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How to set up a "asterisk" phone system with an Obi ATA
#1
I thought about putting a PBX style system phone system in my house for years. I put if off however because it was either really expensive to go down the PBX route, or very technical in nature to go down the Asterisk route. However, after replacing my normal telephone service with Google Voice and an OBI ATA telephone device, I felt like it was time to re-explore the idea of an asterisk server. So began my search on how to install my own system....

Why a digital phone system? What are the benefits?
There are a lot of benefits to running a digital phone system. I will just highlight a few that we really like.
- CNAM lookup with Google Voice - GV doesn't provide full caller ID information, so we only got phone numbers for calls that were not in our Google contact list
- Individual Voice Mail boxes - every person in the family can have their own VM. The system can also email you a notification when a message is left and even include a audio file attachment of the actual message. With individual mail boxes, you only get your messages!
- "Follow Me" Options - you can have the system forward unanswered calls to other phone numbers, even outside numbers like your cell phone. You'll never have to miss an important call again.
- Cheap or Free VOIP Providers - most of these providers are available with devices like the Obi, but there are some that will not work with Obi. Plus it is easier to manage several providers via your call routes (more on that later).
- World-wide Phone Numbers - have family out of state or even out of the country? You can get DIDs (phone numbers) that are local numbers for those family members so they can call you direct without paying for long distance or international rates. Costs obviously vary with each providers, but generally only run a couple of bucks per phone number per month.
- Intercom Features - we really like this feature in our house. All the SIP phones I have can be set up to auto-answer internal intercom calls. So communicating with someone in the house is now as simple as pressing the intercom button and speaking. Because the phones auto-answer with the speaker phone, you have an immediate two way communication line set up without the other person having to answer a ringing phone. It's great for calling the kids to dinner, or asking your spouse a question, etc, etc.
- Potential to use the phone system with CQC - while I haven't set up anything in this way yet, there are some promising developments that may lead to CQC integration. First is the advent of Multicast RTP streaming. With the correct SIP phones, this will likely be an easy way for CQC to provide announcements through the phone system's intercom feature. Secondly, there are several SIP phone models/manufactures that allow scripts to be loaded on to their phone. In the hands of the correct programmer, this may lead to the ability to control features of your system right from the phones's buttons.

Deciding which digital phone system to use
The biggest problem with running an asterisk server is that the core program is run via command lines. There is no GUI built into the base asterisk program. Luckily, there are several GUIs out there that have been created to run along side the asterisk core which provide a much easier way to program and interact with the asterisk server. Without these GUI options, I probably wouldn't have attempted to do this.

I decided to use the PBX in a Flash (also called PIAF) phone system in my house. PIAF actually is made up of three main parts wrapped into a single installation method. First there is the asterisk core. Second is a GUI interface to help manage the asterisk server. PIAF uses FreePBX (which is available as a stand alone product) as this GUI. Finally, PIAF adds some security features and other addons.

There are other similar solutions out there, and initially I started out installing FreePBX alone (which I had installed for one day before researching some more and deciding to start over and install PIAF). Keep in mind that Asterisk is also strictly a Linux based solution. There are several other options for the Windows operating system, but they are not covered in this How To. However, you can also run Asterisk in a Virtual environment on a Windows machine, and this technique will be discussed some in this guide.

Resources that I used
First, I didn't come up with any of this stuff myself. I do not have the technical ability at this point to do any programming or script writing. The good news is that I haven't needed it so far, but I have relied on some excellent sources of information. I'll link to specific helpful pages throughout this guide, but these are a couple of the sites that helped me the most.

The most helpful site for beginners is the Nerd Vittles blog. It provides all sorts of PIAF information including walk through directions and general overviews of the software and addons. The PBX in a Flash forum is also a helpful resource, but honestly it is geared towards specific questions or problems. For the general overview, be sure to browse Nerd Vittles. I also used the Obi Talk forum when I needed information regarding the Obi products. But with PIAF-Green, the Obi products are easier to integrate (more on that later), so you may not even need to refer to that forum.

