Choosing Wisely: Mastering Asterisk IVR and AutoAttendant Design with XiVO




Today we want to talk a little about design choices and IVRs. First and foremost, we don’t want to leave anyone behind during our XiVO adventure. XiVO is a platform that adjectives really can’t describe. It’s that good and, frankly, we’re having a hard time believing it’s been around for almost a decade and nobody much talked about it. Leave it to the crazy Americans to only look at stuff from the U.S. of A. Funny thing is that the two major GUIs for Asterisk® now are both Canadian-based.


One of our PIAF Forum readers posted a comment last week that said:

The only downside I see is that XiVO does not have [a] GUI for building IVRs. To build [a] complex, nested IVR system, everything has to be thought about in great detail writing contexts and dial plans to suit your unique requirements. It would be nice if XiVO offered a GUI for building IVRs.

This raises some issues about GUI design and development that are worth addressing. As with any GUI, the development cycle is lengthy and incredibly complex. This is especially true with XiVO where new versions are released every two weeks! In our second XiVO article, we showed how easy the upgrade procedure was. Those coming from other Asterisk platforms will appreciate this little shocker. XiVO doesn’t break stuff with their upgrades. Frankly, the only other company I can say that about is SONOS. If you don’t have their music platform, you’re missing a treat.

Introducing new components into any “main product” can cause all sorts of problems with the pieces that used to work. If you don’t believe it, look at some of the “other forums” and look at the number of message threads complaining that the new X Widget broke the Y widget and now nothing works. While we can’t speak for everyone, I think it’s safe to say nobody that depends upon their phone system wants to see it go up in flames regularly because some developer had a great new idea that didn’t quite do what it was supposed to do.

To their credit, the XiVO developers were smarter than that. They’ve not only built a mighty mousetrap, but they’ve done it in a way that supports outside integration of additional components without breaking the main product. There are numerous “hooks” that allow anyone with any skill set to add missing pieces. Some of these hooks are exclusively for programmers, but many were designed to let anybody integrate almost anything into the XiVO platform.

So, when a user says “I wish XiVO had an IVR Builder in the GUI,” our first inclination was to chuckle and respond with “You just don’t appreciate how lucky you are not to have an IVR Builder in the GUI.” What the commenter didn’t appreciate is that you don’t need to pre-build components with XiVO before developing an IVR. With the “other” GUI, you first had to create Custom Destinations and Custom Contexts and Miscellaneous Applications and Miscellaneous Destinations and Custom Recordings in the GUI before you could take advantage of the IVR GUI to build much of anything. Think about that for a minute. Yes, there was an IVR builder but, before you could use it, you first had to transform every component to be incorporated into the IVR using a large number of subcomponents to translate all of your Asterisk pieces into the GUI’s special lingo. Think of them as GUI pigeonholes, and you had to decipher which Asterisk square pegs went in which GUI round holes. We can’t count the number of times we’ve begun the IVR creation process only to have to stop and create missing components because the IVR builder simply wouldn’t recognize a feature as being part of our Asterisk dialplan.

Building IVRs and AutoAttendants with XiVO

The anatomy of an IVR in Asterisk could not be more straight-forward. You have a prerecorded message that plays to the caller giving them choices from which to choose from a menu of selections. The caller presses one of the 12 keys on their phone, and the IVR goes off and does some task: calls an extension, plays a recording, runs an Asterisk application, makes an outside call, or kicks off another IVR with another recording and more choices. Some options in the IVR may not be mentioned, and this is commonly referred to as the Stealth AutoAttendant. None of this is rocket science.

To build an IVR, you need these components: (1) a prerecorded message, (2) a list of the choices you want to provide to the caller with the corresponding destinations on the PBX to execute those choices, and (3) a template to follow to create the IVR dialplan code in XiVO.

Trust us when we say the major problem with IVRs is not that they’re difficult to build in XiVO. The real issue with most IVRs is that the person that implemented the IVR spent all their time worrying about the mechanics of PBX implementation and didn’t put sufficient thought into the IVR layout and the caller’s experience when actually interacting with the IVR. If you haven’t heard Allison Smith speak about IVR design, put it on your Bucket List for the next AstriCon or do some reading. That’s a long-winded way of saying that filling in the blanks of an IVR template is just as easy as point-and-click or drag-and-drop except for the eye candy. Just be thankful the XiVO platform gives you the flexibility to do it yourself without having to create imaginary destination hooks and recording linkages before they can be used in the product’s IVR GUI because the developers didn’t have the foresight to think outside their own GUI’s box. Every Windows user can appreciate that problem.

For today, we’re assuming you’ve done your homework and have already sketched out the options you want to incorporate into your IVR or IVRs. No GUI can help with that! So we’ll pick up from there and show you how easy it is to incorporate your IVR design into XiVO.

1. Incorporating Your Prerecorded Messages into XiVO

For openers, you obviously need a recording to greet callers and tell them what their choices are when using your IVR or AutoAttendant. You can build these recordings yourself on the XiVO platform or, for a more professional IVR, you can send the text off to Allison Smith and let her record the voice prompts for you. Digium makes it easy. Visit their web site, type in the text, and you’ll have your recording in a couple of days. No, they’re not free, but they’re not expensive either.

Since we’re just getting started, let’s assume you want to create a recording prototype on your own to work out the kinks in your IVR first. Here’s how. We’re assuming you’ve already read the Nerd Vittles XiVO tutorial and put the Festival TTS platform in place. Next, log into your XiVO server as root. To keep things simple, let’s put the recordings in WAV format in the /var/lib/xivo/sounds/playback directory which is reserved for our custom recordings:

cd /var/lib/xivo/sounds/playback

To actually generate the sound file that Asterisk can play back, execute the command below after placing your text between the quotation marks and giving the sound file a name, e.g. ivr-number1.wav:

echo "Text goes here" | /usr/bin/text2wave -F 8000 -o ivr-number1.wav

Here’s an example:

echo "Thank you for calling. Press 1 for Tom, 2 for Dick, or 3 for Harry. Press 0 to be connected to the operator." | /usr/bin/text2wave -F 8000 -o ivr-number1.wav

2. Marrying IVR Choices to PBX Destinations

Whether you’re deploying an IVR using FreePBX® or XiVO, you still have to translate your Plain English options into code that the GUI understands so that calls get routed successfully to the intended destinations.

Let’s begin with the FreePBX Way. Our previous IVR tutorial showed how it was done:




As you can see from the above routing procedure, there were interim steps for every single option in this IVR menu except #8. What you may not appreciate is that you first had to create both a Misc Destination AND a Custom Extension before these options could be used in FreePBX. Otherwise, the options simply didn’t appear in the IVR GUI’s pull-down pick lists.




If you wished to incorporate a custom context that wasn’t assigned an extension number on your PBX, there was a different GUI procedure. For something as simple as retrieving the time of day, you had to get the custom context registered with FreePBX before the dialplan code could be used in the IVR. According to the FreePBX developers, this functionality was considered an “advanced feature and should only be used by knowledgeable users.”



Our purpose in documenting all of this is to demonstrate that building IVRs even in a GUI is much more than point-and-click. It requires mastery of some fundamental Asterisk dialplan concepts not to mention the GUI’s own labyrinth of secret pigeonholes. Once you’ve had to master all of that, we believe it’s simpler to build IVRs using simple commands rather than jumping through all of the convoluted hoops required just to make your IVR GUI platform happy.

Let’s compare this methodology to the XiVO way of doing things by way of example. Then you can decide for yourself which approach is more complex. Would you know all of these on your own? Probably not. But now you can see how simple it really is. There really are only two words you need to learn: Dial and Goto. 🙂

Call an Extension: Dial(Local/701@default)
Call a Ring Group: Dial(Local/801@default)
Call a PSTN Number: Dial(Local/8005551212@default)
Call a SIP URI: Dial(SIP/2233435945@rentpbx.mundy.org)
Access DISA with permission: Dial(Local/3472@default)
Join a Conference: Dial(Local/2663@default)
Playback Yahoo News: Dial(Local/951@default)
Playback Weather Forecast: Dial(Local/947@default)
Identify IVR Option as Invalid and Repeat Menu: Goto(i,1)
Hangup on Caller for Choosing Invalid Option: Goto(t,1)
Execute Time of Day Custom Context: Goto(new-time,s,1)
Send Caller to a Second IVR and Play Second Recording: Goto(ivr-2,s,3)

Building XiVO IVRs from an IVR Template

We can’t speak for everyone, but we’ve always told folks not to write a book about how to do something. Just give us an example that’s easy to follow and we’ll take it from there. So here you go.

In the XiVO world, IVRs are nothing more than custom contexts. They have a name in [brackets], and they’re stored in config files saved in /etc/asterisk/extensions_extra.d. A config file can include multiple contexts or only one. For IVRs, we recommend you save each one in a single configuration file that houses a single context.

We’re going to give you a template to follow in creating all of the IVRs you can dream up. All you need is a custom recording for each one and your list of choices and destinations for those choices. The examples above tell you everything you need to know to build awesome IVRs.

