NEW: Help decide if FreePBX is right for my home network?

I am relatively new to FreePBX and am trying to decide if this is a good VoIP solution for my home network.
I have a dedicated HP server that runs vmware and can run any operating system including Linux/PBX.
I have a few Cisco 7975G Color IP Display Phones I am trying to make use of.

I am researching to see if FreePBX would be a good solution for my home network and wanted to ask for assistance in understanding how to set up outbound connection (trunks) to call out. This point is very critical to me including available providers. I do not have a landline (POTS) but have Vonage and Magic Jack VoIP services which for now i can test with, if possible. I found some good YouTube videos on this topic but they do not answer my questions in full. Here are my questions.

  1. What is the difference between FreePBX vs. Distro vs. Asterisk?

  2. How much disk space/CPU/Memory should be allocated for new FreePBX installation (10 users)?

  3. What is the stable FreePBX recommended version today?

  4. Should I get external service or can I use current Vonage and or Magic Jack devices for outgoing calls?

  5. How can I set up outbound routes using the above devices or what other options available out there?

  6. For firewall, what ports needs to be opened between the phones and PBX server and PBX server and Internet?

Thanks in advance and please advise.

  1. Asterisk is the PBX software (the part that operates the phones). FreePBX is the management software (which operates Asterisk). “Distro” is the prepackaged software that installs everything you need for FreePBX.

  2. I always presume 320M. If you aren’t saving voice recordings, you can get by with a lot less, as long as you do your housekeeping (getting rid of old log files, etc.)

  3. I’ve got an older FreePBX system that hasn’t rebooted in over 800 days. There have been a ton of improvement in both stability and performance since that system, so I’m pretty sure you’ll be fine.

  4. I’d get a “per minute” service that doesn’t charge a lot for your incoming numbers. I use a couple of different services for outbound and pay less than a couple of cents a minute. Even with my inbound (with a number) connection, my phone service costs less than $5 a month. Magic Jack and Vonage both have restrictions on what you can attach to their systems, so between the native cost of these services and the hassles using them are going to present, I’d disrecommend both of them.

  5. Setting up outbound routes using the hardware devices that connect to the proprietary systems is possible, but you will need additional (FXS/FXO) hardware and (DAHDI) software. I personally do not recommend this as it adds an extra level of complexity and allows for deflection by service providers (too many suppliers).

  6. The “best practice” for setting up a system is to put it behind a firewall and open specific ports (documented in the Wiki and lots of other places).

  7. Setting up the Cisco phones can be challenging. The SIP firmware loads aren’t as flexible as the SCCP loads, but installing the SCCP (Skinny) interface modules can be challenging (it requires a source install of Asterisk, but the driver website has the instructions on how to do that). I’ve been working on a FreePBX management module for the Chan-SCCP-B driver, but the going is slow (paid work almost always takes priority). In spite of that, setting up the SCCP phones aren’t really that hard to do by hand, so if you’re interesting in that, you can do it. If you’d rather go the SIP route, there are literally hundreds of websites out there that talk about how to do it.

Thank you for your responses and thank you for responding in plain English instead of providing a link to lengthy articles.

I still have a few follow up questions requiring your assistance.

  1. Which version to choose and what’s the difference?
    The versions available that are stable in 64 bit are
    A. STABLE – 10.13.66 with Linux 6.6 Asterisk 11 or 13
    B. STABLE – 6.12.65 with Linux 6.5 • Asterisk 11 or 13

  2. What “per minute” service do you use and how does it work in respect to FreePBX?
    Does it provide both outbound and inbound calls?

  3. How to set up inbound and outgoing calls with FreePBX using the above mentioned services?
    Does it require additional hardware (which?). If not, how can it be done on the software level?
    This is the part which I struggle to understand most.

Thanks in advance and please advise.

No good deed goes unpunished. :slight_smile:

  1. The current version of FreePBX is the one you download from FreePBX website. Unless you want to spend a lot of time “having freedom”, just install one of the FreePBX 13 Distros. You are more likely to get good results with newer software. The current FreePBX distro includes an optional “Edge” track that I don’t think you’re going to need.