Equipment and software used
1) You need a computer to run this on. Asterisk and the available GUIs that we are going to discuss in this thread are all Linux based. You can either run it on a dedicated computer (even something as small as a $35 Raspberry Pi) or create a virtual environment on an existing system and run everything from there.
2) You need the Asterisk/GUI software. Again, I decided to use PBX In A Flash which includes the Asterisk backbone, the FreePBX GUI interface, and most if not all of the common Asterisk addons all in a single installable package. This makes getting started extremely easy, even for the greenest users.
3) You need phones. I still use my analog phones by using an Obi 202. However, using a SIP phone will allow you to really take advantage of the neat things available with Asterisk. I plan on slowly replacing my analog phones with SIP phones as I can. At this point, I have converted about half of my phones to SIP phones. Take a look at these posts where I talk about some of the phones I've used.

Deciding which PIAF version to use
The short answer is to use PIAF-Green. The long answer is there are several versions of Asterisk being developed right now - there are three current versions, plus many older versions. The current versions are Asterisk 1.8, Asterisk 10 and Asterisk 11. Asterisk 10 is not going to be supported as long at Asterisk 1.8 or 11. Here is a great explaination regarding the different version and support life expectancy. I think the only logical choice at this point in time is Asterisk 11. While is is officially a "beta" product, it is extremely stable and definitely a "daily driver", especially in a home environment. When you install PIAF, you will have to choose which version you want to use. The version with Asterisk 1.8 is called PIAF-Purple, the version with Asterisk 10 is called PIAF-Red, and my recommended version with Asterisk 11 is called PIAF-Green.

Iinstalling the Software
The PIAF images make it really easy to install everything. They include everything you need from the Linux OS, to the Asterisk software, the GUI, and the add-ons. You start with a blank hard drive and you'll end up with a fully installed digital phone software system that is ready for your setting to be entered to make it all work.

There are three major types of images depending on what you are installing PIAF on. There is a these instructions from Nerd Vittlesbasic image[/url] for installing in on a dedicated computer, there is a Raspberry Pi image for use with that hardware, and there is a Virtual Image option when you want to create a virtual environment and run it inside that.

Pick whichever image is applicable for the hardware you are going to be installing the system on. Each of those links takes you to a Nerd Vittles blog post that goes into explicit detail about how to install the basic software on your hardware. I'm not going to recreate the wheel here because Nerd Vittles does such a great job with the installation method. If you run into any problems, feel free to post them here, or go right to the PIAF forums.

What I AM going to discuss is how to take the initial installation and make it into a working phone system. Installing the software is the easy part, it's getting everything working that take a little time. Please keep in mind that the instructions to follow will work for PIAF-Green. If you are coming to this thread and we are now on PIAF-Orange or Asterisk 16, these instructions will obviously be out of date!
Brian

"Really dear, it was too good of a deal to pass up. Besides, look at what it does now...."
I think my wife is getting a little tired of hearing this :-)
Reply
#2
IncrediblePBX11 and IncredibleFax - two optional addons
I would suggest that you now take the time to install IncrediblePBX11 on your system. This is optional, but it installs a bunch of scripts and apps that help make setting up your system easier and more secure. It is very easy too. Just type in these commands on the computer you just installed PIAF on.
Code:
cd /root
wget http://incrediblepbx.com/incrediblepbx11.gz
gunzip incrediblepbx11.gz
chmod +x incrediblepbx11
./incrediblepbx11

Incredible Fax is an other program that you can install to allow your system to fax. Please keep in mind that faxing across most VOIP providers is problematic unless they provide a specific fax service or you use a POTs line. To install IncredibleFax, you must first install IncrediblePBX11 per the instructions above. Then type in this single command to install IncredibleFax...
Code:
/root/incrediblefax11.sh


That is all that is required to install everything you need. Next we will get into setting up your Asterisk server. This includes setting up trunks and extensions and routes. While this tends to be different with each service (for trunks) or each device (for extensions), I'll go through my set up to give you an idea of how it all works. There is plenty of information on line that will help you set things up for the services/devices you end up with.