After downloading the template, we recommend that you not edit it directly. Make a copy with a new file name and change the context name in the template to match your new file name. We also do one other little trick with all of our custom contexts. They always begin and end with comment lines like this using the context name:

;# // BEGIN ivr-template
;# // END ivr-template

The reason for this is it makes it incredibly easy to remove the entire context with a single command:

sed -i ':// BEGIN ivr-template:,:// END ivr-template:d' ivr-template.conf

This doesn’t matter so much when you only have a single context in a single file. But it is immensely helpful when you’ve stored dozens of contexts within the same file. Some may prefer to store all of the related IVR contexts for their entire IVR tree in a single file. And then you’ll appreciate this tip when it’s time to make major changes in your IVR.

Let’s begin by putting your template in place and then cloning it to ivr-number1:

cd /etc/asterisk/extensions_extra.d
wget http://incrediblepbx.com/ivr-template.tar.gz
tar zxvf ivr-template.tar.gz
rm -f ivr-template.tar.gz
cp -p ivr-template.conf ivr-number1.conf
sed -i 's|ivr-template|ivr-number1|' ivr-number1.conf

The rest of today’s exercise can be performed in the XiVO GUI using its built-in editor. Open the GUI with your browser and navigate to Services -> iPBX -> Configuration files and then open ivr-number1.conf by clicking on the pencil icon beside it.

Anatomy of the XiVO IVR Template

First things first. Change the sound recording in line s,3 to match the recording you made above without the .wav extension: ivr-number1. Leave the directory path just as it is. So your line should now look like this:

exten => s,3(skip),Set(IVR_MSG=/var/lib/xivo/sounds/playback/ivr-number1)

Next, take a look at the structure of the file. You’ll note that there are options labeled exten => 0,1, through exten => 9,1,. These match the numeric keys on a telephone obviously. In the IVR world, it’s called a phone tree. All you need to change is what comes after the second comma on each line. This destination should be one of the XiVO commands we documented above telling XiVO how to process the call. For option 0, let’s assume you wanted to route the call to extension 701. Your 0 branch would look like this:

exten => 0,1,Dial(Local/701@default)

The remaining dial options should be obvious. If you want to designate a particular option to be invalid, make the option look like this:

exten => 9,1,Goto(i,1)

Another alternative is to remove the line entirely; however, we prefer the above approach because it makes it easy to change things down the road if you decide to use option 9 as a call destination.

Two other options warrant a brief explanation. The i option tells XiVO how to process the call if the caller chooses an invalid option. The t option tells XiVO what to do if the 3-second timeout occurs without the caller pressing a key. You can modify these to meet your own requirements. As configured, an invalid option sends the caller back to the recording to start over. And the timeout option hangs up the call.




Finally, phone trees can get quite complex. A GUI can’t fix that either. Pressing option 2 might trigger phone tree 2 while pressing 3 might trigger phone tree 3. Programmers could obviously rewrite the dialplan to handle all of these separate phone trees with their separate branches in one giant, convoluted chunk of dialplan code. But why? Just make each phone tree a separate IVR housed in its own file with its own context. And navigate between the IVRs using simple Goto commands such as Goto(ivr-number2,s,3). To return to the main IVR, do the same thing pointing to the line number to which the call should be redirected, e.g. Goto(ivr-number1,s,3). You obviously don’t need to answer each call but once so skip those lines in the IVR dialplan when choosing the line number to which to redirect processing.

Routing Incoming Calls to Your IVR

If you’ve already set up one or more DIDs on your PBX, then you probably routed those Incoming Calls to a user or ring group. Changing the routing to send the calls to your IVR is easy. Just edit the DID entry for the Incoming Calls you wish to redirect and set the Destination to Customized and the destination Command to the context of your IVR: Goto(ivr-number1,s,1). Save your change and you’re all set. Remember, XiVO is a real-time Asterisk server so all of your changes take effect immediately. There’s no rewriting of the entire Asterisk dialplan. Enjoy!

Taking Nerd Vittles’ XiVO IVR for a Test Drive

There’s a Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles Demo IVR:

Nerd Vittles Demo IVR Options
1 – Call by Name (say “Delta Airlines” or “American Airlines” to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)

Published: Thursday, May 26, 2016





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  • Some Recent Nerd Vittles Articles of Interest…

    2016, Celebrating The Preakness: CallerID Superfecta Rides Again with XiVO


    If you missed The Preakness Saturday, another Triple Crown bit the dust… or the mud in this case. But, if you had the winning Superfecta ticket, you made a $316 profit on your $1 bet. For the rest of us, there’s still a Superfecta win to celebrate, and this one’s free. We’ve begun porting CallerID Superfecta to the XiVO platform and today we’ll share that code with you together with lots of other goodies in our third roundup of Incredible PBX add-ons for the XiVO PBX. If you’re just joining the party, start with the first and second articles on XiVO, and then you’ll be ready to roll up your sleeves for Chapter 3.

    Installing CallerID Superfecta for XiVO

    As we mentioned in April, it’s always nice to see your baby grow up. Nearly a decade ago, we introduced an AGI script for Asterisk@Home known as CallerID Trifecta for FreePBX® 2.2.0. As sources of CNAM lookups expanded, a number of other individuals contributed code to support those lookups. When we added a fourth CNAM lookup source, the original application morphed into CallerID Superfecta. Then we gave up. The source lookups became too numerous to mention.

    For today, we’ve changed the design a bit to better accommodate the XiVO platform. There’s a single AGI script that houses the various CNAM lookup sources and the code to extract CallerID names from those sources. And there’s a dialplan script that let’s you specify which CNAM sources to use and in which order. As with the original release, CallerID lookups take the phone number of the caller and walk through your CNAM lookup sources in the order you specify until a CallerID name match is found. Then the result is returned to the PBX for use with the incoming call. The reason for all of this is historical. The Bell Sisters decided it was more profitable to dump CallerID name information in the bit bucket rather than passing it along with incoming calls. In that way, they could charge folks for looking up the matching name in their proprietary databases. A few CallerID lookup sources remain free, but many now are pay-as-you-go platforms with a typical lookup costing about half a cent. Unfortunately, all providers consider “WIRELESS CALLER” a successful lookup. Ka-Ching! We’ve documented the procedure to add additional CNAM lookup sources on the PIAF Forum. Please share your work!

    This release of CallerID Superfecta provides four lookup sources. That’s what a Superfecta is all about, picking four winners:

    0 - AsteriDex SQLite3 database
    1 - OpenCNAM (free from cache or commercial)
    2 - BulkCNAM (commercial only with free trial)
    3 - TelcoData (provider, city, and state of caller)

    There are three simple steps to putting everything in place. First, run the scripted commands below. Second, specify which CNAM sources you wish to use and in what order. Third, register with the commercial providers you’d like to use and plug your credentials into the CallerID Superfecta script.

    To install CallerID Superfecta, log into your server as root and issue the following commands:

    cd /
    apt-get -y install php5-xmlrpc
    wget http://incrediblepbx.com/cid-superfecta.tar.gz
    tar zxvf cid-superfecta.tar.gz
    rm -f cid-superfecta.tar.gz
    /etc/init.d/asterisk restart
    

    By default, CallerID Superfecta will attempt to use all four of the providers in the order shown to retrieve a CNAM match. If you have migrated your AsteriDex database to XiVO as we covered in last week’s article, then CallerID names will be provided for your most frequent incoming calls without ever accessing external sources. You won’t break anything by leaving all four CNAM sources activated. But, without signing up for service with OpenCNAM or BulkCNAM, your CNAM results will be diminished considerably. And a result of “WIRELESS CHARLESTON SC” from TelcoData doesn’t provide much of a clue as to who is calling. But at least you don’t get charged for that one.

    In the next release, we will add an optional feature that will populate entries in AsteriDex from CNAM data returned from OpenCNAM and BulkCNAM. The good news is, if you leave AsteriDex at the top of the CallerID Superfecta search list, you’ll never pay for the CNAM lookup of the same number twice. The bad news is, to keep the bad guys from self-populating your database with expensive phone numbers, you’ll need to password-protect the Voice Dialing application if it is part of your inbound IVR.

    To change the source list or sequence of CNAM lookups, open the XiVO GUI and navigate to IPX configuration -> Configuration files. Then edit cid-superfecta.conf. Find the line that looks like the following and specify the sources you wish to use and the sequence in which they should be searched using the source numbers listed above to replace 0-1-2-3. Separate your entries with hyphens. Then SAVE the file.

    same = n,AGI(nv-cid-superfecta.php,${XIVO_SRCNUM},0-1-2-3)
    

    To use the commercial CNAM services of either OpenCNAM or BulkCNAM, you first must register with them and provide a credit card. You then will be provided credentials to use for your CNAM lookups. These need to be inserted at the top of /var/lib/asterisk/agi-bin/nv-cid-superfecta.php. Then SAVE the file.



    Activating Traditional Asterisk Call Detail Recordings

    If you want to preserve the numbers AND names of those that call your PBX, you’ll need to activate the traditional CDR reporting mechanisms in Asterisk®.

