention to the firewall instructions which display at the end of the TFTP install procedure. Complete documentation for OSS Endpoint Manager is available here. Enjoy!
Originally published: Monday, February 18, 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.
Adding SIP URI Dialing to Asterisk for Free Worldwide Calling
Since giving up on Google Voice, we’ve been extolling the virtues of SIP URI dialing which gives you unlimited free calls to anyone else in the world that happens to have their own SIP URI address. SIP URIs look very much like email addresses except they’re used to share phone conversations instead of email messages. And, as we’ve mentioned previously, if everyone in the world had their own SIP URI, paying for phone calls would become a thing of the past. We hope you’ll join us in making that happen. As a fallback, give our $50 credit at Skyetel a try.
One of the drawbacks of Asterisk® PBXs using the FreePBX® GUI has been the inability to place outbound SIP URI calls from SIP phones registered as extensions on the PBX. Today we first want to address that shortcoming. Our SIP URI dialing solution for Asterisk should work with any FreePBX-based implementation including Incredible PBX® and Issabel as well as on Raspberry Pi platforms. We’ll wrap things up by providing some tips on obtaining and deploying your own SIP URI at little or no cost and pointing you to some excellent resources that facilitate calling millions of SIP phones around the world at zero cost. All you need is an Internet connection, and we’ll point you to a terrific softphone to begin your adventure.
Let’s begin by examining why SIP URI dialing is a problem with FreePBX. The reason is pretty simple. FreePBX interprets dial strings by matching them against some rules to determine whether you’re making an internal call or a call outside your PBX. It matches internal calls against a list of available internal extensions. External calls are matched against rules defined in your outbound routes which are associated with trunks. Since SIP URI calls don’t match any extension or outbound route, the caller receives a congestion tone.
The traditional workaround has been to define a custom extension using the FreePBX GUI which points to a SIP URI. Then the user can dial the custom extension, and the call will be routed to the defined SIP URI. These custom extensions also can be defined in extensions_custom.conf within the from-internal-custom context. For example, the following dialplan code would let users dial 411 to reach AT&T’s Toll-Free Directory Assistance: exten => 411,1,18005551212@switch.starcompartners.com
.
But there’s a better way. Wouldn’t it be nice to be able to dial any SIP URI from a softphone or to store SIP URI addresses in the phonebook of your SIP phone?1 Well, now you can. Before we actually put the dialplan code in place, let us explain how this will work. First, FreePBX still needs to be able to distinguish a SIP URI call from a "regular call." The reason this gets tricky is because Asterisk typically throws away the destination hostname when you place a call. For example, calls to 8005551212 and 8005551212@sip2sip.info are processed by Asterisk in exactly the same way, i.e. dropping the host address before dialing.
Using the new dialplan code in the next section, here’s how calls will be processed:
User dials Asterisk processes call as ------------------------ --------------------------------------------- 701 internal call to local extension 701 4045551212 external call using NXXNXXXXXX outbound route 2233435945@sip2sip.info SIP URI call to Lenny by acct at sip2sip.info lennybgood@sip2sip.info SIP URI call to alias lennybgood@sip2sip.info
Cautionary Notes: Our code should work fine with any Asterisk 13 and FreePBX 13 or Incredible PBX deployment on any Linux platform; however, with servers other than Incredible PBX, make sure you have added the following entries to sip_general_custom.conf, or you can configure them in the GUI by making the changes in Settings -> Asterisk SIP Settings -> Chan SIP Settings:
srvlookup=yes allowguest=yes
You also need to test a traditional outbound call (e.g. 8005551212) immediately after you finish the install procedure. Monitor the Asterisk CLI (asterisk -rvvvvvvvvvv
) and observe the first few lines of the log after you place a call. The second line will show SIPDOMAIN which should be either the FQDN of your server or an IP address depending upon how you registered your softphone extension. The first line should display the MyDomain variable. If it is empty or doesn’t match the SIPDOMAIN entry, the outbound call will fail. To fix it, add an entry to the Asterisk database from the Asterisk CLI using syntax like the following: database put MyDomain FQDN 10.0.0.11
or database put MyDomain FQDN sip.me.com
where 10.0.0.11 or sip.me.com matches the SIPDOMAIN entry shown on the second line. Then retry your outbound call, and it should complete successfully. We’ve tested this back to the early Asterisk 11 days with FreePBX 2.11 without any problems. If your calls still fail, then you will probably need to remove the new code from your platform until you upgrade to a more current version of Asterisk and FreePBX. The code hasn’t been tested with FreePBX 14 and 15.
Finally, you may want to manually set the CallerID for your outgoing SIP URI calls. From the Asterisk CLI, issue a command for every extension from which you will be placing SIP URI calls, e.g. extension 701 syntax: database put 701 user_sipname "Nerd Uno"
Enabling SIP URI Dialing with FreePBX
To enable SIP URI dialing from phones registered with your Asterisk PBX, we’ll modify the dialplan in order to detect SIP URI dial strings entered into a softphone or retrieved from a phonebook associated with almost any SIP phone. When a SIP URI dial string is detected, we’ll send the call out as requested rather than passing the call through the outbound routes and trunks associated with your PBX. All of this dialplan code is open source and is licensed pursuant to the GPL2 license.
SECURITY ALERT: Never use the SIP URI MOD on a server with a publicly-exposed SIP port as it is possible for some nefarious individual to spoof your FQDN in the headers of a SIP packet and easily gain outbound calling access using your server’s trunk credentials.
FEB. 21 UPDATE: There was a bug in the original code which caused some internal calls to fail including calls to a DISA extension. Simply install the application again, and it will overwrite the previous version.
MAR. 5 UPDATE: A bug was discovered in previous releases that treated 911 and 933 calls as internal calls when, in fact, they should have been routed out using your outbound trunks. Simply install the application again, and it will overwrite the previous version.
MAR. 13 ALERT: This software is not compatible with the Debian, Raspbian, and Ubuntu platforms.