Logging into the System
Open an internet browser from a computer other than your PIAF server. You should see a screen that looks like this.
[Image: PIAFUserLongin_zps21dc0862.png]

If you click on the "User" button in the lower left corner of the screen, it will turn the "Administrative" account on and add a few more options to the screen.
[Image: PIAFLogin_zpsf3d86dcb.jpg]

To get into the main GUI, click on the "PBX Administration" icon. A login popup will appear and you simply need to enter your MAINT password that you setup during the installation process.

OK - now it is time to get started with setting up the system! I find that it is easier to start with the basic building blocks, extensions, and work up from there. Let's get started...

Extensions
Extensions represent the connection between the Asterisk server and a phone or SIP device (which could be a SIP phone, or a computer or smart phone running SIP software). Think of these as internal connections to each phone (although it is possible to set up an extension to a device that is not local). If you have a star network (which most modern data networks are), then each device can have it's own extension. Unfortunately, most older POTS telephone systems used a ring network where the phone line is run from device to device to device. In this situation, all the phones on the ring network will be on the same extension. This will limit your ability to use some neat features like being able to intercom or transfer a call to a specific analog phone. For example, because all the analog phones work as one extension, forwarding a call to the analog phone extension will cause ALL the analog phones to ring. This isn't the end of the world, but having SIP phones will make your system more flexible vs using analog phones on a ring network.

Extensions are also the basic building block of the system which is why we will start with them. Without any extensions set up, incoming calls will have no where to go and it will be impossible to make outgoing calls too. After logging into the Administration GUI, click on the "Applications" menu and select "Extensions"

That will bring up a screen similar to this...
[Image: Extensions1_zps6ec25911.png]

We will primarily use the "Generic Extensions" for phones and "None" for virtual extensions like Voice Mail. The other options are more applicable for more advanced scenarios. For our first extension, choose "Generic Extensions" which will bring up a new page with lots of options to possibly fill out. Don't panic, it's not as bad as it looks!

Here is the top portion of the page.
[Image: Extensions2_zps83881f3b.png]
First you have to fill in the "User Extension" with the number of the extension you want to create (ie 201 for extension 201). Then fill in the "Display Name" which is the description of the extension you are creating (ie "Den Phone"). Most of the other options on this top section can be left as is. I do change the "Internal Auto Answer" option from "Disable" to "Intercom" because we use the system's intercom feature and want the extensions to automatically answer and provide two way audio communication when dialed. We'll still need to change a similar option on the phone itself, but that will happen latter.

The middle part of the screen looks like this.
[Image: Extensions3_zps5d8106e3.png]
Again most of the options here don't need to be changed. There is only really one important option which is assigning the extension "Secret" (or password). The system assigns a random long password automatically for you. You can change this to something a little easier to remember, but it is important that this password be very secure. You only need to enter it into the phone once during setup, so I would probably tell you to use a random password that won't be easily guessed.

The only other setting that you might want to change is the "Recording Options" With PIAF, you can set to record phone conversations "Never", "On Demand" or "Always". The system comes set up for "On Demand" with one exception - The "On Demand Recording" option is set to "Disable". All the other options above that are set to "Don't Care" which would allow on demand recording if you change the "On Demand Recording" option to Enable. Please check with your local laws regarding recording private conversations before enabling this feature. Some places require both parties to be aware of the recording while other locations only require one side of the party to be aware of the recordings. In virtually every location, it is against the law to record conversations when both parties are unaware of the recording (ie you record your kids phone conversations without their knowledge). Of course even if you enable the "On Demand Recording" will only occur when a feature code is entered into the phone.

The bottom of the page looks like this and deals mainly with the Voice Mail system....
[Image: Extensions4_zpsae2156ea.png]
Now in a normal office environment, you would probably want to turn the voice mail on each extension because an office worker is usually tied to a specific extension - the phone in their office. But in a residential setting, I don't recommend using voice mail on your actual phone extension and I leave those options turned off on any physical phone extension. Instead, I set up an virtual extension for each person in the house that needs voicemail. We'll do that next..... But first I should note that there are times when you will need to change some of these default settings to get a particular phone to work. But this is specific to the needs of each phone and honestly most modern SIP phones will use the default settings. You can check out the phone section of the PIAF forums for details on a specific phone if you run into trouble.