    To activate SQLite3 logging of calls:

    cd /etc/asterisk
    sed -i 's|no|yes|' cdr.conf
    echo "[master]" >  cdr_sqlite3_custom.conf
    echo "table = cdr" >>  cdr_sqlite3_custom.conf
    echo "columns => calldate, clid, dcontext, channel, dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags, accountcode, uniqueid, userfield"  >>  cdr_sqlite3_custom.conf
    echo "values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}', '${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration)}', '${CDR(billsec)}','${CDR(disposition)}','${CDR(amaflags)}', '${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}'" >>  cdr_sqlite3_custom.conf
    chown asterisk:www-data cdr_sqlite3_custom.conf
    chmod 660 cdr_sqlite3_custom.conf
    sed -i 's|noload => app_cdr.so|;noload => app_cdr.so|' modules.conf
    sed -i 's|noload => cdr_sqlite3_custom.so|;noload => cdr_sqlite3_custom.so|' modules.conf
    sed -i 's|noload => func_cdr.so|;noload => func_cdr.so.so|' modules.conf
    /etc/init.d/asterisk restart
    

    To also activate CSV logging of calls:

    cd /etc/asterisk
    echo "[csv]" >> cdr.conf
    echo "loguniqueid=yes" >> cdr.conf
    echo "loguserfield=yes" >> cdr.conf
    echo "accountlogs=yes" >> cdr.conf
    sed -i 's|noload => cdr_csv.so|;noload => cdr_csv.so|' modules.conf
    /etc/init.d/asterisk restart
    

    To retrieve SQLite3 call log data, here are a few examples to get you started:

    ALL: sqlite3 /var/log/asterisk/master.db "select * from cdr"
    DATE: sqlite3 /var/log/asterisk/master.db "select * from cdr where calldate >= '2016-05-22'"
    NPA: sqlite3 /var/log/asterisk/master.db "SELECT * from cdr WHERE clid LIKE '%<843%'"
    DEST: sqlite3 /var/log/asterisk/master.db "SELECT * from cdr WHERE dstchannel LIKE '%411%'"
    FLDS: sqlite3 /var/log/asterisk/master.db "PRAGMA table_info(cdr)"

    To retrieve the CDR log in CSV format suitable for spreadsheets, download:

    /var/log/asterisk/cdr-csv/Master.csv
    

    Adding Asterisk ULAW Sound Files to Your XiVO PBX

    At least for us, the default sound files distributed with XiVO didn’t work. Here’s how to add the ulaw versions of all the files to your server:

    cd /usr/share/asterisk/sounds/en
    wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-ulaw-current.tar.gz
    wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-ulaw-current.tar.gz
    tar zxvf asterisk-extra-sounds-en-ulaw-current.tar.gz
    tar zxvf asterisk-core-sounds-en-ulaw-current.tar.gz
    rm -f *.tar.gz
    chown asterisk:asterisk *.ulaw
    

    Adding DISA Support to Your XiVO PBX

    If you’re new to PBX lingo, DISA stands for Direct Inward System Access. As the name implies, it lets you make calls from outside your PBX using the call resources inside your PBX. This gives anybody with your DISA credentials the ability to make calls through your PBX on your nickel. It probably ranks up there as the most abused and one of the most loved features of the modern PBX.

    We use two-step authentication with DISA to make it harder for the bad guys. First, the outside phone number has to match the whitelist of numbers authorized to use your DISA service. And, second, you have to supply the DISA password for your server before you get dialtone to place an outbound call. Ultimately, of course, the monkey is on your back to create a very secure DISA password and to change it regularly. If all this sounds too scary, don’t install DISA on your PBX.

    1. Download the DISA dialplan script into your /root folder where it can be edited:

    cd /root
    wget http://incrediblepbx.com/disa-xivo.tar.gz
    tar zxvf disa-xivo.tar.gz
    rm -f disa-xivo.tar.gz
    nano -w disa-xivo.txt
    

    2. When the editor opens the dialplan code, move the cursor down to the following line:

    exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1)  ; Good guy
    

    3. Clone the line by pressing Ctrl-K and then Ctrl-U. Add copies of the line by pressing Ctrl-U again for each phone number you’d like to whitelist so that the caller can access DISA on your server. Now edit each line and replace 701 with the 10-digit number to be whitelisted.

    4. Move the cursor down to the following line and replace 12341234 with the 8-digit numeric password that callers will have to enter to access DISA on your server:

    exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)
    

    5. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.

    6. Now copy the dialplan code into your XiVO setup, remove any previous copies of the code, and restart Asterisk:

    cd /root
    sed -i ':// BEGIN DISA:,:// END DISA:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
    cat disa-xivo.txt >> /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
    /etc/init.d/asterisk restart
    

    7. The traditional way to access DISA is to add it as an undisclosed option in an IVR that is assigned to one of your inbound trunks (DIDs). For the demo IVR that we installed last week, edit the ivr-1.conf configuration file and change the “option 0” line so that it looks like this. Then SAVE your changes.

    exten => 0,1(ivrsel-0),Dial(Local/3472@default)
    

    8. Adjust the inbound calls route of one of your DIDs to point to the demo IVR by changing the destination to Customized with the following Command:

    Goto(ivr-1,s,1)
    

    Here’s how ours looks for the Nerd Vittles XiVO Demo IVR:



    9. Now you should be able to call your DID and choose option 0 to access DISA assuming you have whitelisted the number from which you are calling. When prompted, enter the DISA password you assigned and press #. You then should be able to dial a 10-digit number to make an outside call from within your PBX.

    SECURITY HINT: Whenever you implement a new IVR on your PBX, it’s always a good idea to call in from an outside number 13 TIMES and try every key from your phone to make sure there is no unanticipated hole in your setup. Be sure to also let the IVR timeout to see what result you get.

    Adding Vitelity to XiVO for Flawless VoIP Calling

    We already have shown you several ways to take advantage of free VoIP calling in the U.S. and Canada as well as internationally. But, the old adage still holds true. You get what you pay for. And, if you’re using XiVO for your business or if you like a good night’s sleep without worrying about whether your spouse is going to stab you because of lousy phone connections, then splurge and spend a penny and a half a minute for outbound calls while getting unlimited incoming calls (4 at a time!) for only $3.99 a month. You’re worth it. The signup link for Vitelity is at the end of today’s article. Once you have your credentials, create a subaccount on the Vitelity site and then you’re ready to set up your Vitelity trunks with XiVO. We’ll use one trunk for incoming calls and a second trunk for outbound calls. The setup procedure for both trunks is already documented on the PIAF Forum. Make that your next stop!


    Simultaneous Cellphone Ringing for Inbound Calls with XiVO

    Speaking of incoming calls, wouldn’t it be nice if your cellphone also rang when XiVO calls arrived on your main extension. Then you don’t have to worry about missing a call just because you stepped out of the office.

    If you took our earlier advice and purchased a RingPlus phone with free monthly service, then you’re already covered. Setting up the RingPlus SIP trunk last week covered all the bases. And, there’s more good news from RingPlus. Now you can buy a phone in their Classifieds section without previously owning a phone. So you can hit the ground running with a phone AND a free calling plan. For example, $149 currently buys a brand new Moto E with 3,000 4G/LTE and SIP minutes, 3,000 SMS messages, and 3,000 MB of LTE data every month. And the monthly cost: ZERO!

    But, let’s assume you’re not the sharpest tool in the shed, and you still want your cellphone to ring when extension 701 rings on your PBX. Here’s how.

    In the User setup for your extension:

    1. Enter your cellphone number in the Mobile Phone Number field. Be sure it includes any necessary dial prefix so that it’s routed out through the correct trunk.

    2. On the same screen, you’ll find a Preprocess subroutine field. Enter the following there: pre-mobility

    3. SAVE your changes.

    Keep in mind that outbound calls in XiVO are routed out using dialing prefixes. If you have set up a trunk with a provider that allows CallerID spoofing such as Vitelity, Anveo Direct, or VoIP.ms, then you can preserve the caller’s original CallerID number on the forwarded call to your mobile phone provided the dial string for your cellphone number matches the format you set up for the trunk you wish to use. For example, if Exten for Vitelity is 8NXXNXXXXXX, then you would enter the number for your cellphone with an 8 prefix: 89991234567.

    Munin Makes XiVO Shine

    If you look under the Services tab and choose Graphics, the World of Munin will suddenly appear. There are literally dozens of gorgeous charts to tell you anything and everything you’d ever want to know about your server’s performance. Enjoy!



    Endpoint Management on Steroids… and It’s FREE

    If you’ve longed for an endpoint manager that would automatically configure your phones, the wait is over. XiVO supports literally dozens of phones out of the box. And the setup is integrated into the setup procedure for the users and devices. To get started, choose the Configuration tab and click Plugins. Next click on the + icon to load the default endpoint config files. We couldn’t do justice to this topic in a blog. That’s what tutorials are for. And XiVO has a 700+ page reference guide that will tell you everything you ever wanted to know about endpoint management.



    Adding NeoRouter VPN to XiVO

    We’ll finish up for this week by showing you how easy it is to add the NeoRouter Client to XiVO. In less than five minutes, you’ll be able to use XiVO’s NeoRouter private IP address to access your server securely from anywhere in the world. Start by reading our last introduction to NeoRouter.