To begin or update your installation, log in to your PBX as root using SSH or Putty and issue these commands:
cd /tmp wget http://incrediblepbx.com/sipuri-mod.tar.gz tar zxvf sipuri-mod.tar.gz rm -f sipuri-mod.tar.gz ./install-sip-uri-mod.sh
Obtaining Your Own SIP URI
There are a number of ways to obtain your own SIP URI. Perhaps the easiest is to set up the open Incredible PBX cloud platform that we introduced several weeks ago. Then you can create as many SIP URIs as you like, and they can be used to perform any task that’s available with Asterisk. If you’re not quite ready to make that leap, a free or almost free SIP URI is available from the following sources. VoIP.ms provides a SIP URI for every subaccount you create. Just set up an internal extension number for the subaccount, and that becomes a SIP URI to connect back to your registered server or SIP phone. In the alternative, VoIP.ms will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. CallCentric provides a SIP URI matching your account number which can be reached @in.callcentric.com. CallCentric will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. LocalPhone provides the same two options as CallCentric: you can be reached by your account number @localphone.com. Or the LocalPhone-assigned iNUM DID can be reached @81.201.82.50. Then there’s pbxes.org. Your account name can be used for SIP URI access @pbxes.org. And, of course, if you’re a 3CX user, you can set up a SIP URI for each extension on your PBX. Just navigate to the Options tab of the desired extension(s) and enter a unique SIP ID for each extension. The SIP URI becomes SIPID@YOUR-3CX-FQDN. SIP URI calls to 3CX Clients on smartphones are also free! This list is not exhaustive. There are now more than 2,000 VoIP networks that support SIP URI access. Using a SIP URI dialing prefix, call any of the referenced networks @sipbbroker.com.2
Choosing a SIP Phone or Softphone
You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum. For today we’ll get you started with one of our favorite (free) softphones, YateClient. It’s available for almost all desktop platforms. Download YateClient from here. Run YateClient once you’ve installed it and enter the credentials for an extension on your PBX. You’ll need the IP address of your server plus your extension number and its password. Fill in the Yate Client template using the IP address of your PBX as well as your extension credentials. Click OK to save your entries.
Once the Yate softphone shows that it is registered, try a test call to Lenny using one of the following SIP URIs: 2233435945@sip2sip.info or 883510001198938@81.201.82.50. Better yet, try out a few Incredible PBX samples from the public server we previously deployed:
Yahoo News Headlines - news@demo.nerdvittles.com Weather by Zip Code - weather@demo.nerdvittles.com Directory Assistance - information@demo.nerdvittles.com Lenny for Telemarketers - lenny@demo.nerdvittles.com
Originally published: Monday, February 11, 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.
- Special thanks to Olivier Adler and voip-info.org for their early work on SIP URI dialing with Asterisk. [↩]
- Some of our links refer users to sites or service providers when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from these providers to help cover the costs of our blog. We never recommend particular products solely to generate commissions. [↩]
Introducing Skyetel: A VoIP Provider for All Seasons
Having been around the block more times than we can remember, suffice it to say it takes a lot to get us excited about a VoIP provider. Let us tick off some criteria to even get our attention: terrific pricing, failsafe reliability, and first class performance. So just imagine our excitement to discover that an early follower of Nerd Vittles now provides one of the most compelling VoIP services we’ve ever tested with triple redundancy in multiple data centers. And Skyetel now has added what, for some, was the most important piece: support for VoIP servers with dynamic IP addresses. While it’s still beta code, it’s easy to use and reliable. There’s yet another hidden benefit. Incredible PBX coupled with Skyetel makes a perfect platform for redundant servers. We’ll cover it in a future article, but here’s the basic design.
Let’s sweeten the pot a bit more. We were looking for a service provider that could offer a compelling price for the hobbyist and home user while also having the depth to provide millions of minutes to organizations and resellers that actually have such a need. Skyetel now offers Nerd Vittles readers two special offers. First, you can claim a $10 credit for your new account simply by opening a ticket once you sign up. Once you have kicked the tires and are satisfied with the service, you won’t want to miss the Nerd Vittles BOGO offer. Skyetel will match your original deposit up to $250. Deposit $50 and Skyetel will double it. Or plan ahead with a $250 deposit and Skyetel will still double it. That translates into $500 of half-price VoIP service! Once you have funded your account with your money, Skyetel will provide 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 Effective 10/1/2023, $25/month minimum spend required. For resellers and high volume users, document your requirements on your Nerd Vittles signup form and let us put you in touch with someone at Skyetel that will make you a deal you can’t refuse. And what does Nerd Vittles get out of this? Glad you asked. We’re delighted to have Skyetel as a platinum sponsor to keep the lights burning and the deals flowing for another decade of articles and open source offerings for our dedicated followers.
Original Skyetel Deposit | Skyetel Deposit Match | Available SIP Service $'s |
---|---|---|
$20 | $20 | $40 |
$50 | $50 | $100 |
$100 | $100 | $200 |
$200 | $200 | $400 |
$250 | $250 | $500 |
We want to also address the elephant in the room. Some have asked about our relationship with Vitelity, a long time sponsor of Nerd Vittles and our open source projects. They’re alive and well. However, the company has gone through several acquisitions in the past few years, and their focus now has shifted more to the reseller and wholesale market. ALL EXISTING VITELITY CUSTOMERS ARE UNAFFECTED BY THIS CHANGE IN DIRECTION. And we are more than happy to put new resellers and wholesalers in touch with someone at Vitelity that can address your requirements. The good news is that you’ll now have two companies to compare while new home users and small businesses have a viable alternative moving forward.
Skyetel’s State-of-the-Art Network Design
Because Skyetel’s system architecture is radically different from most other VoIP providers, we wanted to spend a minute documenting their setup. Typically, a VoIP provider may offer a failover server in case their primary server fails. But all calls flow through the primary server unless there is a system failure. As we noted previously, Skyetel’s current setup includes three redundant data centers, all of which receive incoming calls while being firewalled from each other. Once you place or receive a call from the Skyetel network, their data center is completely removed from the audio path of the call which flows directly between your server and the outside party. Thus, even if the data center experienced a total system failure in the middle of your call, neither you nor the other party would ever know it. This design also eliminates the potential of a man-in-the-middle attack from your VoIP provider’s server.
Skyetel Pricing Overview
This summary is not intended to be an exhaustive listing of all Skyetel services. Follow this link for a complete summary of fees and services. Traditional DIDs are $1 per month. Toll free numbers an additional 20¢ per month. Outbound conversational calls are $0.012 per minute. DIDs can be SMS/MMS enabled for 10¢ per month. E911 service is $1.50 per month. Incoming conversational calls are a penny a minute. CallerID lookups are $0.004 per call. Voicemail transcription is available for 10¢ per message.
Signing Up for Skyetel Service
So here’s the drill to sign up for Skyetel service and take advantage of the Nerd Vittles specials. 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 your free $10 credit to kick the tires. You cannot port in numbers at no cost until you actually fund your account out of your own pocket. Once you have funded your account, open another ticket for the BOGO credit for your account by referencing the Nerd Vittles special offer. You then can initiate your free number porting requests on the portal and request a credit for the porting fees. BOGO credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, 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!