Voice Mail
Although you don't need to set up voice mail right away, I'll walk through it since it is handled in the extension setup. Create another extension, but this time choose "None (Virtual Exten)" for the type of extension. Like a real extension, you will need to choose a "User Extension" (ie the extension number - choose something well outside your physical extension numbers - like 10000) and the "Display Name" (ie a descriptive name for the extension - like "Brian VoiceMail").

Everything else on the top and middle of the page above can be ignored or left in the default setting. It's only the voice mail settings found on the bottom of the page that we care about. This time we want to "Enable" Voicemail on this extension.
[Image: Extensions5_zpsc7e8b5d5.png]

Obviously you need to pick a voice mail password - something that a user can input using a phone's dial pad. Then you can provide the user's email address and have email notifications sent to them anytime a new voicemail is left. If you select the "Email Attachment", the system will attach an audio file of the actual message to the email. So users can listen to the message right from the email account and not have to log into the system's voicemail system. If you choose this option, then I recommend that you also select the "Delete Voicemail" option. This prevents users from having to log into the system to delete message that they have already received and listened to via the email system.
Brian

"Really dear, it was too good of a deal to pass up. Besides, look at what it does now...."
I think my wife is getting a little tired of hearing this :-)
Reply
#3
Voice Mail Continued
The second section deals with the "VmLocator" option. If enabled, this option will allow users to press a number (0, 1, 2, etc) and be connected somewhere else. This could be another extension, but it can also be an outside phone number. I have my system set up with my cell phone number (10 digit format) in the Press 1 slot. This allows users in my mail box to press 1 and be connected to my cell phone. There are a couple of notes I need to make regarding this. First, the caller ID information coming into the cell phone will be the phone number of the trunk being used by the system and NOT the original caller's information. Therefore it looks like someone from home is calling me. Also, the cell phone's voice mail system will answer any unanswered calls and the user will not leave a message on the PIAF system. It is possible to get the phone call to be sent back to the PIAF system before the cell phone's VM answers, but it is more advanced than I want to go into with this How To.

It is also important to let people know that this option exists. So be sure to include the available options in your voice mail greeting . Something like "Leave me a message, or press 1 to be connected to my cell phone". You need to choose which calls will be allowed to use this option (Unavailable or Busy). But given that this is a virtual extension, I don't think the extension is ever "Busy". That being said, I usually select both options just to make sure everything works as expected.

Finally, you need to record your Voice Mail greetings for the system to work properly. To do this, dial your voice mail extension number from any extension on the system. The system will dump you into the voice mail system and start allowing you to record a message. We want to get into the Voicemail administration section and pressing "*" button and entering your password will get you into the admin section. Pressing "0" will get you to the greetings where you can record your "Busy" and "Unavailable" greetings as well as your name. Do all three to ensure the system works as expected.

IVRs
IVR stands for Interactive Voice Response. These are menus where callers can select different options by pressing numbers. It's typically the first thing you are greeted by when you call a large company now. Obviously most people hate them, but they serve a purpose, even in a residential setting. I use an IVR as the front part of my voicemail system. If a call goes unanswered, it actually gets sent to a IVR that allows a user to select who's voice mail box they want to be sent to. To set this up, select the "IVR" option found under the "Application" menu at the top of the GUI. You will be taken to the IVR page where you need to select "Add a New IVR" which will lead you to this page....

[Image: IVR_zps812ac576.png]

As usual, select a name for your IVR and a short description of what it is used for. Currently you will need to leave the "Announcement" on none, but we will change this once we actually record the IVR. You can set a destination for any invalid destination (ie a button pressed that isn't one of the IVR options) or if the system times out waiting for a response. In my residential environment I just throw the call back into the same IVR (creating a loop). You can always send them somewhere else, or even disconnect the call. Then you need to create the different IVR button options and their destinations. For my voicemail IVR, I have option 1 set to go to my wife's VM extension and option 2 to go to my VM extension.

So that gets the underlying code in place to handle the IVR, but we still need to record a set of directions to play to a caller so they know what the options are. To do this, we are going to go to the "Administration" tab and choose "System Recordings" which will take you to this screen.