    If you’re running XiVO on a 64-bit platform, issue the following commands to install the free NeoRouter client:

    cd /root
    wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/Ubuntu/nrclient-2.3.1.4360-free-ubuntu-amd64.deb
    dpkg -i nrclient-2.3.1.4360-free-ubuntu-amd64.deb
    

    If you’re running XiVO on a 32-bit platform, do this instead:

    cd /root
    wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/Ubuntu/nrclient-2.3.1.4360-free-ubuntu-i386.deb
    dpkg -i nrclient-2.3.1.4360-free-ubuntu-i386.deb
    

    Unless you want your server identified in NeoRouter as localhost, we recommend changing your hostname and rebooting your server at this juncture. Just edit /etc/hostname and give it a name, e.g. xivo. Then reboot.

    Now log back into your server as root and then log into your NeoRouter client. This will assign a private IP address to your XiVO server. The nrtap entry running ifconfig will tell you what that address actually is.

    nrclientcmd
    ifconfig
    

    Taking Nerd Vittles’ XiVO IVR for a Test Drive

    There’s a Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles Demo IVR:

    Nerd Vittles Demo IVR Options
    1 – Call by Name (say “Delta Airlines” or “American Airlines” to try it out)
    2 – MeetMe Conference
    3 – Wolfram Alpha (Coming Soon!)
    4 – Lenny (The Telemarketer’s Worst Nightmare)
    5 – Today’s News Headlines
    6 – Weather Forecast (enter a 5-digit ZIP code)
    7 – Today in History (Coming Soon!)
    8 – Speak to a Real Person (or maybe just Lenny if we’re out)

    Published: Monday, May 23, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


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

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

  • Some Recent Nerd Vittles Articles of Interest…

    The XiVO Adventure Continues: Adding Incredible PBX Goodies to Your Sandbox

    We began our XiVO adventure last week by introducing a terrific new communications platform for both businesses and hobbyists. This week we begin the task of incorporating the Incredible PBX Goody Bag into an already amazing PBX, and we’ll cover about a dozen new topics. We’ll also address a few XiVO basics such as where to find and how to use the backups that XiVO makes every morning while many of us are still sleeping. Since a new XiVO release is imminent, we also want to show you how easy it is to upgrade your server. Before we get to the good stuff, we want to take a moment and document a fourth platform for XiVO that will appeal to many large organizations and perhaps some of our pioneers. It’s our platform of choice for development of new applications.

    Installing XiVO as a VMware Virtual Machine

    If your organization runs VMware, you may not need to worry about finding your own platform for XiVO. You can get your IT guys to build you a XiVO VM using XiVO’s Debian-based ISO. Then again, you might have followed our tutorial and chosen to run your own VMware ESXi server. In either case, a quick refresher on getting XiVO installed may be helpful. Begin by downloading XiVO to your Windows desktop. Then log into VMware vSphere Client on your Windows machine to access ESXi.

    First, you’ll want to upload the XiVO ISO as a VMware guest operating system so that it can be used to create virtual machines at any time. From your inventory, click on the Configuration tab. Then click Storage under the Hardware listing. When your Datastore appears, right-click on datastore1 and choose Browse Datastore. Finally, click the Upload Files to this Datastore icon in your Datastore Browser and choose Upload File option. Choose the XiVO ISO from the Upload Items menu to upload it into your Datastore.

    Now we’re ready to create a Virtual Machine. Right-click on the IP address of your VMware server and choose New Virtual Machine. Leave the Typical Configuration option selected and click Next. Give the virtual machine a name and click Next. Select the Destination Storage device and click Next. For the Operating System, choose Linux and pick Debian 8 (64-bit) then Next. Choose the NIC to use for the VM and click Next. Choose your Virtual Disk Size and Thin Provision option then Next. Check the box to Edit Virtual Machine Settings Before Completion and click Continue. Click the Options tab in Virtual Machine Properties and click Boot Options. Check the Force BIOS Setup option on next boot. Click Finish.

    Starting your virtual machine the first time is not exactly intuitive so follow these steps carefully and in order. Keep in mind that, on the initial bootup of your virtual machine, what we want to do is run the XiVO ISO installer just as if we had booted a standalone machine using a CD on which we had burned the XiVO ISO. To begin the boot process correctly, first highlight your new VM by clicking on it and then choose Power on Virtual Machine. Next, click on the CD/DVD icon in the toolbar, choose CD/DVD Drive 1, choose connect to ISO image on Datastore. Double-click on datastore1 and then double-click on the XiVO ISO we uploaded previously. Now click on Launch Virtual Machine Console icon in the toolbar. When the BIOS setup utility appears, click in the window and use the Right Arrow key to move to the Boot tab. Move the CD-ROM option to the top of the list by highlighting it and pressing the + key to move it up. Press F10 to Save and Exit from the BIOS Setup and boot into your XiVO ISO. Click Install option to begin the regular XiVO installation procedure. When you finish the install, log into your server as root and obtain your IP address: ifconfig. You then can exit from the Console window by pressing Ctrl-Alt and use a browser to complete the install by pointing to the IP address of your virtual machine. Don’t forget that root SSH access is disabled by default. Our original tutorial will show you how to fix it AND install the Travelin’ Man 3 firewall whitelist to protect your server.

    Adding a RingPlus SIP Trunk for Unified Communications with Sprint

    Last week we began the XiVO adventure by turning on free Google Voice calling in the U.S. and Canada. Today we want to integrate smartphones into the mix by providing an incredibly simple and dirt cheap way to expand your XiVO communications platform while transparently meshing it with a RingPlus smartphone and the Sprint cellular network. When we’re finished, calls to your smartphone will also ring on one or more XiVO extensions. And designated users of your XiVO PBX will be able to place free calls to U.S. destinations using a SIP trunk tied directly to your RingPlus cellular account. These calls won’t be cellular. They’ll be pure VoIP calls using Sprint’s Internet backbone so listen for that pin to drop. If you have a (free) unlimited calling plan with RingPlus, then you’ll inherit a (free) unlimited calling plan for your XiVO PBX. Stated another way, whatever calling minutes you have with RingPlus can be shared on your XiVO PBX as inbound and outbound VoIP calls. The silver lining is that voicemails left on RingPlus get transcribed and delivered to your email address in seconds. So you get the best of both worlds. That’s what Unified Communications is all about!

    Don’t worry if you’re late to the party and not yet a RingPlus user. They announce new deals every week so just check every few days until you find a plan that meets your needs. You won’t have to wait long. Here’s a list of all the previously announced PROMOS to give you a good handle on the scope of the RingPlus offerings. Deals don’t last but a couple hours or days so check often or sign up for RingPlus Alerts on SlickDeals and you’ll be the first to know! There’s a terrific deal tonight only from 8 p.m. until midnight.



    We’ve already documented the XiVO setup procedure on the PIAF Forum so hop over there to see how easy this is. Keep in mind that XiVO differs a bit from FreePBX® in the way Outbound Calls are managed. In FreePBX, you prioritized the routes by arranging them in a hierarchical list. In XiVO, you use unique dial strings, e.g. NXXNXXXXXX, for every Outbound Route. So, if you’re adding RingPlus to an existing XiVO server that already is using the NXXNXXXXXX dial string, then you’d need to use a different dial string to route calls out through the RingPlus trunk, e.g. 77NXXNXXXXXX with Stripnum=2. That tells XiVO that your users will dial calls to be handled by RingPlus with a prefix of 77 (RP), and then we want XiVO to strip off the first two digits before passing the call to the RingPlus SIP trunk for processing.

    If you’re new to RingPlus, start with the original Nerd Vittles article for some background and then follow the RingPlus threads on the PIAF Forum and DSL Reports for the latest tips and tricks.

    Adding a FreeVoipDeal (Betamax) SIP Trunk for Free International Calling

    Before deploying a SIP trunk from one of the Betamax companies, read our latest article about Betamax for tips and tricks and land mines to watch out for. Then click the link below when you’re ready to deploy FreeVoipDeal as a trunk on your XiVO PBX:


    Everything You Need to Know About XiVO Backups

    Another feature of XiVO that separates the men from the boys is its documentation. In the case of backups, you’ll find everything you need to know here. All backups are stored on your XiVO server’s local drive in /var/backups/xivo. Be sure you have ample storage space available and, if you’re smart, you’ll copy both data.tgz and db.tgz from the local drive to a safe remote location periodically just in case disaster strikes. The documentation shows you how to quickly restore a backup should that ever become necessary.

    Upgrading XiVO to the Latest Release

    The XiVO development cycle is nothing short of miraculous. A new version is released every two weeks! The average time to close a bug has dropped from 315 days in 2009 to 28 days in 2012! You’ll probably want to keep your system current. 🙂

    Upgrading XiVO is even easier than restoring a backup. Upgrade documentation is available here. Because we’ve added the Travelin’ Man 3 firewall, we recommend stopping IPtables during an upgrade and then restarting it when you’re finished. Your phone system is disabled during the upgrade. When upgrading XiVO, remember to also upgrade all associated XiVO Clients. Be sure to verify that things are back to normal once the upgrade procedure is completed: xivo-service status.

    The commands to upgrade your XiVO PBX are as follows:

    /etc/init.d/netfilter-persistent stop
    xivo-upgrade
    iptables-restart
    

    Update: There’s a great tip from one of the XiVO developers on a better way to do this. See the first comment below.