For those that may be concerned that one day, after your credit expires, you could be paying a penny a minute for phone calls, let me provide a little Ma Bell history lesson for you. When my roommate and I were in law school, our typical phone bill often exceeded $200 a month because we both had girlfriends a couple hundred miles up the road. In today’s dollars, that phone bill translates into roughly $1,200 a month. That would have been 120,000 minutes a month at a penny a minute in today’s dollars. So, yes, VoIP is having a profound influence on the AT&T and Verizon Bell Sisters.
Skyetel Endpoint Group Configuration
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
Skyetel DID Configuration
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.
Incredible PBX Firewall Setup for Skyetel
The Travelin’ Man 3 firewall included with all Incredible PBX platforms limits access to your server based upon whitelisted IP addresses of outside providers and users. In order to receive calls from the multiple Skyetel data centers, the following entries need to be included in the whitelist of your PBX. For new installs of Incredible PBX 13-13 for CentOS, the entries already are included. Otherwise, issue the following commands from the Linux CLI and choose the 0 option using the add-ip utility in /root:
- /root/add-ip Skyetel-NW 52.41.52.34
- /root/add-ip Skyetel-SW 52.8.201.128
- /root/add-ip Skyetel-NE 52.60.138.31
- /root/add-ip Skyetel-SE 50.17.48.216
- /root/add-ip Skyetel-EU 35.156.192.164
NOTE: If your PBX is sitting behind a NAT-based router, then you will also need to forward UDP port 5060 from your router to the internal IP address of your PBX. Otherwise, incoming calls from Skyetel will fail. You also may need to add a NAT=yes entry to each of the Skyetel trunk configurations using the GUI. The telltale sign that the NAT entry is required will be incoming calls with one-way or no audio.
Incredible PBX Trunk Setups for Skyetel
Because Skyetel uses multiple data centers without trunk registrations, you’ll actually need to configure 6 separate Skyetel trunks in the Incredible PBX GUI. The same setup applies for those using generic FreePBX aggregations. We’ve created a script to create all of the trunks for you. Just issue the following commands. The last command assures that you don’t accidentally run the script a second time which would cause all sorts of issues. Feel free to review the code if you want to learn how to create trunks in FreePBX from the command line.
cd /root wget http://incrediblepbx.com/add-skyetel chmod +x add-skyetel # uncomment next line if your incoming calls all have 10-digit numbers # sed -i 's|from-trunk|from-pstn-e164-us|' add-skyetel ./add-skyetel chmod -x add-skyetel
Incredible PBX Inbound Routing for Skyetel
Next we need to tell your PBX how to route incoming calls from Skyetel. Using a browser, log into the IP address of your PBX using your admin credentials. Because there is no trunk registration with Skyetel trunks, you will need to create an Inbound Route for every Skyetel DID. You cannot rely upon a Default inbound route 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.
Incredible PBX Outbound Routing to Skyetel
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.
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 a 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 in 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"
SMS and MMS Messaging with Postcards
Skyetel now has released a terrific, open source Docker app, Postcards, that lets you build an SMS and MMS messaging platform for your entire organization. Suffice it to say, anything you ever wanted to do with SMS and MMS messaging, you can do with Postcards. We won’t repeat Skyetel’s excellent tutorial, but you certainly need to visit their site and take Postcards for a spin.
NEW: Skyetel Support for Dynamic IP Addresses
You asked for it, and Skyetel has delivered. For Nerd Vittles users running servers with dynamic IP addresses, Skyetel now provides support for your platform. Log into your server as root and cd /usr/src
. Then review this tutorial which describes the steps to put the pieces in place. Be advised that this is beta software at this juncture. If you run into issues, please post your questions on the PIAF Forum. Here are the actual steps:
(1) Log in to your Skyetel portal and Add a New Endpoint Group for your server giving it the name and current public IP address of your server.
(2) While still logged in, tap the Gear icon to open Settings dialog and choose API Keys tab.
(3) Add a new API key and write down your new SID and SID password.
(4) If your server is behind a router or firewall, log into that device and map UDP 5060 and UDP 10000-20000 to the private LAN address of your server.
NOTE: If your server is on the Debian, Ubuntu, or Raspbian platform, substitute the following command for the first two yum commands in step #5 below:
apt-get -y install coreutils curl git jq
(5) Log into your server and issue the following commands to install the EndPoint Updater:
yum -y install coreutils curl git epel-release yum -y --enablerepo=epel install jq cd /usr/src git clone https://bitbucket.org/skyetel/ip-endpoint-group-update.git cd ip-endpoint-group-update ./ip-update-endpointgroup.sh
(6) Fill in your credentials when prompted, and the cron script will be installed to keep your server’s dynamic IP address registered with Skyetel.
Introducing Skyetel’s New Fax Platform
Every time we read an article predicting the demise of fax technology, we have to chuckle. We’ve been reading the articles for about 30 years now, and fax still is the goto solution for many organizations. Can you spell HIPPA? Finally, Skyetel has dipped its toes in the fax waters by offering an easy-to-use fax solution for receipt of traditional and T.38 faxes. Simply purchase a Skyetel DID and configure it for vFax routing. Enter an email address for delivery of the faxes, and you’re done.
Sending faxes from the Skyetel portal still is on the drawing boards, but it’s coming. In the meantime, Incredible Fax™ which is bundled with all Incredible PBX® platforms will let you send faxes ’til the cows come home with our easy-to-use Hylafax/AvantFax implementation.
Implementing the New Spam Call Filter
One of the most often requested features for any PBX is spam call filtering. Skyetel takes it to the next level by dealing with the spammers before the calls ever reach your PBX. For each of your Skyetel phone numbers, click on the Features tab and set the Spam Call Filter as desired.
Recording and Transcribing Skyetel Calls
As with spam call filtering, recording and/or transcribing Skyetel calls is only a click away. For each of your Skyetel phone numbers, click on the Features tab and set the option desired for Recording and/or Transcribing calls. Recordings and Transcriptions can be managed from your Skyetel Dashboard. Storage is free for up to 30 days, after which they are deleted.
Skyetel Monitoring of Endpoint Health
In addition to monitoring and reporting the health of all Skyetel services in your web portal, this latest addition allows you to configure Skyetel to not only monitor the State of every registered endpoint but also its Health with realtime metrics of the Latency, Packet Loss, and Jitter of each of your endpoints. Simply check the Network QOS options desired.
Skyetel Expansion for Canadian Users
Here’s some great news for our Canadian friends. Skyetel has been listening!
- Porting to Skyetel in Canada now is significantly easier and faster
- Awesome reductions in audio round trip times
- Epic reductions in time-to-deliver
- Faster response times to technical issues (and fewer of them!)