2, I like Voip Innovations for both in and out. I also use CallWithUs and Alcazar Network for just outbound. You can also do worse than the VOIP service that the FreePBX folks sell (SIP Station) - helps support the project, etc. There’s a SIPStation module that will help you get everything set up if you go through them. Note that you will probably end up with a pre-paid service - I’m down with that.

  1. Setting up Trunks is handled through the GUI. Whatever service you go through will provide you with the basic information (registration strings if needed, IP Port addresses, that kind of stuff.) It’s not very challenging. Once you get the trunk set up, you’ll map your inbound and outbound routes to your trunks. Once again, through the GUI it isn’t really very challenging. You’ll probably want to look at the Wiki for FreePBX to get more details.

Speaking of which: almost all of this stuff is documented in the FreePBX Wiki. While I don’t mind answering questions, I feel like I’m filling out a proposal for you. Not a fan of people that aren’t willing to do a little work on their own.

This is actually a FreePBX 13 feature and not exclusive to the distro.

Go with 10.13.66 with Asterisk 13.

In the near future the next release will be out with FreePBX 14, EL 7 etc and the 6.12 track will drop off.

Thank you @cynjut I called the VoIP Innovations and they provided me with a comprehensive wiki but the forum would not allow me to share it here :frowning:

I guess the squeaky wheel gets the oil :slight_smile:

Thank you @jfinstrom I will go with: STABLE – 10.13.66 with Linux 6.6 Asterisk 13.

The VoIP Innovations has a one month free to try out their account which is perfect for practice.

Are there any free outbound only providers that exist today?

Is Google voice still an option today?

VoIP Innovations provide SIP service only; are Cisco 7975G Phones not set to SIP by default?

What does it take to become VoIP Innovations today, a land line to the vendor?

Thanks in advance.

Your questions are now getting confusing.

  1. I wouldn’t bother finding a ‘free’ service. Their business model is vague and I don’t see how they can do what they do. VOIP isn’t for hobbyists anymore - this is a well established commercial service space. The cheap outbound only (like CallWithUs, for example) don’t require a contract or a commitment - send them $25 and call out till it runs out of money.

  2. GV has some very specific terms of use in their EULA, and there is some debate on whether connecting an Asterisk system to it violates those. There are also some very specific (and technical) changes that need to be made to the base Asterisk installation. Recent changes to GV made it inaccessible for many Asterisk users, and we’ve had a lot of conversation about it on here. Honestly, I don’t consider it a task appropriate for a first-time, new user. Get some time under your belt and we’ll see if you think GV is still a good idea.

  3. VOIP Innovations is a trunk provider - they aren’t really good at doing individual phones. You want to connect a PBX server (like as Asterisk server with FreePBX, for example) to their system. The 7975G is (out of the box) an SCCP phone. As I mentioned earlier, there are two ways to get that working:

    a) Install an SCCP Channel Driver (like Chan-SCCP-B) and connect the phone to your local network. Set up is non-trivial, but is reasonably well documented (I wrote a lot of Wiki pages for Chan-SCCP-B and FreePBX). Once you do that, your 7975 will be usable on the network, through Asterisk, and out to your ITSP (VI or SipStation, for example).;

    b) Change the firmware in your phone from SCCP to SIP. For many people, this is easier to get set up, since it doesn’t have an many features or options. You can get a solid 90% of what the phone can do and not have to install any new channel drivers.

  4. To connect to VI, you will need an Ethernet connection (cable modem, etc.) and a static IP address since they use Host-Based (IP Address) authentication. I don’t know if they support DynDNS (which would allow you to use a connection that isn’t dedicated), This also means you will connect to them with Chan-SIP (and not PJ-SIP). As PJ-SIP matures, it will probably be able to do host-based authentication, but for now, you’ll need to use the older driver. The configuration for their connections is about as simple as you get.

“b) Change the firmware in your phone from SCCP to SIP. For many people, this is easier to get set up, since it doesn’t have an many features or options. You can get a solid 90% of what the phone can do and not have to install any new channel drivers.”

By “this is easier to get set up” going from SCCP to SIP on 7975G. Did you mean the below instructions?