    Prerequisites for Today’s XiVO Adventure

    If you’re just getting started with XiVO, DON’T START HERE. Read our first article. Be sure you have completed the following 8 steps before proceeding:

    1. Set Up Root SSH Access to Your XiVO PBX
    2. Set Up the Travelin’ Man 3 IPtables Firewall Using an SSH/Putty Connection
    3. Complete the XiVO Setup Using a Web Browser
    4. Create At Least One User with a 701 Extension
    5. Create At Least One SIP Trunk to Use for Outbound Calls
    6. Configure Outbound Call Settings for Your Trunk Using NXXNXXXXXX
    7. Configure an Inbound Route for Trunk Pointing to Your User Account
    8. If Behind NAT Firewall, Set externip and local network in General Settings -> SIP Protocol -> Network

    Creating a MeetMe Conference Room for XiVO

    There are just two steps to setting up a conference room. First, you need to add the extensions you will use for your conferences in the Default context. Then you add the Conference Room under IPBX Settings. Let’s set up a conference room extension 2663 (C-O-N-F). In your Default context, click on the Conference Rooms tab and enter an extension range of 2663-2664 and click Save. Then, in the Conference Rooms tab, click the + icon to add a new CONF conference room at extension 2663 in the Default context. You can experiment with the other settings when you have some spare time. The entries are pretty much self-explanatory. Click Save to activate your conference room. You won’t have music on hold for the first participant just yet. We’ll do that next.

    Adding Music on Hold to XiVO

    By default, XiVO doesn’t come with any music on hold. Fortunately, Digium has negotiated a music on hold license that you can use to add it to your PBX at no cost. While logged into your XiVO PBX as root, issue the following commands:

    cd /
    wget http://incrediblepbx.com/moh-xivo.tar.gz
    tar zxvf moh-xivo.tar.gz
    /etc/init.d/asterisk restart
    

    Asterisk Application Development with XiVO

    For those coming from the FreePBX world, here’s a quick introduction to Asterisk application development on the XiVO platform. First and foremost, there are more similarities than differences. In the FreePBX environment, custom dialplan code was stored in /etc/asterisk/extensions_custom.conf. For custom extensions that you wanted to add, that code had to appear in the [from-internal-custom] context. For custom dialplan contexts, those appeared immediately below the last entry in the [from-internal-custom] context. If your custom code appeared anywhere else, there was always the risk that it might be overwritten with your next FreePBX reload.

    The XiVO design is quite different. As we noted last week, it is not an Asterisk code generator at all, unlike FreePBX. Instead, it has a realtime interface to Asterisk using its PostGreSQL database engine. Updates are nearly instantaneous without reloading Asterisk modules from disk.



    The other advantage is you won’t have to worry about XiVO stepping on your custom code as long as you leave PostGreSQL alone. HINT! The good news is there still are hooks to add your own custom dialplan extensions and code as well as PHP/AGI scripts. And it’s easy. In XiVO, custom extensions are stored in xivo-extrafeatures.conf which you’ll find in the /etc/asterisk/extensions_extra.d directory. Don’t edit files in /etc/asterisk/extensions_extra.d from the Linux command prompt! Instead, use the editor built into the XiVO GUI by selecting Configuration Files under IPBX configuration. This will automatically assure that realtime updates are posted correctly. To add additional contexts to your dialplan, create separate files for each context and store them in this same directory. Again, the easy way to make certain that Asterisk is updated automatically when you add new code snippets is to create and edit them within the XiVO GUI. These files all will appear under IPBX Configuration -> Configuration Files as well.

    In order to better mimic the FreePBX way of doing things so that your PHP/AGI scripts work in either environment, we recommend issuing the following symlink while logged into XiVO. We’ll do it as part of the SQLite3 install below.

    ln -s /var/lib/asterisk/agi-bin /usr/share/asterisk/agi-bin
    

    Once you’ve established the symlink, PHP/AGI scripts can be migrated from FreePBX to XiVO directly using the same directory structure for storage: /var/lib/asterisk/agi-bin. As with FreePBX, all files in this directory should be owned by asterisk with 775 permissions:

    chown asterisk:asterisk /var/lib/asterisk/agi-bin/*
    chmod 775 /var/lib/asterisk/agi-bin/*
    

    There are many other powerful features in XiVO that weren’t available at all in FreePBX. We’ll cover some of them in coming months. In the meantime, this brief overview of the dialplan environment should be sufficient to let you start building.

    Installing SQLite3 to Support Incredible PBX Applications

    There’s one other difference between XiVO and FreePBX that we’ve already touched upon. But it bears repeating here. XiVO doesn’t use MySQL or MariaDB for its database management tasks. Instead, the XiVO development team chose PostGreSQL which is equally powerful, but different. For the Incredible PBX application suite, we’ve chosen to rewrite the ones that depend upon MySQL so that they can run under SQLite3 which is considerably less processor intensive than running both PostGreSQL and MySQL 24/7. We also didn’t want to interfere with the PostGreSQL setup of XiVO since it is an integral component of the product and will get upgraded automatically as part of the regular XiVO upgrade cycle.

    Here’s how to put the SQLite3 and corresponding ODBC components in place on your new server. While logged into your server as root, simply issue the following commands:

    cd /
    wget http://incrediblepbx.com/sqlite3-xivo.tar.gz
    tar zxvf sqlite3-xivo*
    rm -f sqlite3-xivo.tar.gz
    cd /root
    ./sqlite3-xivo.sh
    

    Running a couple SQLite3 queries using the ZIPCODES and ASTERIDEX databases will give you a feel for the performance you can expect from SQLite3. The queries might look like this:

    sqlite3 /var/lib/asterisk/agi-bin/zipcodes.sqlite "select zip,city,state from zipcodes where zip=29401;"
    sqlite3 /var/lib/asterisk/agi-bin/asteridex.sqlite 'select name,out from user1 where name LIKE "%Airlines%";'
    

    And here are the results of the two queries:

    29401|CHARLESTON|SC
    --------------------------------
    American Airlines|8004337300
    Continental Airlines|8005250280
    Delta AirLines|8002211212
    Frontier Airlines|8004321359
    Iberia AirLines|8007724642
    Midway Airlines|8004464392
    Northwest Airlines|8002252525
    Southwest Airlines|8004359792
    Ted Airlines|8002255833
    United Airlines|8002416522
    WestJet Airlines|8005385696
    Yemen Airlines|8009368300
    

    We’ve included a bonus script in /root that will let you convert existing MySQL databases to SQLite3. For example, if you’re currently using AsteriDex on another Incredible PBX platform, it only takes a couple seconds to convert your MySQL database to SQLite3. The syntax to run the script should look like this:

    ./mysql2sqlite3.sh -u root -ppassw0rd yourdatabase | sqlite3 yourdatabase.sqlite
    

    You obviously cannot run the script on your XiVO server because your MySQL databases and MySQL itself are missing. So move the script to the server on which your MySQL databases are stored and run it there using the above syntax. Then copy the asteridex.sqlite file to your XiVO server and save it in /var/lib/asterisk/agi-bin.

    Installing and Activating the Festival TTS Engine with Asterisk

    We’ve got a couple more building blocks to put in place to support Incredible PBX applications. Then we’ll be ready to kick the tires with a few applications to get you started. In coming weeks, we’ll finish up the conversion of the remaining apps, and then we’ll publish an Incredible PBX installer for XiVO with all the pieces. But why wait? Finish up installing the remaining pieces today, and you’ll have something to play with. And, as we said, it will also provide you with simple scripts so you can actually see how Incredible PBX is put together.

    Many of the Incredible PBX applications rely upon text-to-speech and/or voice recognition (speech-to-text) to work their magic. Neither comes installed with XiVO by default, but Asterisk was properly configured to support Festival so let’s work with that. Festival is the Big Brother of FLITE and includes some additional voices of fairly good quality. The XiVO Demo IVR will give you an idea of the TTS voice quality you can expect:

    To get Festival installed and activated for use with Asterisk, issue these commands:

    cd /
    wget http://incrediblepbx.com/festival-xivo.tar.gz
    tar zxvf festival-xivo.tar.gz
    cd /root
    ./festival-xivo.sh
    

    Installing Dial Plan Code for Sample Incredible PBX Applications

    Now we’re ready to put today’s Dial Plan Code and IVR in place and load the PHP/AGI components necessary to make the sample applications work. Here’s how:

    cd /
    wget http://incrediblepbx.com/ivr-xivo.tar.gz
    tar zxvf ivr-xivo.tar.gz
    /etc/init.d/asterisk restart
    

    Installing and Activating Voice Recognition for XiVO

    Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your XiVO PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Don’t forget to add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.

    Taking XiVO on a Test Drive with the Incredible PBX Apps

    Now set up a softphone using the IP address of your XiVO server and the Line credentials for Extension 701. When you obtain your credentials, double-check to make sure all of the fields for the Line are filled in correctly as shown below:

    Once your softphone is registered, you can try out some of the sample applications:

    • 4871 (IVR1) – Allison’s Demo IVR
    • 411 (Voice Dialing) – Call by Name (try “Delta Airlines”)
    • 2663 (CONF) – MeetMe Conference with Music on Hold
    • 951 – Yahoo! News Headlines (TTS)
    • 947 (ZIP) – NWS Weather by ZIP Code
    • 53669 (LENNY) – The Telemarketer’s Worst Nightmare

    You can review the Dialplan code in the GUI by choosing Configuration Files and clicking xivo-extrafeatures.conf. The sample IVR code is in ivr-1.conf.