- Audio for Canadian calls will now originate from Canadian data centers
- SMS and MMS available on Canadian ported numbers
Originally published: Thursday, November 1, 2018 Updated: Wednesday, June 12, 2019
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. [↩]
A Sobering Look at Asterisk and the 2019 VoIP Landscape
Every six months or so we like to gaze into our crystal ball for a quick look at the VoIP landscape. 2018 has been quite the transformative year with the acquisition of Digium® and Asterisk® by Sangoma®. Unfortunately, as we predicted, the Digium layoffs have already begun, and 2019 may only get worse. While we have no inside information, we wouldn’t be surprised to see Digium’s headquarters in Huntsville closed within six months in an effort to balance the books. Part of the problem may be attributable to the terms of the purchase itself. However, we sense there’s a more troubling development. And that is the reality that VoIP is becoming less and less appealing to home users and small businesses as more and more folks migrate purely to cell phones. Those with teenagers already know this transformation is underway. With services such as Google Fi starting at $20 for unlimited calling and texting, it’s difficult to justify VoIP services even at bargain basement prices. Making the cellular switch even more appealing are offers such as a $400 credit with the purchase of an LG G7 smartphone from Google or a free LG G7 with new Sprint service.
What you lose with a pure cellular platform are many of the features that have made PBXs popular in the VoIP space: call routing, text-to-speech and voice recognition applications, conferencing, SPAM call blocking, and much more. But 2018 also was the year that Google finally pulled the plug on free calling through your PBX. Instead, you now have to purchase and configure a $50 OBi200 to continue with Google Voice, and the integration is painful to put it charitably. The demise of Google Voice added one more nail to the free VoIP coffin. And, as many of you know, Vitelity, our long-time platinum sponsor, now has bowed out of the VoIP retail business due to a change in focus from Voyant, the company’s new owner. Finally, our bargain-basement cloud provider for experimentation, HiFormance, appears to have bitten the dust. Details here. Suggestions here. Reminder: "You get what you pay for."
It’s not all bad news for 2019. First, all of the Incredible PBX platforms are still alive and well. And they will remain open source GPL code. Second, we’ve found a terrific new VoIP provider, Skyetel, that will give you a $50 credit so you can kick the tires for a good long while. Effective 10/1/2023, $25/month minimum spend required. Third, if you’re looking for a robust Cloud platform, Digital Ocean still is offering a $100 signup credit for your first 60 days of service, and Incredible PBX runs swimmingly on their $5/month platform with CentOS. Spend another $1 a month, and you get automatic backups of your cloud-based server. It’s cheap insurance for something as important as your phone system.
If you’re like us, you may be getting a little nervous about the future of Asterisk. We’ve already provided a series of articles on FusionPBX for FreeSWITCH. Our original tutorial and the follow-on articles showing how to create voice prompts using IBM Watson and how to create and deploy TTS applications such as news and weather reports are worth a careful read. And, if you consider yourself a pioneer, then you owe it to yourself to try out the FreeSWITCH developers’ new cloud-based platform, SignalWire. Here’s the $55 Promo code that worked for us: ITEXPO2019. That should get you off to a great start. And check out the pricing: U.S. DIDs are $0.08 per month, U.S. Origination rate (incoming) is $0.00325 per minute, U.S. Termination rate (outgoing) is $0.0072 per minute, U.S. SMS Outbound is $0.0009 per message, and U.S. SMS Inbound messages are free. MMS also available. Once verified, you can spoof any CallerID name and number that you own! What’s not to like? Asterisk Trunk setup example available here.
CAUTIONARY NOTE: SignalWire should be considered EXPERIMENTAL SOFTWARE and is not yet suitable for production use.
That should be enough excitement to keep all of you entertained over the holidays. We’re planning a few days off to be with family and friends. Let us be the first to wish each of you a very Merry Christmas. We’re looking forward to an exciting 2019!
Originally published: Monday, December 17, 2018
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.
Skyetel Smorgasborg: SMS Blasting, SMS Dictator, and more
Just in time for Santa, we’ve got a great treat for those of you that have taken advantage of the Nerd Vittles special offer from Skyetel which gets you a $50 credit on their powerful VoIP platform. Today we’re adding not one, but three, SMS messaging utilities to the Incredible PBX UC platform. Effective 10/1/2023, $25/month minimum spend required. In addition to a command line utility to send SMS messages, we’re also introducing SMS Message Blasting which lets you send an SMS message to as many recipients as you would like. It’s perfect for sports team and community group messaging. To round out the trifecta, we’ve updated our SMS Dictator utility by integrating Skyetel messaging with IBM’s powerful voice recognition software.1 Simply dial S-M-S (767) from any extension on your PBX and dictate an SMS message to send to a recipient of your choice. Gone are the days of wrestling with Google’s ever-changing voice recognition platform. Good riddance!
To get started, you’ll need to have an IBM Watson account with an APIkey for their Speech-to-Text (STT) engine. Next, you will need a Skyetel SMS-enabled DID. Before we install today’s SMS scripts, it should be noted that SMS messages must be sent from the PBX registered as the Skyetel Endpoint Group for the SMS-enabled DID specified in the Skyetel SMS scripts. So let’s begin with the configuration steps to put all the pieces in place.
Getting Started with IBM Watson STT Service
We’ve created a separate tutorial to walk you through obtaining and configuring your IBM Watson credentials. Start there.
Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the (upper left) Menu icon and select Dashboard. 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 or LITE 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 APIkey. Then logout of IBM Watson.
Getting Started with Skyetel Messaging
If you haven’t already signed up for a Skyetel account, read our tutorial and take advantage of the $50 coupon for free service. Sign up for a DID and activate the SMS feature for your number. Create an Endpoint Group with the public IP address of your PBX. Then edit your phone number and link it to the Endpoint Group of your server. If you want to forward incoming SMS messages to either an email address or to your smartphone’s messaging service, configure it under the SMS & MMS tab. Finally, click on the settings icon beside your account name in the upper right corner of the Skyetel portal and then click the API Keys tab. Click the Create button and copy down your SID and SECRET for Skyetel’s API service. This secret is not retrievable once you close the window so put the credentials in a safe place for subsequent use. Then logout of the Skyetel portal.
Installing the SMS Components on Your PBX
There are three separate applications which we will install on your PBX: (1) a stand-alone utility that lets you send SMS messages from the Linux CLI by entering a recipients 11-digit phone number and an SMS message surrounded by quotes, (2) an SMS message blasting utility that lets you send a previously prepared SMS message to a group of recipients whose 11-digit SMS numbers have been entered into a text file, and (3) the SMS Dictator application which lets you pick up any phone on your PBX and dial S-M-S (767) to dictate a message and send it to a recipient whose number you’ve key in from your phone. For those not residing in North America, the number of phone number digits can easily be changed in all of the scripts. After we install the three applications, we’ll edit each of the scripts to insert your IBM STT and Skyetel API credentials. Then you’re ready to start messaging.