[Image: SystemRecordings_zpse59e9d61.png]

To record a new message, type in the extension number of the phone you want to use to create the recording and press "Go". You will get a message that says, "Using your phone, dial *77 ?and speak the message you wish to record. Press # when finished." Do just what it says. Pick up the extension you said you would use and press "*77". The system will immediately start recording everything you say. My VM message goes like this, "We are unable to take your call now. Please press 1 to leave a message for Brett, and press 2 to leave a message for Brian." Press "#" when you are done recording. You'll have a chance to review the recording and rerecord it if you need to. Once you are happy with it, accept the recording and then hang up. Go back to the GUI and complete step three which is naming your recording. You cannot have any spaces in the name, so I use the underscrore a lot to break words up. A short, but descriptive name works best.

Once the recording is made, go back into the IVR page and choose that recording under the "Announcement" option. Now when calls are directed into the IVR, they will hear the message you recorded and have the option of pressing what selection they want.

Ring Groups
Ring groups are a set of extensions that you want to ring together. In a residential setting, most people will probably have one ring group that has every extension in the house on it. All incoming calls can be directed to this ring group so that all the phones in the house will ring when a call comes in. Go to the "Applications" tab and select "Ring Groups" to be taken to a page like this.
[Image: RingGroup_zpsb39c7d85.png]

You can choose a different ring group number if you need to for some reason (first one will default to 600) and you need to name it in the "Group Description". In the "Extension List" you want to add the extension numbers of all the phones you want to ring when a call is placed in this group. Then at the bottom you want to select the destination for any call that goes unanswered. For my main ring group, I have unanswered calls go to my IVR I created for callers to choose which voice mail box they want to go to (ie the IVR you just set up).

Trunks
Trunks are the gateway between the system and your service provider - basically the phone lines coming into the system. In other words, this is your connection to the world. Although this is done over the internet rather than a copper phone line, you still need a service provider that will handle this for you. Google Voice can do this for you and/or you can get service with any of the number of VOIP providers. Personally I use both Google Voice and I also have incoming and outgoing service from Call Centric. Without a service provider, your server will not be able to connect to the outside world (although you could communicate internally between the different extensions). How you set up a trunk is very specific for the provider that you use. Luckily, most providers have very specific how to instructions available on their website. Therefore I'm not going to go into detail on how to set up a every trunk, but I will talk about how to set up Google Voice to work with the PIAF system.

Google Voice
It was much harder on previous versions of PIAF to use Google Voice as a trunk. In fact, I actually used an Obi device to provide this GV connection prior to upgrading to PIAF-Green. The good news is that if you installed PIAF-Green as suggested, getting Google Voice to work is actually very easy now. Under the "Connectivity" menu choice, select "Google Voice (Motif). That should bring up this screen....

[Image: GoogleVoice_zpse12a5836.png]

Type in your Google Voice Username (with or without the @gmail.com ending) and password. Then put in the 10 digital GV phone number. Check the "Add Trunk" option as well as the "Add Outgoing Route" (if you are going to use GV for outgoing calls). I leave the "Send Unanswered Calls to GV" unchecked since I want to use the VM built into PIAF. Press the submit button and the Process button. If everything was entered corractly, you should see a green connected status.

[Image: GoogleVoice2_zpscb119c7a.png]

You will also want to make sure your Google Voice account is set up properly to work with PIAF. Log into your GV In Box (the actual Google system In Box) and choose settings. Normally you would have calls being forwarded to a phone number. But in this case, you want to uncheck any forwarding phone numbers and only leave the "Forward Calls to Google Chat" option selected. The PIAF system uses the chat functionality to connect these calls.
Brian

"Really dear, it was too good of a deal to pass up. Besides, look at what it does now...."
I think my wife is getting a little tired of hearing this :-)
Reply
#4
Routes
Routes are a little hard to grasp at first. Basically they are connections between your system and the trunks (service providers). In our basic residential setting, we may only have one incoming route and one or two outgoing routes. An Outgoing Route will decide what trunk to use for making calls based on how it is set up. An Incoming Route decides how to handle an incoming call based on how it is set up.