    Taking Nerd Vittles’ XiVO IVR for a Test Drive

    There’s also a new Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles IVR: 1-843-606-0555. Setup at PacificNX took less than a minute. Enjoy!

    Published: Thursday, May 12, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


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

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

  • Some Recent Nerd Vittles Articles of Interest…

    2016, The Year of (real) VoIP Choice: Meet XiVO, a UC Solution for Any Business

    We promised you that 2016 was going to be a year filled with surprises, and today we’re pleased to introduce another open source, pure GPL3 solution for any business. Whether your requirements are a call center or a versatile phone system for hundreds of employees, XiVO™ offers a compelling unified communications solution that checks all the boxes. Unlike some products that function merely as a code generator for Asterisk®, XiVO is in a league of its own. XiVO is actually an integral component of the Asterisk application itself. It manages your telephony server in realtime using its versatile PostGreSQL database platform. Did we mention it’s also a great playground for hobbyists and SOHO VoIP enthusiasts? Let’s get started.

    There’s no way to do justice to a product like XiVO in a single article. So our plan is to introduce XiVO today and get your platform up and running where you can make and receive free calls throughout the United States and Canada. Then you can add Incredible PBX components and additional SIP providers as we continue to build them out. Just follow along with our Incredible PBX development for XiVO on the PIAF Forum, and you’ll get a first-hand look at how sausage is made. We already have text-to-speech applications for news and weather up and running. You can take them for a test drive by calling the XiVO demo:

    And, of course, we’ve integrated the Travelin’ Man 3 IPtables firewall to provide rock-solid security for XiVO, and we’ll cover that today as well. As part of this development process, you’ll discover how easy it is to build Asterisk applications for XiVO on your own. And hopefully you’ll share some of your creations with the rest of us. That’s what open source development is all about.

    Choosing an Experimental Platform for XiVO

    We’re just getting started with XiVO development so, like us, we’re assuming you’ll want to kick the tires a bit before jumping into a new VoIP solution for the long haul. That means you first must choose a platform on which to install XiVO. We have several recommendations for you. If you have a robust desktop machine with lots of RAM and processing power, then installing XiVO under VirtualBox may be the way to go. We actually use an iMac with 16GB of RAM, and it provides plenty of horsepower to run VirtualBox and XiVO. With VirtualBox, we’ll start by downloading the XiVO ISO.

    We didn’t mention that XiVO has been under development for over 10 years and is supported by the original developers with financial support from Avencall. Because of its Canadian roots, it seems only fitting that many may wish to consider CloudAtCost in Canada as an appropriate site to host your experimental XiVO server. A one-time payment of $10.50 still buys you a sandbox in the cloud for life with coupon code TAKE70, and XiVO installs on the CloudAtCost platform without a hiccup. For a CloudAtCost implementation, we’ll start by creating a Debian 8 server.1 And then we’ll download and run the XiVO installation script to build our XiVO server. Finally, we’ll walk you through setting up XiVO on a $5/month Digital Ocean Droplet which provides state-of-the-art performance at rock-bottom Cloud pricing. So begin by choosing your hardware platform from the three options below:

    1. Installing XiVO as a VirtualBox Virtual Machine

    For standalone implementations including VirtualBox, we’ll begin by downloading the 64-bit XiVO Server ISO to your desktop. Next, create a VirtualBox 64-bit Debian VM platform with 1024 MB RAM and at least a 10GB virtual drive. In System Settings, enable I/O APIC and disable the other options. Select a Sound Card to match your machine and configure Network Adapter 1 as a Bridged Network Device. In the Storage Settings (shown below) for your (1) Empty IDE Controller, (2) select the downloaded XiVO ISO as your installation media. Start the VM and proceed through the initial install.

    Click Install, choose your language, pick your time zone, choose your keyboard map, create a very secure root password, and choose a Debian mirror that’s close to your server. Choose /dev/sda as your bootloader assuming that’s the disk drive configured by VirtualBox. In less than 10 minutes, the install will complete and your VM will reboot. Log into your server as root and obtain your IP address: ifconfig. You’ll need it for the web configuration step that comes next.

    2. Installing XiVO as a CloudAtCost Cloud-Based Server

    You can’t use an ISO as the installation media at CloudAtCost so we have to start by building a 64-bit Debian 8 virtual machine with at least 512 MB RAM and a 10GB virtual drive. No need to choose a larger drive at the moment since there’s a bug in CloudAtCost’s installer for Debian 8. See the footnote for details. Once your virtual machine is built, log in as root and issue the following commands to kick off the XiVO install:

    apt-get -y remove apache2*
    apt-get update
    apt-get -y upgrade
    reboot
    # log back in as root and...
    wget http://mirror.xivo.io/fai/xivo-migration/xivo_install_current.sh
    bash xivo_install_current.sh
    

    3. Installing XiVO as a Digital Ocean Droplet

    As with CloudAtCost, you’ll need to begin your XiVO adventure at Digital Ocean by first signing up for an account. With our referral code, you’ll get a $10 credit (and so will Nerd Vittles). That’s good for two full months of service to kick the tires of XiVO without ever spending a dime. Once your account is set up, create a $5/month Debian 8 (64-bit) Droplet. When you receive the email with your droplet credentials, log into your new server as root using SSH/Putty and issue the following commands to get Debian 8 squared away:

    apt-get update
    apt-get upgrade -y
    dd if=/dev/zero of=/swapfile bs=1024 count=1024k
    chown root:root /swapfile
    chmod 0600 /swapfile
    mkswap /swapfile
    swapon /swapfile
    echo "/swapfile          swap            swap    defaults        0 0" >> /etc/fstab
    sysctl vm.swappiness=10
    echo vm.swappiness=10 >> /etc/sysctl.conf
    free
    reboot
    

    After the reboot, log into your server again with your new root password and kick off the XiVO install:

    wget http://mirror.xivo.io/fai/xivo-migration/xivo_install_current.sh
    bash xivo_install_current.sh
    

    Enabling SSH Root Access with XiVO

    If you installed XiVO using the XiVO ISO, then root logins via SSH are disabled by default. Only enable it if you plan to also implement the firewall in the next step! To enable root logins via SSH, log into the server console as root and edit the SSH config file: nano -w /etc/ssh/sshd_config. Find the line in the Authentication section that begins with PermitRootLogin and change it to: PermitRootLogin yes. Save your change (Ctrl-X, y, ENTER) and then restart SSH: /etc/init.d/ssh restart.

    Setting Up a Firewall to Protect XiVO

    We don’t build PBXs without a rock-solid firewall, but it’s your phone bill so the choice is all yours. The Travelin’ Man 3 implementation of the Linux IPtables firewall provides a safe computing platform using a WhiteList to only allow access by trusted users and providers. You can add additional users to the whitelist as desired using add-ip and add-fqdn in the /root folder. Restart your firewall using only this command: iptables-restart. If you’ll be using FQDNs in your WhiteList, then add the ipchecker script to your cron jobs. Then review Step #5 in the TM3 tutorial.

    echo "*/10 5-22 * * * root /root/ipchecker > /dev/null 2>&1" >> /etc/crontab
    

    It’s imperative that you set this up from a client workstation that’s running SSH or Putty. Otherwise, you may inadvertently lock yourself out from your own server. While logged into your server via SSH as root, issue the following commands:

    cd /root
    wget http://incrediblepbx.com/firewall-xivo.tar.gz
    tar zxvf firewall-xivo.tar.gz
    rm -f firewall-xivo.tar.gz
    ./tm3-xivo.sh
    

    Configuring XiVO with a Web Browser

    Once the basic install is completed, you use a web browser to actually configure and manage your XiVO server. To get things started, point your browser to the IP address of your XiVO server. Choose your Language. Accept the GPL3 license agreement. Then fill in the blanks to create a Hostname for your server (XiVO), a domain name (some domain that you own or one chosen from your favorite dynamic DNS provider), a very secure Web interface password (choose as if your phone bill depends upon it). The network interface and DNS server entries should already be correct. Click Next.

    On the second configuration screen, choose an Entity (department/organization name or IncrediblePBX will suffice). Then set up the Contexts to manage calls on your PBX:

    • Internal Calls Context: manages extension numbers that can be reached internally
    • Incalls Context: manages calls coming from outside of your system
    • Outcalls Context: manages calls going from your system to the outside

    Here’s what we’ll be using by way of example:

    Finally, validate your entries to complete the configuration. Now log into your XiVO server as root using your newly created web password. You should get a status screen that looks something like this. If you had any doubts about the quality of the XiVO product, this should put your mind at ease. 🙂

    Logging Into the XiVO Web Interface

    To make changes in your XiVO setup, you’ll need to log into the web interface at the IP address of your XiVO PBX. Login with root as the username together with the Web Interface Password you set up above. You can change this password at any time under the Configuration tab by clicking on Users and editing your existing settings.