First, let’s install the stand-alone and message blasting SMS utilities. Log into your server as root and issue the following commands:
cd /root mkdir sms-skyetel cd sms-skyetel wget http://incrediblepbx.com/smsblast-skyetel.tgz tar zxvf smsblast-skyetel.tgz rm -f smsblast-skyetel.tgz
Next, let’s install the SMS Dictator application while still logged into your server:
cd /var/lib/asterisk/agi-bin wget http://incrediblepbx.com/sms-767-skyetel.tgz tar zxvf sms-767-skyetel.tgz rm -f sms-767-skyetel.tgz ./install-sms767-dialplan.sh
Configuring the Skyetel SMS Components
While still positioned in the agi-bin directory, edit smsgen.sh. Insert apikey as your API_USERNAME and your actual STT APIkey as API_PASSWORD in the fields provided. Insert your Skyetel SID, SECRET, and 11-digit DID in the fields provided. Then save the file.
Next, change directories to /root/sms-skyetel and edit BOTH sms-skyetel and smsblast and insert your Skyetel credentials and DID in the fields provided at the top of both files.
Finally, when you’re ready to use the message blasting application (smsblast), first insert your SMS message in the smsmsg.txt file. Then insert the list of SMS numbers in smslist.txt.
Testing the Skyetel SMS Components
To try out the SMS Dictator application, dial S-M-S (767) from a phone connected to your PBX. When prompted, enter the 11-digit number of the SMS recipient. When prompted, dictate the message to be sent and press #.
To try out the stand-alone SMS application, navigate to /root/sms-skyetel and issue the following command using the 11-digit number of the SMS recipient followed by a space and an SMS message to be sent surrounded by quotes: ./sms-skyetel 18005551212 "Howdy."
To try out the message blasting SMS application, navigate to /root/sms-skyetel. Enter the message to be sent in smsmsg.txt and enter the list of SMS numbers in smslist.txt. Kick off the message blast by entering the command: ./smsblast
.
Originally published: Monday, December 10, 2018
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? Join our new MeWe Support Site.
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.
- Skyetel outbound SMS messages are billed at 1¢/message plus a monthly SMS surcharge of 10¢ per SMS-enabled DID. With IBM’s STT service, users have a choice of the LITE tier providing 100 minutes a month of free transcription or the STANDARD tier providing unlimited message transcription at a cost of 2¢/minute. [↩]
Spam Phone Call Blocker and CNAM Caching for FreePBX
Blocking spam phone calls has been a challenge to put it charitably. Thanks to some earlier work by Stewart Nelson on the DSLR forum as well as Stewart’s considerable hand-holding in the development of today’s tutorial, we want to introduce a new approach to blocking these calls. The way it works is first time callers that pass the TrueCNAM SPAM check will be prompted to "press 5 to connect." Since most spam calls sit in a queue for several seconds before a live person chimes in, that person won’t hear the prompt. After 10 seconds or an invalid response, a SIT tone is played and the call is disconnected. If you’d prefer, you can send the failed calls to voicemail by uncommenting a single line in your dialplan. When a successful caller calls again, the caller will be connected without encountering the press 5 prompt.1 While today’s approach won’t block every robocaller, our testing suggests that, in combination with TrueCNAM, it will catch more than 95% of the spam callers. Using CallerID Superfecta with CNAM lookups from OpenCNAM coupled with AsteriDex and the Asterisk® Phonebook will provide an extremely low-cost solution both for blocking spammers AND for displaying accurate CNAM data for incoming calls since you’ll only pay for CNAM and TrueCNAM lookups from legitimate callers once.
Here’s the actual dialplan addition that will monitor your incoming calls:
[sub-log-caller] exten => s,1,NoOp(*** begin sub-log-caller ***) exten => s,n,GotoIf(${DB_EXISTS(cidname/${CALLERID(num)})}?CNAMCHECK) exten => s,n,GotoIf($[${DB_EXISTS(SPAMCHECK/deactivate)} = 0]?ACTIVATE) exten => s,n,GotoIf($[${DB(SPAMCHECK/deactivate)} = 1]?CONNECTNOW) exten => s,n(ACTIVATE),NoOp(Not yet WhiteListed) exten => s,n,AGI(truecnam.sh,${CALLERID(number)}) exten => s,n,GotoIf($["${SPAM}"="SPAM"]?FLUNKED) exten => s,n,Playback(silence/1) exten => s,n,Playback(to-call-num-press) exten => s,n,Playback(digits/5) exten => s,n,Read(MYCODE,beep,1,n,1,10) exten => s,n,GotoIf($["${MYCODE}" = "5"]?ANONTEST) exten => s,n(FLUNKED),NoOp(*** Caller FLUNKED screening ***) ;exten => s,n,Dial(local/*701@from-internal) ; uncomment to send to 701 VM exten => s,n,Zapateller() exten => s,n,Hangup exten => s,n,Return() exten => s,n(ANONTEST),GotoIf($[${CALLERID(num)} > 0]?WHITELIST:CONNECTNOW) exten => s,n(CNAMCHECK),Set(CNAM1=${CALLERID(name)}) exten => s,n,Set(CNAM2=${DB(cidname/${CALLERID(number)})}) exten => s,n,GotoIf($["${CNAM1}" = "${CNAM2}"]?WHITELISTED exten => s,n(WHITELIST),Set(DB(cidname/${CALLERID(number)})=${CALLERID(name)}) exten => s,n,Set(CALLERID(all)="${CALLERID(name)} < ${CALLERID(number)}>") exten => s,n(WHITELISTED),NoOp(WhiteListed: ${CALLERID(all)}) exten => s,n(CONNECTNOW),NoOp(*** end of sub-log-caller ***) exten => s,n,Return()
We first introduced some of the CallerID caching concepts in our previous article last May. That article also documented the procedure for adding inbound call processing logic into FreePBX. If you already have implemented the steps outlined in that article, then the only modification required to deploy today’s new spam blocking technique is to replace the [sub-log-caller] context and reload the Asterisk dialplan. NOTE: Some deployments of CallerID Superfecta have an incorrect database password in the Default setup for AsteriDex. The original article will walk you through making the necessary change.
If you’re starting from scratch, stop here for a bit and follow all of the steps in our previous article which now incorporates the spam blocking code as well. Here’s the link to get started. Return here once you’ve completed the initial setup.