Let me give some examples of routes and hopefully it will make things a little clearer..... You might want all calls to the US and Canada to go through GV since it is free. You might want calls going to international numbers to use another service provider because the cost is cheaper than GV. You would set up an outgoing trunk that would direct all US and Canadian calls to GV, and another outgoing route that would send all international calls to another service provider. But it can get even more complicated than that. Let's same that company XWY has the cheapest call to Mexico, but company ABC has cheaper rates to all other international calls. You could set up another outgoing trunk to send all calls to Mexico through company XYZ, etc, etc, etc.

For incoming routes, most homes will just have a single route that handles all calls. But if you have several DIDs or phone numbers coming into the house (perhaps one for business calls, one for your teenager, etc, etc), then you can have routes set up that specify that all the business calls go to the office extension only and your teenager's calls go to their room extension only, and all other calls go to the ring group that rings all extensions.

The incoming route is easier because most people only need one route that handles all calls. I'm going to make this assumption in this example. Select "Connectivity" and "Incoming Routes"

[Image: IncomingRoute_zps78aa28f0.png]

Give the route a name. For a generic "catch all" route, leave the DID number and caller ID information blank. The DID number is the number calls are coming into from your provider. If you have a business phone line, you might want to put that business phone DID in that field to split those calls off to another destination. The Caller ID number is the phone number of the person calling you. If you want all calls from your mother-in-law to go to a special destination, you can put her phone number there. Moving on, you can select CID lookups and finally select where you want calls to go. In this case they go to my "ring all extensions" ring group. See that was easy.

Outgoing Routes
Outgoing routes are a little more complicated to set up. They are also extremely important to set up properly, or your calls may fail. If you ever find that some calls seem to work but other calls fail, take a look at your outgoing routes because something has been set up wrong.

To set up an Outgoing Route - select "Connectivity" and the "Outgoing Routes"

Here is an example of my Google Voice Outgoing call route....
[Image: b587a25e-e8a2-4efb-a0e0-d830f507db46_zps21c9a43d.jpg]

The most important and most confusing thing you have to set up are the Dial Patterns. In the above example, I have three dial patterns set up. The first one is for any eleven digit number dialed (1+10 digit number). The second pattern is for a 10 digit number dialed. It's good practice to assume that your provider needs the preceding "1" to be dialed along with the normal 10 digit number. By placing a "1" in the prepend column, the system will add a "1" in front of any 10 digit number. Likewise the third example prepends a "1" and the area code I live in to a normal 7 digit number. People are use to dialing just seven digits for all local numbers, but your provider cannot handle 7 digit numbers. So without this third dial pattern, any number dialed with just 7 digits would fail while long distance numbers would go through. (See why I said this is the hard part!). At the bottom, you select the trunk you want to use with any call that meets these dial patterns. In this case, it is set to use the Google Voice Trunk that I set up. FYI - "X" matches any number from 0-9 and "N" matches any number from 1-9. Because I've used the "X" and "N" wildcards, this route will send any call with 7, 10, or 11 digits to the Google Voice trunk.

I have two glaring missing dial patterns. Do you know what they are?

First I'm missing an entry for 911. If this was my only outgoing route, all calls to 911 (or 411, or any other shortened number) would fail because that number doesn't match the 7, 10, or 11 digit dial patterns I have defined! That's not good at all and it shows the importance of setting up these outgoing call routes very carefully. In my case I left out the 911 dial pattern on purpose because Google Voice will not handle 911 calls properly. Instead I have another service provider (Callcentric in case anyone cares) and subscribe to their E911 service (which means they electronically provide my correct address to the 911 operator). I have another outgoing route that only has one dial pattern, 911, and it is set to use my callcentric trunk.

The second missing entry is any/all international dial patterns. We don't have a need to make international calls, so I left them out intentionally. If anyone tied to call a number with an international prefix, that call wouldn't go out because there is no route set up to handle those dial patterns. If you need to make international calls, then you will likely have at least one more outgoing call route with the international dial patterns in it sending those calls to the cheapest provider you have.