    Creating Users and Lines with XiVO

    For those migrating from the FreePBX® world, you’re probably most familiar with the procedure for creating extensions. More advanced administrators may have switched to device and user mode where users and devices are created separately. Phone numbers or extensions were associated with users while phone instruments were associated with devices. In the World of XiVO, we’ll start with the simplest configuration, and you can move on from there when you’re ready. In our scenario today, we’ll create a couple of users. Each user has a Name, Language, Time Zone, and other optional characteristics such as a Mobile Phone Number which can ring simultaneously whenever a user receives a call to his or her local XiVO phone number. By adding a Line (aka Phone Number) for the user as the user account is created, XiVO will automatically generate a separate Line with username and password credentials. This Line will be associated with the User during the initial user setup procedure, and this Line then can be registered to a SIP phone, softphone, or XiVO client (which we will cover separately down the road). In the example below, we’re using Nerd Uno’s extension 701 (associated with line 3jz8tsr0) to call Nerd Dos’ extension 702 (associated with line 8fmne2x4).

    XiVO has an excellent tutorial that covers creating Users with a SIP Line. So jump there and add a couple of Users following the steps in the tutorial. When you’re finished, you’ll have two Users and two associated Lines with credentials to set up SIP phones. Since you’re just getting your feet wet and will probably make some mistakes, it’s probably a good idea to turn off Fail2Ban while you’re experimenting. Otherwise, you may accidentally lock yourself out of your server (ask us how we know) and think it’s a problem with XiVO. Here’s how:

    /etc/init.d/fail2ban stop
    

    To set up your SIP phones, you’ll need the credentials for each of the two lines. Under the Lines tab, click on the Pencil icon to reveal the Username and Password. Fill in the missing pieces as shown below and make certain that your NAT entry is set to Yes.

    With those credentials in hand, go ahead and configure a couple of SIP phones and make certain you can call between them with audio in both directions before proceeding. For those with a Mac, Telephone is perfect for experimentation because you can set up multiple softphones and place calls between them.

    IMPORTANT: If your server is sitting behind a NAT-based firewall, you must set the external and local network IP addresses for XiVO in General Settings -> SIP Protocol. You’ll find the fields in the Network tab.

    Configuring a SIP Trunk for Google Voice with XiVO

    Now that you have internal calls working, let’s turn our attention to connecting your PBX to the rest of the world. We obviously can’t cover the setup for every SIP provider, but we can provide a good example that will get our U.S. friends free calling in the U.S. and Canada. We’ve chosen the Simonics SIP Gateway to Google Voice because a one-time payment of $5.99 gets you a traditional SIP trunk to interface with any existing Google Voice number. If you don’t have a Google Voice number, sign up here. In your Google Voice Settings, make sure Forward Calls to Google Chat is enabled and disable Call Screening in the Calls tab. Then, with your Google credentials and Google Voice number in hand, visit the Simonics web site to sign up for service. Sign in with your Google credentials and complete the registration process. Once you have your Simonics account name and password, log into your XiVO web portal.

    With credentials in hand, on the XiVO side, start by choosing the SIP Protocol tab under Trunk Management. There are actually three tabs to configure for the SIP trunk. Begin in the General tab and make it look like this using your credentials. NOTE: The complete FQDN for the Simonics gateway should be gvgw.simonics.com:

    Next, click on the Register tab and reenter your credentials. Leave the empty fields exactly as shown. Be sure the Register box is checked.

    Next, in the Signaling tab, change the Monitoring option to Yes and then click Save. Monitoring is the XiVO equivalent of the SIP Qualify option.

    We also need to make one minor adjustment in the SIP Protocol Defaults in the General Settings. Just Save your settings after checking Match users with ‘username’ field.

    Next, we need to tell XiVO how to process Incoming and Outgoing Calls using the Google Voice SIP trunk. Under the Call Management section, let’s begin with the Incoming Calls setup by creating a new Incoming Calls DID for your 11-digit Google Voice number. To keep things simple, we’ll route the incoming calls to the User mapped to extension 701:

    For Outgoing Calls, we need to route calls with a specific dial string out the Simonics SIP trunk using the to-extern context. By way of example, we’ve set this up using a dialing prefix of 48 (GV) and a 10-digit number. We’re letting XiVO supply the missing 1 country code required by Google Voice, and we’ll let XiVO strip off the 48 prefix in processing the outbound calls. If this is your only outgoing trunk, you may prefer not to use a dial prefix at all. In that case, change the dial string to a 10-digit number (NXXNXXXXXX) and set Stripnum to 0.

    Well, that’s enough for today. There’s complete XiVO PDF Documentation available here. We’ll have lots more to say about XiVO in coming weeks. Come join the party!

    Continue reading Part 2.

    Published: Thursday, May 5, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


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

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

  • Some Recent Nerd Vittles Articles of Interest…

    1. There’s a glitch in the CloudAtCost builds for Debian8. Regardless of how much disk storage you allocate, CloudAtCost will only use 10GB. Moral: Don’t waste your resources by allocating more than 10GB of disk space. This is an experimental platform, and 10GB will suffice. If you really need more space, this thread on the PIAF Forum will walk you through expanding the storage allocation beyond the 10GB threshold. []

    Sleep Well: Create a $10.50 Incredible Backup Server in the Cloud with WebDAV

    With the impending demise of Copy.com, it seemed like a good time to revisit the subject of backups and to do a little advance preparation for that rainy day when your Incredible PBX™ server decides it’s taken its last breath. We recently documented how to build an Incredible PBX in the Cloud for a one-time cost of $10.50. And we showed you how to build a Linux Sandbox in the Cloud for the same bargain-basement price. Today, we’re adding a third way to spend one day’s lunch money with our new Backup Server in the Cloud at CloudAtCost. And, like the other two, a one-time investment of $10.50 gets you a 10GB cloud repository to store your most important Asterisk® files for life!1 If you’re feeling really adventurous, you can double or quadruple your resources and your storage capacity at the same great 70% off rates with CloudAtCost coupon code: TAKE70. Some have asked us for a referral code to give credit where credit is due. Thanks for thinking of us, but we already have all of the CloudAtCost resources we could ever use. So this one, like the two before it, is on us!



    We recommend you start by building an Incredible PBX platform at CloudAtCost using our previous tutorial. Is it production-ready? Probably not. Is it a good standby server which can swing into action when your primary server croaks? Absolutely. Can it be used for off-site storage of backups from your primary Incredible PBX server? You bet. And today we’ll show you how. It’s about a 10-minute process once you have Incredible PBX up and running in the Cloud. We’ll also provide an updated Incredible Backup script to transparently upload backup images to your new CloudAtCost backup server.

    Got DAV?It’s been quite a while since we first explored WebDAV back in 2005. Today we’re going to bolt on WebDAV to your existing Incredible PBX platform so that some of that spare storage space in the Cloud can be used to house snapshot images of your Incredible PBX production server. Since this will be a fully-functioning Incredible PBX server in addition to serving as a backup server, it can perform double-duty as a hot standby on a moment’s notice. When disaster strikes, restore the latest backup which happens to be colocated on your Cloud server, and you’ll be back in business.

    Overview. As you probably know, WebDAV is an acronym for Web-based Distributed Authoring and Versioning. Simply put, it is an HTTP protocol extension that allows people anywhere on the Internet to edit and manage documents and other files using the same protocol and port used for surfing the web. In the Mac and Linux worlds, WebDAV provides a Disk Volume that “looks and feels” like any other networked hard disk. In the Windows world, WebDAV is called Web Folders. They can be used like any other mapped drive in Network Neighborhood. If you’re still a little fuzzy about the WebDAV concept, think of how you link to another drive on your local area network. WebDAV gives you the same functionality across the entire Internet with virtually the same ease of use. Depending upon user privileges, of course, you can copy files to and from a WebDAV volume, and the protocol imposes versioning control through file locking to assure that multiple people with access rights don’t change the same file at the same time.

    Initial Setup of WebDAV in the Cloud. For today, we’re assuming you already have a functioning Incredible PBX server at CloudAtCost running under CentOS 6.7. If not, start with our tutorial here. If you’d prefer to use the Linux Sandbox configuration for your WebDAV platform, skip down to the next section. To keep things simple, we’re going to set up a separate dav directory within your existing Incredible PBX cloud server to use for WebDAV storage. This means files and folders managed with WebDAV will appear in /var/www/html/dav on your server. We’ll password-protect the directory using Apache web credentials for the admin user. You first must set up these credentials by issuing the following command while logged into your server as root:

    htpasswd /etc/pbx/wwwpasswd admin
    

    To activate WebDAV on your Incredible PBX server at CloudAtCost, while still logged into your server as root, issue the following commands:

    mkdir /var/www/html/dav
    chown asterisk:asterisk /var/www/html/dav
    chown asterisk:asterisk /var/lib/dav
    cd /etc/pbx/httpdconf
    wget http://incrediblepbx.com/dav.conf
    service httpd restart
    

    Keep in mind that WebDAV is running on an Incredible PBX server which means that remote HTTP access will require that your remote IP address be in the IPtables WhiteList. You can add it easily using the add-ip or add-fqdn utilities in /root. Don’t forget, or none of this will work.

    Setting Up WebDAV on a CloudAtCost Linux Sandbox. If you’d prefer to set up WebDAV on a Linux Sandbox at CloudAtCost rather than the Incredible PBX platform, begin by installing the sandbox by following along in the Nerd Vittles tutorial. Once you’re up an running, issue the following commands to activate WebDAV:

    mkdir /etc/pbx
    htpasswd -c /etc/pbx/wwwpasswd admin
    mkdir /var/www/html/dav
    chown apache:apache /var/www/html/dav
    cd /etc/httpd/conf.d
    wget http://incrediblepbx.com/dav.conf
    service httpd restart
    

    You won’t have to whitelist the IP address of your local Incredible PBX server in the IPtables firewall running on your WebDAV server at CloudAtCost because port 80 already is whitelisted in the default Linux Sandbox setup.