If you’re updating a previous deployment, here are the steps. Edit extensions_custom.conf in /etc/asterisk and remove the [sub-log-caller] context toward the end of the file. Then save the file. Next, issue the following commands to move the TrueCNAM script into place and insert the updated [sub-log-caller] context as well as the new [macro-dialout-trunk-predial-hook] context. Then reload your Asterisk dialplan. The dialplan additions will populate the Asterisk Phonebook and also whitelist calls from your PBX as well as incoming calls making it through the Spam Blocker.
cd /tmp wget http://incrediblepbx.com/sub-log-caller.tar.gz tar zxvf sub-log-caller.tar.gz rm -f sub-log-caller.tar.gz mv truecnam.sh /var/lib/asterisk/agi-bin cd /etc/asterisk cat /tmp/sub-log-caller.txt >> extensions_custom.conf asterisk -rx "dialplan reload"
Rotary Dial Phones & Blocked Numbers
If someone you know and love still has a rotary dial phone, then you will need to manually add their number to either AsteriDex or your Asterisk Phonebook. Otherwise, the calls will never make it through the Spam Catcher. You can do this within the FreePBX GUI by accessing Admin -> Asterisk Phonebook. Click + Add Phonebook Entry and enter the 10-digit number for Grandma as well as her name. Add a second entry with Grandma’s 11-digit number in case some of your VoIP providers happen to send 11-digit CallerID numbers. We hasten to add you should normalize the formatting of your CallerID numbers as quickly as you can to avoid double entries. For those in the U.S. and Canada, we recommend the from-pstn-e164-us context for all of your trunks.
If you have lots of friends with rotary dial phones or if you get calls from important, but unknown numbers such as medical offices where Caller ID numbers are blocked, then you probably should consider uncommenting the voicemail option in [sub-log-caller]. Then you at least will get voicemail notifications when one of these callers attempts to contact you. You still will have to manually add them to AsteriDex or the Asterisk Phonebook so they can contact you directly in the future. HINT: Most medical office calls now spoof the main number of the office so you only need to add the office number just as you did with grandma.
Toggling Spam Blocker On and Off
We’ve also included the ability to turn off the Spam Blocker should you ever wish to do so. To disable the Spam Blocker, issue the following command at the Asterisk CLI:
database put SPAMCHECK deactivate 1
To once again enable the Spam Blocker, issue the following command at the Asterisk CLI:
database deltree SPAMCHECK
WhiteListing Previous Callers
We appreciate that you may not want to aggravate callers that have been calling you for years by making them jump through hoops the next time they call. So here’s a quick way to populate your Asterisk Phonebook with the names and numbers of previous callers. For entries where the CNAM is merely the CallerID Number, future calls from these numbers still will be looked up with OpenCNAM to obtain an actual CNAM match. We’ve made a couple of assumptions that you are more than welcome to adjust to meet your own needs. First, we’ve limited the list to callers from the past two calendar years. Second, we’ve only captured calls that lasted more than 15 seconds. We’ll drop down to the Linux CLI to build the list of callers to import. Then we’ll use the FreePBX GUI to import the list into the Asterisk Phonebook. While we’re building the import list, you’ll have two opportunities to prune the list using your favorite text editor. To get started, issue the following commands from the Linux CLI:
mysql -u root -ppassw0rd asteriskcdrdb -Ns -e "select distinct src, clid \\ from cdr where calldate > '2017/01/01' and duration > 15 \\ order by clid asc" > 2YR-full
Now edit the 2YR-full file and remove any complete lines you don’t want to import.
Next, we’ll reformat the CallerID Numbers and Names into a format needed for the import:
cat 2YR-full | cut -f 1 -d '"' | sed 's|[[:space:]]||' > 2YR-numbers cat 2YR-full | cut -f 2 -d '"' > 2YR-names paste 2YR-numbers 2YR-names | awk '{print $1,$2,$3,$4}' > 2YR-all awk '{print $2 " " $3 $4 ";" $1";"}' 2YR-all > 2YR-freepbx.csv
Now we should have our 2YR=freepbx.csv file in its final form for import. Open the file in your favorite editor. The syntax of the entries should be CallerID Name, then a semicolon, then CallerID Number, and then a semicolon. Discard any additional lines you wish to exclude from the import. Once you have all the entries squared away, copy the file to your desktop PC and open FreePBX in your browser. Navigate to Admin -> Asterisk Phonebook. Click Import Phonebook and then Browse. Select the 2YR-freepbx.csv file from your desktop. Then click Upload. Take a final look at the new entries in your Asterisk Phonebook to make sure nothing came unglued, and you’re all set.
TrueCNAM: The Icing on the Spam Cake
A couple years ago we introduced TrueCNAM, a service that provides not only CNAM data but also Caller Reputation scoring. Those that flunk using the revolving caller reputation matrix get disconnected automatically. We strongly encourage you to add the TrueCNAM service to your PBX. The service includes a free tier as well as incredibly reasonable commercial tiers. For background on the service, here’s a link to our previous TrueCNAM tutorial. For today, start by signing up for a TrueCNAM account and obtain an APIkey and APIpassword. Then register at least one of your DIDs with the service. Once you have your credentials and your DID number in hand, edit truecnam.sh in /var/lib/asterisk/agi-bin. Insert these three items at the top of the file and save it to activate TrueCNAM. It doesn’t get much easier than that.
Now make a few test calls to your PBX to assure that everything is working as documented. Enjoy!
Originally published: Monday, November 26, 2018
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? Join our new MeWe Support Site.
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.
- Once installed, you can change the voice prompt to a number other than 5 by modifying lines 10 and 12 of the context sub-log-caller which you will find in extensions_custom.conf in the /etc/asterisk directory at the completion of this install. [↩]
Free Asterisk Voicemail Transcription with IBM Watson STT
There are many commercial voicemail transcription services for Asterisk® PBXs, but none hold a candle to the speech-to-text (STT) quality of the IBM Cloud offering known as Watson® STT, formerly known as Bluemix TTS. Despite a recent price increase that takes effect in December, the pricing remains competitive. On the Standard Pricing Plan, voicemail transcription is 2¢ per minute. Or you can try things out on the LITE plan which offers 100 minutes a month at no cost. When the messages are delivered by email, you get the voicemail recording in MP3 format AND transcribed text courtesy of Watson TTS. With IBM services, there no longer are username:password credentials. Instead, you will have a new apikey.