Obi Device
I haven't talked about the Obi device yet. Prior to upgrading to PIAF-Green, it actually acted as my Google Voice trunk. All incoming calls came into the Obi and then were sent to the PIAF system. Outgoing calls went from the PIAF to the Obi device. It was actually very complicated and definitely the hardest part of the entire system to set up. However, all that was greatly simplified when I upgraded to PIAF-Green. Now that Google Voice support is built into PIAF-Green, I let the PIAF system handle it all.

Now my Obi device is simply defined as a extension of my PIAF system. All my analog phones are connected to that one extension. So if I dial that extension number from one of my SIP phones, all the analog phones will ring. But the setup is no harder than any other SIP device. The only thing to make sure is that when you set up the Obi side of things, you will enter the extension number and password. But you also want to make sure you check the "All Income and Outgoing Calls Use this extension". Keep in mind this is on the Obi side of things and the Obi can have several accounts set up. All this means is that all connections to/from the analog phones will go through the PIAF extension. What was the hardest portion of the setup has now become one of the easiest!

Here is a great Nerd Vittles article about the benefits of an Obi202 which is the device I am using. Having the option to connect a cell phone via bluetooth is pretty cool. They talk about some of the other features available as well as how to set it up in the PIAF system.

Setting up multiple lines or DIDs
Learn how to set up multiple lines or DIDs in your system here.

The End
Hopefully this How To has helped you understand the overall PIAF system. As you can see it is very customizable, but that doesn't mean it has to be complicated. By following the steps outlined in this thread, you can set up a working system in short period of time.

I obviously wasn't able to go through every scenario or get into some of the more advanced options. I highly recommend that you visit the PIAF Forum and read as much as you can. Feel free to ask questions here, or PM me, or ask on the PIAF forum. I am certainly a digital phone newbie. There are some real experts on the PIAF Forum. So tap into that knowledge!

Good luck!
Brian

"Really dear, it was too good of a deal to pass up. Besides, look at what it does now...."
I think my wife is getting a little tired of hearing this :-)
Reply
#5
Good write-up. I've never used PIAF but I've heard good things about it. I've used strait * and FreePBX and a few other GUIs that are now dead over the years. Maybe I can chime in on this thread and help out since I know a lot about * and SIP and almost nothing about HA.

I've just been researching stuff since Christmas to potentially make my house unlock when I pull up. I currently don't have a desire for * in my house, but maybe I'll want a button on my Polycom desk phone to unlock the front door and talk to the person at the door. Who knows. Smile
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#6
Asterisk Definitive Guide 4th Edition was just published today by O'Reilly in ebook format. With this discount coupon DSUG2 the book is 50% off... (bought one this morning)

O'Reilly's electronic book policy is extremely liberal!!!

-Ben
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#7
batwater Wrote:Asterisk Definitive Guide 4th Edition was just published today by O'Reilly in ebook format. With this discount coupon DSUG2 the book is 50% off... (bought one this morning)

O'Reilly's electronic book policy is extremely liberal!!!

-Ben

Did you get the download version and if so have you already received it? I'm just curious if it is worth getting if we already have an older version. I have one of the older versions that they offered for free at some point, but it is probably from the Asterisk 1.4 or 1.6 era.
Brian

"Really dear, it was too good of a deal to pass up. Besides, look at what it does now...."
I think my wife is getting a little tired of hearing this :-)
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#8
sic0048 Wrote:Did you get the download version and if so have you already received it? I'm just curious if it is worth getting if we already have an older version. I have one of the older versions that they offered for free at some point, but it is probably from the Asterisk 1.4 or 1.6 era.

I got the ebook version, hardcopy not est to be available until May. New one covers Asterisk 11.

-Ben
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#9
Darn it, you made me buy it! ;-)
Brian

"Really dear, it was too good of a deal to pass up. Besides, look at what it does now...."
I think my wife is getting a little tired of hearing this :-)
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#10
Feel like this was left as a season ending cliff hanger....I've got my pi (set up with Asterisk) and my Obi202, they just dont get along yet. I think it is all pretty cool and it is sweet that I can drop my "normal" $60/mth landline for a couple of bucks a month but think I may need some more technical help. No rush though, I will just keep hitting refresh on this screen until I see the next steps.
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