    Accessing WebDAV in the Cloud. As installed, you’ll need your username (admin) and your Apache password assigned above to access your WebDAV server in the Cloud. Use a browser for read only access to the dav directory at the IP address of your server, e.g. http://23.45.67.89/dav. Or establish a network share to the WebDAV resource for read and write access.

    Configuring a Local CentOS/SL Server for WebDAV Access. Linux needs something special in order to treat remote WebDAV resources as part of your local file system. Fortunately, there is a packaged solution that does all the heavy lifting for you. On every CentOS/Scientific Linux server from which you want to access remote WebDAV resources, issue the following commands while logged into the server as root:

    yum -y install davfs2
    mkdir /dav
    cd /root
    wget http://incrediblepbx.com/incrediblebackup-dav
    chmod +x incrediblebackup-dav
    

    Configuring a Local Debian/Ubuntu/Raspbian Server for WebDAV Access. The setup drill is much the same as it is for CentOS except the package installation syntax needs to be adjusted. On every Debian, Ubuntu, or Raspbian (Raspberry Pi) server from which you want to access remote WebDAV resources, issue the following commands while logged into the server as root:

    apt-get -y install davfs2
    mkdir /dav
    cd /root
    wget http://incrediblepbx.com/incrediblebackup-dav
    chmod +x incrediblebackup-dav
    

    Connecting to Your WebDAV Server in the Cloud. The new Incredible Backup script, /root/incrediblebackup-dav, will automatically make a connection to your new WebDAV server in the Cloud once you’ve entered your admin credentials and the IP address of your WebDAV server. Do this by editing incrediblebackup-dav. Just plug in your admin password and the IP address of your WebDAV server in the Cloud. Then save the file.

    In case you’re curious, here is the command to access WebDAV as a file system from your local server. Assuming admin:passwd555 were your remote Apache credentials and 23.45.67.89 was the IP address of your CloudAtCost server, the mount command would look like this:

    echo passwd555 | mount.davfs http://23.45.67.89/dav /dav -o username=admin
    

    All of the /dav files on the WebDAV server in the Cloud then would be accessible in the /dav directory on your local server until the WebDAV connection was closed/unmounted. You can add, edit, and delete files and directories. All of your local changes will automatically be synchronized with your WebDAV server in the Cloud.

    To close the WebDAV connection, issue the following command:

    umount.davfs /dav
    

    Making a Backup to Your WebDAV Server in the Cloud. This is the easy part. Once everything is in place and you have configured the Incredible Backup script with your admin credentials and WebDAV server’s IP address, you’re ready to kick off a backup. Just issue the following command while logged into your server as root:

    /root/incrediblebackup-dav
    

    Restoring a Backup from Your WebDAV Server in the Cloud. There are two ways to do this. If your local server and Cloud-based server are running identical versions of Incredible PBX, then you can restore the backup image to your Cloud server and run Incredible PBX in the Cloud. Simply move the desired backup file from /var/www/html/dav on the Cloud server to /backup and then run incrediblerestore from the /root folder. Once the restore completes, reboot your Cloud server, reconfigure the IP addresses of your phones, and you’re back in business.

    If you’d prefer to restore a backup from the Cloud to a local server, then you would first build a new server to match the one from which the backup was originally made. Next, configure the new server to support WebDAV access to your Cloud-based server following the tutorial above. Then execute the following commands after logging into your local server as root. Use the credentials, IP address, and actual backup filename saved on your Cloud server:

    mkdir /backup
    cd /root
    echo passwd555 | mount.davfs http://23.45.67.89/dav /dav -o username=admin
    cp /dav/backupfilename.tar.gz /backup/.
    umount.davfs /dav
    ./incrediblerestore /backup/backupfilename.tar.gz
    rm /backup/backupfilename.tar.gz
    

    WebDAV Cautionary Notes and Gotchas. First, WebDAV does a lot of heavy lifting under the covers because its intended for use as a collaboration tool by multiple people accessing and updating the same resources. So synchronization is important. When we’re moving huge files from a local server to the WebDAV cloud, this synchronization activity can give the appearance that your server has hung either during the backup procedure or thereafter. It hasn’t. So, after you run the Incredible Backup script to upload a new backup image, leave your server alone for a while. On your local server, don’t attempt to list /dav or otherwise use it for about an hour to be safe. On a Raspberry Pi, just be patient while the backup procedure completes. After that, you should be good to go. Depending upon the Linux flavor of your local server, the Incredible Backup script may not dismount your WebDAV resource successfully. You can do this manually LATER although it won’t hurt anything to leave the connection in place. As noted above, the dismount command is umount.davfs /dav.

    Second, be very careful in configuring Incredible Backup to make certain that you specify the correct IP address for your WebDAV server in the Cloud. WebDAV will try to connect to any IP address, and you don’t want to inadvertently upload your backup files to someone else’s server. Third, ALWAYS use a web browser to access your WebDAV server in the Cloud after your backup completes to make certain that a backup with the current date and time is shown in the directory listing. Particularly with RedHat OS flavors, it may take some time for the entire tarball upload to complete even though the script will indicate it has finished. Again, patience is a virtue. Don’t reboot. Things will get sorted out in due course.

    Finally, as with other network connections, if the WebDAV connection fails for some reason, your backup would be stored locally in the /dav folder rather than on WebDAV in the Cloud. That’s obviously not too helpful in the event of a local disk crash. So don’t forget to check your WebDAV server in the Cloud to verify successful completion of the backup.

    Enjoy!

    Republished: Monday, April 25, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


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

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

  • Some Recent Nerd Vittles Articles of Interest…

    1. The lifetime promise is, of course, in the eye of the beholder. It may be your lifetime but, more than likely, it’s the lifetime of CloudAtCost. The two are not necessarily the same so plan accordingly. 🙂 []

    No Brainer: Free Cell Service, Free Texting, Free Data Plan + Free SIP Trunk



    Suppose we told you there was a cellular reseller in the United States that would give you 3,250 minutes of free calling every month with a free Sprint phone. And, to sweeten the pot, you could also use those minutes as a SIP trunk on any Asterisk® server to make 3,250 minutes of free calls in the United States every month. Let’s not stop there. Suppose the provider would also throw in 3,250 SMS messages as well as 3,250 megs of data each month so you could surf the web, read your emails, and watch movies on your new smartphone. Crazy, huh? Too good to be true? Suppose we told you our family has been using this service since February with crystal-clear calling, zero outages, and flawless texting and Internet service on four phones! Suppose we told you we were using these same four lines to provide free calling on four different Incredible PBX servers scattered across the United States.

    Well, folks, it’s all true and today only starting at 5 p.m. Eastern daylight time until midnight, it’s your lucky day! What’s the catch? There’s a one-time, $32.50 non-refundable deposit to cover overages in minutes, messages, and data. If you’d prefer to borrow a Sprint-compatible phone from the company, there’s a deposit on the loaner phone. The cost ranges from $30 to $140 which is refunded when you return the phone in good condition. Complete plan details are available here.



    We’ve had running discussions about RingPlus on both the PIAF Forum and DSL Reports for a couple months so you can read all the history and comments if you’re interested. Our bottom line goes like this. What if RingPlus goes out of business? What’s my Breakeven Date, i.e. the day on which I will recover my initial deposit on the phone service plus the cost of the phone versus the cost of comparable service with a competitor? Frankly, that’s all you should care about. And, for today’s deal, that works out to less than two months regardless of which other provider you choose. Any free service after that date is pure gravy. RingPlus may last an extra month, or it still may be going five years from now. Either way, you win. And we’ll be looking forward to your Nerd Vittles donation on June 24 when you reach your Breakeven Date. Just click Help the Nerdy in the upper right corner of our site. 😉

    If you really believe in CYA and need a new smartphone anyway, then trot down to your local Apple Store today and purchase an unlocked iPhone SE for $399. Be sure to specify the Sprint model. It can be used to sign up with RingPlus at 5 p.m. And, if RingPlus croaks, this Sprint-model phone still will work with AT&T, T-Mobile, Sprint, or any Sprint MVNO. You also have 14 days to return it for a full refund!

    Today’s To-Do List. So you want to take the plunge. Here’s how to get started. First, go to RingPlus.net and click the Sign Up button promptly at 5 p.m. This plan is advertised on SlickDeals so the loaner phones will go quickly. Click Select Plan under the Elevator free plan description. When the Sign Up page appears, click Purchase a Phone at the RingPlus Store. Click on the Phones tab and choose Loaner Phones. If you prefer, you can also purchase a smartphone from about two dozen choices with prices starting at $65. Choose your phone and click Add to Cart. Then complete the rest of the checkout procedure to order your phone. When your phone arrives in a couple of weeks, you’ll receive instructions to sign up for the Elevator Free Plan. Our previous article on RingPlus will guide you through the rest of the activation process. Enjoy!

    Originally published: Sunday, April 24, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


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

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

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