ARTICLE : Configuring Cisco 7975 IP Phones for SIP

Anyone who has had the “pleasure” of trying to configure a Cisco 7975
phone with a SIP IP PBX system such as Asterisk, trixbox, or 3CX can
attest to the fact that the phones are very picky with their config
files and much of the information that is really handy to know is simply
not documented anywhere. On top of that, to get the phones to even work
right you have to use very specific versions of the SIP firmware as the
phones are not designed to work with third-party phone systems that
talk generic SIP messages. While I can tell you that Cisco recommends
using SIP firmware 8.3.2 SR1S, I cannot provide the firmware files
because of their licensing agreements. Assuming you can get the correct
firmware copied to your tftp server, and your DHCP server is serving out
Option 66 to tell the phones where to look for its config files (yes,
these are mandatory requirements), then we can get into how to actually
configure the phones themselves.

To start off with I have created a set of files that will help you get started, you can download the set here (download).
Inside the zip file are the following files:

SEP7975.cnf.xml – A clean example fileSEP0023331BEA0A.cnf.xml – Same as the previous but showing how the mac address is added to the fileDesktops\320x216x16\7975Logo.png – Background image for phoneDesktops\320x216x16\7975Logo-TN.png – Thumbnail imageDesktops\320x216x16\List.XML – Background image config file
Let’s assume your phone’s mac address is 0023331BEA0A, we will need a
file named SEP0023331BEA0A.cnf.xml for the phone to configure itself.

Using the included files as a guide, we need to edit the following

Line 27:

Change IP address to IP address of your PBX system. Using a host name tends to cause the phone to not parse the file properly.

Line 71: http://{TBexternalIPaddress}/xmlservices/authentication.php

Line 72: http://{TBexternalIPaddress}/xmlservices/PhoneDirectory.php

Line 75: http://{TBexternalIPaddress}/xmlservices/index.php

Line 78: http://{TBexternalIPaddress}/xmlservices/index.php

If you are using XML services you will need to change the above lines to the correct URLs.

Line 155: VoipStore

Now I know you don’t want to change this

Line 160: 402

Line 162: displayName>402

This is the text shown next to the line key, usually the extension number

Line 161: 001FCA368894

Line 163: 001FCA368894

Line 170: 001FCA368894

Lines 161, 163, and 170 are the SIP login, on some systems this is
the extension (trixbox CE, 3CX, FreePBX) and on others it is the mac
address (trixbox Pro).

Line 164:

The proxy IP address is the IP address of your IP PBX system.

Line 171: 9svHcd92t4H

The auth password is the SIP password (secret on some systems) for this extension

Line 174: 8555

This value is the number to dial to access the voicemail system. On
trixbox Pro this is 8555, on trixbox CE it is *97, for 3CX this is 999.
Lines 184 – 209 repeat the same information to add another active
line key, you can continue adding sections like this for all eight line

Adding the background image
If you copy the existing files from the zip file to the right
locations, you should be able to activate a new background image for
your phone. Press the Settings key, then 1 for User Preferences, then 2
for Background images, use the directional keypad to move to the image
you want to use, then press the Select softkey, then press the Save
softkey. Your new background image will now display in all its glory.

I cannot begin to stress how picky these phones are with their
configuration files. If you are making changes and rebooting the phone
and your changes do not seem to be taking effect, go to Settings,
Status, Status Messages to get a list of errors that the phone is
encountering. Typical ones to ignore are the following:
Error updating LocalNo CTL installedFile not found: CTLFile.tlv
Those are actually “normal”, what you do not want to see is “Unable
to Parse file”, that will tell you that it did not like something in
your config file. This can be something like using a host name when it
wants an IP address. Just understand that you will probably experience
quite a bit of frustration getting the 7975 to work properly, but with
enough persistence it actually is possible.

Yup. Lots of people think that’s easier than installing a new channel driver.

OK. Since there is a slight chance that the above Cisco 7975G configurations may not succeed, what are some other SIP phones that are considered to be great phones with flexibility and quality and will work with FreePBX :wink:

I mean I don’t even understand what the instructions mean by saying: “If you are using XML services you will need to change the above lines to the correct URLs.”