Those with existing configurations can update your credentials by inserting a new apikey using the following commands, or you can simply insert apikey as your $API_USERNAME and enter your actual APIkey as your $API_PASSWORD.
cd /usr/local/sbin sed -i 's|$API_USERNAME:$API_PASSWORD|"apikey:x-yy-zzz"|' sendmailmp3 sed -i 's|$API_USERNAME:$API_PASSWORD|"apikey:x-yy-zzz"|' bluemix-test
IBM Cloud’s STT solution is a real game-changer for one simple reason. Their STT API performs more accurately than any speech recognition engine in the world. As an added bonus, you won’t have to worry about Google breaking our middleware every month. It’s worth noting that IBM doesn’t round up minutes. Transcribing two 30-second messages counts as one minute.
https://youtu.be/JWnLgZ58zsw
Overview. What we’ve done today is integrate the Watson STT API directly into existing Asterisk voicemail systems. We started with Nicolas Bernaerts’ terrific sendmailmp3 script. It works on both the Wazo and FreePBX® platforms. If you have deployed Incredible PBX®, then the setup takes a couple of minutes. For everyone else, there’s an additional configuration step using your favorite GUI. To get started, you’ll sign up for an IBM Cloud account and obtain your credentials. Next, you download today’s script for your platform and insert your credentials. Finally, you set up voicemail on the extensions desired and insert an email address for each voicemail account. On generic FreePBX systems, you’ll need to add the name of our script to manage your voicemail recordings. And, regardless of your PBX platform, you obviously need outgoing SMTP email working reliably.
Start by sending yourself a test email and get that working first:
echo "test" | mail -s testmessage yourname@your-email-domain.com
What About the Quality? Here’s the bottom line. Speech recognition isn’t all that useful if it fails miserably in recognizing everyday speech. The good news is that IBM Watson’s speech recognition engine is now the best in the business. If you want more details, read the article below which will walk you through IBM’s latest speech recognition breakthrough:
Why IBM's speech recognition breakthrough matters for AI and IoT. Via @techrepublic https://t.co/AJi8MA3E20
— IBM Developer (@IBMDeveloper) March 15, 2017
Obtaining IBM Cloud Speech to Text Credentials
Follow this link to set up your IBM account and obtain credentials for both Speech to Text (STT) and Text to Speech (TTS) services. Please note that your STT and TTS API keys will NOT be the same. So don’t accidentally use the wrong one.
Installing STT Engine with Incredible PBX for Wazo
1. After logging into your Incredible PBX for Wazo server as root using SSH/Putty:
cd /usr/sbin wget http://incrediblepbx.com/sendmailibm.tar.gz tar zxvf sendmailibm.tar.gz rm -f sendmailibm.tar.gz
2. Edit sendmailibm and insert IBM STT API_KEY and URL.
3. Edit bluemix-test and insert IBM STT API_KEY and URL.
4. Apply the patch documented above if using LITE plan using sendmail filename instead of sendmailmp3.
5. Copy the updated sendmailibm file to sendmail:
cd /usr/sbin cp -p sendmailibm sendmail
6. Test your Bluemix STT setup: bluemix-test
7. Result should be: please record your message after the beep
8. Set up voicemail account for a Wazo extension with your email address.
9. Place a test call to the extension and record a voicemail when prompted.
10. Your message will be transcribed and delivered via email.
Installing STT Engine with Incredible PBX for RasPi
1. After logging into your Raspberry Pi server as root using SSH/Putty:
cd /usr/sbin wget http://incrediblepbx.com/sendmailibm-raspi.tar.gz tar zxvf sendmailibm-raspi.tar.gz rm -f sendmailibm-raspi.tar.gz
2. Edit sendmailmp3.ibm and insert your Bluemix STT API_KEY and URL. Save file.
3. Edit bluemix-test and insert your Bluemix STT API_KEY and URL. Save the file.
4. Copy the updated sendmailmp3.ibm file to sendmailmp3:
cd /usr/sbin cp -p sendmailmp3.ibm sendmailmp3
5. Apply the patch documented above if using LITE plan.
6. Test your Bluemix STT setup: bluemix-test
7. Result should be: your dictation is now being processed and emailed please wait
8. Set up voicemail for a RasPi extension with your email address.
9. Place a test call to the extension and record a voicemail when prompted.
10. Your message will be transcribed and delivered via email.
Installing STT Engine with Incredible PBX 13-13
1. After logging into your Incredible PBX 13 server as root using SSH/Putty:
cd /usr/local/sbin wget http://incrediblepbx.com/sendmailibm-13.tar.gz tar zxvf sendmailibm-13.tar.gz rm -f sendmailibm-13.tar.gz
2. Edit sendmailmp3.ibm and insert your IBM STT API_KEY and URL. Save file.
3. Edit bluemix-test and insert your IBM STT API_KEY and URL. Save the file.
4. Copy the updated sendmailmp3.ibm file to sendmailmp3:
cd /usr/local/sbin cp -p sendmailmp3.ibm sendmailmp3
5. Test your Bluemix STT setup: bluemix-test
6. Result should be: we are now transferring you out of the company directory…
7. Set up voicemail for an extension and include your email address.
8. Place a test call to the extension and record a voicemail when prompted.
9. Your message will be transcribed and delivered via email.
Installing STT Engine with VitalPBX
For those using VitalPBX with or without Incredible PBX, we’ve written a new tutorial to walk you through the procedure to get voicemail transcription with IBM Watson STT up and running. Here’s the link.
Installing STT Engine with Legacy FreePBX Servers
1. Follow steps #1 through #8 from the Incredible PBX 13 tutorial above.
2. Choose Settings -> Voicemail Admin -> Settings in the GUI.
3. In the format field, insert: wav|wav49
4. In the mailcmd field, insert: /usr/local/sbin/sendmailmp3
5. Click Submit to save your settings and then Reload the FreePBX Dialplan.
6. Place a test call to the extension and record a voicemail when prompted.
7. Your message will be transcribed and delivered via email.
Update: Matt Darnell reports that, depending upon your existing setup, you may need to add the unix2dos and lame packages with legacy FreePBX servers to get MP3 messages delivered correctly.
Originally published: Monday, March 12, 2018 Updated: Monday, November 12, 2018
Got Friends? 7 Countries Have Never Visited Nerd Vittles. 2018 Is Calling! https://t.co/wMfmlhAr16 #asterisk #freepbx #wazo #issabel #IncrediblePBX #3CX pic.twitter.com/kAmAEnwVIw
— Ward Mundy (@NerdUno) January 9, 2018
Need help with Asterisk? Visit the PBX in a Flash 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.
Some Recent Nerd Vittles Articles of Interest…
Double-NAT Blues: Tackling Asterisk’s Thorniest Problems
Whether you’re new to VoIP technology or an Old Timer, nothing is quite as frustrating as wrestling with one-way audio and no audio on SIP calls either because of poorly designed NAT-based routers or poorly implemented SIP ALG solutions on low-end residential routers. To make matters worse, you get to deal with calls originating behind not one, but two, NAT-based routers neither of which complies with the basic SIP Rules of the Road. In a perfect world, SIP and RTP packets arriving from the Internet would have their public IP address translated into a private LAN address upon arrival at the NAT-based router. And the departing packets would have their private IP addresses translated into the public IP address of the router when leaving. If your PBX and SIP phone happen to be behind different NAT-based routers and hardware from the likes of Comcast, Spectrum, and AT&T, the odds of SIP calls working reliably are somewhere between slim and none. Perhaps it’s no coincidence that each of these providers also happens to offer competing (expensive) telephony service.