Honestly, almost any SIP phone is going to do what the average home phone user wants to go. The Sangoma phones (made by the people that own the FreePBX name) are pretty good and are supported well.

If you want to go for something cheap, just browse around on E-Bay. There are lots of super cheap phones that you can implement that will get you where you want to go.

If you’d rather go with a soft phone, there are lots of reasonable choices out there. For testing and experimentation, I actually recommend starting out with a couple of free soft phones. This minimizes your cash outlay while you are still experimenting.

Thanks. I will look into the Sangoma phones but as stated in my post I am not looking for cheap but great modern phones that provide the flexibility and work well with FreePBX. For example, I had older Polycom phones that are SIP but they did not work well with FreePBX. Please advise and if you could name some specific models, that would be greatly appreciated.

Any advice regarding the SIP phones?

Also would it be easier instead of dealing with the SCCP phones just get Cisco UC560 router and connect existing phones there? How is this solution for example different from having FreePBX? Isn’t it the same outcome? What are advantages and disadvantages in favor of FreePBX?

Thanks in advance…

Without a doubt the new Sangoma phones enjoy the tightest integration with FreePBX. If you’re up for buying new hardware and want “the best” then don’t bother looking elsewhere. Similarly, SIP Station would be the place to go for the tightest integration and easiest setup/maintenance in FreePBX.

BTW - you shouldn’t need to manually “open” any ports on your router unless you want to allow devices outside your network to register.

Thanks. I found Sangoma s300, 500 and 700 phones. Which do you recommend?

Also I found Sangoma FreePBX Phone System 100 model box.

What is it and if purchased, do I have to still install FreePBX on my server or does the box already have it?

What’s the difference between Sangoma 100 and my FreePBX server?

How does Sangoma FreePBX Phone system 100 connects to the SIP vendors?

Does nobody from Sangoma feel like answering this? :confused:

It has been a while since we discussed my Cisco 7975G IP phones. Since then Cisco routers like CISCO UC520-8U-4FXO-K9 Voip Gateway PoE or CISCO-UC520W-16U-4FXO-K9-UC520-Wireless-VoIP-Router or UC560 models are end of life and dramatically dropped in price. You can find them under $200 on eBay. These routers should support my Cisco 7975G IP phones so there would be no need for me to convert them to SIP or install drivers etc.

Although I realize there arent many fans here of Cisco equipment, the question is will the routers listed above work with VOIP Innovations via SIP trunking? If yes, then all my problems are solved.

Any assistance is greatly appreciated.

Thanks in advance and please kindly advise.

My first post - I actually came here looking for a solution to a problem, but found I could contribute!

The UC520 supports the 7975 phones just fine with a SIP trunk to the outside world. We have used Hostcomm, and I can verify the UC500 series can be made to work with this trunk.

Funnily enough, I am just decommisioning a system with 7975G handsets and a UC520, and migrating to a FreePBX based system.

I realize that the OP has probably already solved all of his problems, but I’ll chime in for those reading this thread about my experience

I wanted to play with this stuff for a while and got my hands on some used Cisco 7941 and 7961 phones for almost nothing (like $10 each). They’re great phones, work well and are quite cheap.

I run my FreePBX bistro on a Raspberry Pi that is dedicated to just running the PBX. The cost of that thing was like $20 and an extra $5 for the plastic case. It’s plugged into my wireless router and I use the router’s USB port to power it. You can squeeze everything from the “raspbx” distro onto a 4GB card, but if you want to do anything else having an 8GB card is a better solution.

Finally, I have set up my google voice accounts in FreePBX so that I can use Google voice as the primary line for my “home office”, on the Cisco 7961 with 2 outgoing lines, one for my GV and one for my wife’s GV number. If I wanted to add extra phones in the house I could do that as well, but we use our mobile phones for pretty much everything.

All in, it cost me about $50 to set up and I haven’t paid a phone bill for 2 years. I can dial any US number for free and have a reasonably good desk/speakerphone at my desk.

Overall pretty happy with the experience.