Today we’d like to offer some Asterisk® solutions that resolve these issues. First, if you are the subscriber to cable or DSL Internet service, you may have some success by talking to your provider and persuading them to set up their hardware in bridged mode so that you can install your own NAT-based router that properly handles SIP traffic. Second, it’s almost always a good idea to disable SIP ALG service on routers that you control. The reason is because of the poor ALG implementations on almost all low-cost routers. Third, configuring the Public and Private IP NAT Settings for your PBX using the FreePBX® GUI (Settings->Asterisk SIP Settings->NAT Settings) often solves the problems. Fourth, make sure NAT=yes is set in your extension and trunk settings.
If you happen to be traveling and have no control over the network architecture, the chances of the above recommendations resolving your SIP problems are not likely. This includes offerings in hotels, rental units, cruise ships, and WiFi HotSpots worldwide. In most of these locations, you would want to use a SIP phone to connect back to your home or office PBX so that you could receive incoming calls and place outbound calls just as if you were sitting at your desk at home. In these situations, we have a failsafe solution for you, but it requires a little advance planning because you need to configure your home or office Asterisk server to support the design.
The easiest way to eliminate NAT problems is to take NAT out of the equation when making and receiving SIP calls. With Asterisk, this is easy. What we typically do is interconnect the home or office Asterisk PBX with a local Asterisk PBX using an IAX2 trunk. Thus, no SIP traffic passes between your local PBX and your home or office PBX regardless of the number of layers of routers that are present between the two servers. If you can make SIP calls through a provider while sitting at home, you have solved the SIP connectivity issues at the home/office end. If your local PBX and SIP phone or softphone are on the same local LAN whether wired or wireless, then there is no SIP connectivity issue locally either. So how?
Rule #1: Always travel with a notebook computer that includes VirtualBox and a reliable SIP softphone. We’re big fans of all of the Mac notebooks, any of them will suffice. Windows and Linux notebooks work as well. Steer clear of Chromebooks which lack a crucial Linux kernel driver required by VirtualBox. There’s a solution, but it’s painful. On the Mac platform, you can’t beat the free Telephone app for your SIP phone.
Rule #2: Set up a NeoRouter VPN to provide secure interconnectivity between your home or office PBX and your local PBX. With Incredible PBX platforms, the NeoRouter client is included. You’ll just need to install the NeoRouter server component on some server with a public IP address. Complete details are here. To obtain a NeoRouter private IP address on each PBX, run this command after logging in as root: nrclientcmd
.
Configuring IAX Trunk on Home/Office Server. You’ll need the NeoRouter IP address and a secure password to set up the trunk that will interconnect your Home-PBX with your local PBX. We’re going to refer to the two servers as Home-PBX (10.0.0.1) and Travel-PBX (10.0.0.2) to keep things simple. On the Home-PBX, create an IAX trunk using the FreePBX GUI with a Trunk Name of Travel-PBX. The PEER Details should look like the following using a very secure password that will be used on the trunk at the other end as well:
type=friend secret=very-secure-password host=dynamic context=from-internal requirecalltoken=no deny=0.0.0.0/0.0.0.0 permit=0.0.0.0/0.0.0.0
The Registration String would look like the following where very-secure-password is your actual shared secret for the two trunks and 10.0.0.2 is the actual VirtualBox IP address of the Travel-PBX: Home-PBX:very-secure-password@10.0.0.2
Configuring IAX Trunk on Travel-PBX Server. You’ll need the NeoRouter IP address and a secure password to set up the trunk that will interconnect your Travel-PBX server with your Home-PBX. On the Travel-PBX, create an IAX trunk using the FreePBX GUI with a Trunk Name of Home-PBX. The PEER Details should look like the following using a very secure password that will be used on the trunk at the other end as well:
type=friend secret=very-secure-password host=dynamic context=from-internal requirecalltoken=no deny=0.0.0.0/0.0.0.0 permit=0.0.0.0/0.0.0.0
The Registration String would look like the following where very-secure-password is your actual shared secret for the two trunks and 10.0.0.1 is the actual VirtualBox IP address of the Home-PBX: Travel-PBX:very-secure-password@10.0.0.1
Once you get this far, log into both servers as root and start up the Asterisk CLI. On each server, issue the following command to be sure the two trunks are registered with each other: iax2 show registry
Routing Calls from Home-PBX to Travel-PBX. What follows is one scenario for call routing. We’re assuming calls to your Home-PBX are routed to a Ring Group consisting of various extensions in your home or office. We’re also assuming you want to now add an extension on Travel-PBX to that Ring Group so that incoming calls to your Home-PBX will also ring the softphone connected to an extension on your Travel-PBX. In the Asterisk/FreePBX world, we accomplish this by adding an Outbound Route for the Travel-PBX extension and then adding this number to the Ring Group with a # prefix to tell FreePBX that it’s a trunk call rather than a local extension. In our example, we’re assuming the softphone extension on Travel-PBX is 701, but we’re also assuming there is a different extension 701 on Home-PBX. To avoid confusing the Home-PBX, we’ll add a 7 prefix for the Travel-PBX extension and then strip it off before passing the call to Travel-PBX.
First, create an Outbound Route called Travel-PBX-Out. For the Dial Pattern, enter a Prefix of 7 and a Match Pattern of 701. For the Trunk Sequence, choose Travel-PBX. Move the Outbound Route near the top of your route list to assure that it gets processed before any other 4-digit extensions. Second, edit your Ring Group and add 7701# to the existing list.
Routing Calls from Travel-PBX to Home-PBX. On the Travel-PBX, we’re assuming you’d like calls placed from your softphone to be processed exactly as if you were calling from a local extension on Home-PBX. Create an Outbound Route called Home-PBX-Out. For the Dial Patterns, add one for 10-digit calls: NXXNXXXXXX. If you want to be able to reach 3-digit extensions on Home-PBX, add a second dial pattern with a 9 prefix and XXX for the Match Pattern so it doesn’t conflict with local extensions. For Trunk Sequence, choose Home-PBX.
Originally published: Monday, August 20, 2018
Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.
Need help with Asterisk? Visit the PBX in a Flash Forum.
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.
Some Recent Nerd Vittles Articles of Interest…