Best guides for getting FreePBX running locally

FreePBX Installation / Upgrade
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I have nearly infinite questions, zero answers, only 5 I can ask. Subtracting 5 from the infinite, I really only have a 5 questions circling around me, the rest will come as I go.

I wanted to spend more time looking for guides around in the forum but ended up leaving “FreePBX” guide search at the end of the day. Today was gonna be the day, approaching the relatively high priority task, prepare for Monday.

We’re a I.T. company trying to get into FreePBX / Asterisk to offer to our major clients, on a mission to implement it at two companies. One will be replacing a Nortel system, and at the other replacing a Ring Central VOIP system. I’ll throw in some numbers, on any given day both of these companies have ~100 inbound calls maxing 10 concurrently, outbound ~50 @ 5 concurrent; and a lot of inter-office mingling calling.

My question to you is,
How do I setup this **** thing. Darn. I have a server that’ll eventually be at one of the companies with overkill hardware, possibly best. Dell R520 with a Xeon processer, OS on SSD, 24GB of ram and a total of 6 Gigabit Nics.

Also a Sonicwall. I saw a post that I’ll follow, “Happiness with SonicWALL afterall”. I’m an absolute butcher with all things.

I followed a CrossTalk Solutions video, not going to nail the guy, he’s a good guy, good stuff. I followed along; with him installing and configuring FreePBX 14 and I on FreePBX 16.

The installation and interface doesn’t appear different. I know nothing basically.

got it activated, trial of SIPStation, created users, created extensions, used a softphone to call another - nothing, bought PolyCom VX301 - nada. I realize maybe my problem is with th

I rather start from the complete beginning of time. What posts have you used or seen that offered a rich amount of study content and setup information.

Assume I have zero knowledge of anything, as I do. Okay networking.

Is DMZ absolutely necessary for softphone and FreePBX to communicate? SonicWALL blocks UDP port 5060, does this issue affect my phones on lan or just FreePBX to SIP trunk provider?

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The wiki is pretty comprehensive, but some would think that building two systems in two days for hundreds of clients with zero experience is a bit optimistic , I would encourage you to get paid support.

(Maybe you will need to offer some weekend overtime)

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This coming from another commercial IT company Owner who wanted to incorporate a on-site PBX solution to our larger commercial clients a year ago…you aren’t doing it in 2 days… heavily experienced tech…sure but that’s a bunch of work, long hours to do what you want in 2 days

We tested multiple products and settled on FreePBX/PBXact after a month…we installed FreePBX, bought the commercial modules and many different phones to test and play with…We’ve been digging through and testing all aspects of FreePBX and it works great, also with Sipstation…became a Sangoma partner, went through the Sangoma University stuff which is excellent… and have done lots and lots of reading and testing on our test machine.

All the info you need is available to you if you actually put in the time and want to learn it. We have FreePBX and PBXact running at a few places problem free. Local and remote endpoints…Sangoma Connect, Zulu… if you know your networking well, know the basics of how to use Putty and FTP Client you will be fine. I’m no expert but I have a good handle on FreePBX and know how to troubleshoot any issues or roadblocks we’ve ran into.

If you want to sell it as an IT company I highly suggest setting up a full test environment and learning how to properly program, use and troubleshoot FreePBX/PBXact before selling and deploying at a client. Just my $0.02 after doing all the above over the last 12 months.

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Are you a Sangoma Partner? Have access to Sangoma University?? The PBXact webinar is excellent…step by step from unboxing to complete setup and programming…

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Mike here from Sangoma… My team has two goals- Goal #1 help grow the FreePBX userbase. #2 help partners grow their businesses with FreePBX. If you can see how these two are connected and would like to learn more, let me know. I’m here to help.

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I hope that you don’t mean cutover of two fairly large systems on Monday, 3/15/2021 – IMO that’s impossible, even with professional help.

What is your time frame for these systems?

I assume that the Nortel system is using proprietary phones that are incompatible with FreePBX and you’ll be replacing them. With what? Are the wall jacks RJ45? Is there Cat-5 from there to the equipment room? Or, will you be inserting the phones in an Ethernet connection intended for a PC?
What trunks do they have now (POTS, PRI, SIP from ISP)? Will they be keeping them or getting new SIP trunks? This stuff is likely more work than setting up the PBX.

IMO RingCentral is highly reliable, good support, overpriced, mediocre feature set. Are they switching just to save a few loonies, or to gain much better functionality? If the latter, implementing that may be much more work than setting up extensions and trunks.

Will you be using the existing phones? If so, they are likely ‘locked’ to RingCentral and depending on details may require cooperation of RC and/or the phone manufacturer to unlock them. Also, physical access is probably needed to provision them the first time.

Why do they (or you) want to switch from cloud-based to on-site? The cloud server is simpler, more robust and probably less expensive.

Do you mean the phones don’t register? If so, do the attempts show in the Asterisk log? What shows in the softphone’s log?

Or does the call complete but there is no audio? If so, in Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config, restart Asterisk, then reboot the phones and test.

Phones and PBX on the same LAN subnet and not using VLAN communicate without going through the firewall at all, unless for some strange reason they are connected via different physical ports on the firewall.

If you still have trouble, provide your observations in detail.

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Any which way, given his ‘cricket synthesis’ he better be prepared to drop $+4figures by Monday. :wink: (tick-tock)

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Work overtime on the weekend!!! ahhh! Honestly, maybe that’s going to have to happen. Last week I was supposed to have free time to get to it.

Does the order of operations matter when it comes to configuring?

If I remember correctly, when I used searched the Wiki I skipped “Configure E-Mail Settings”, all email related settings, and “Configuring a Phone Using EndPoint Manager (EPM)”, instead just went into “Setting Up SIPStation Trunks”.

Following the video instructions published by CrossTalk Solutions, we went from

  1. Installing FreePBX
  2. FreePBX Administrator Account
  3. Registering FreePBX (Activation)
  4. Sangoma Smart Firewall
  5. Network Interface Configuration
  6. General SIP settings
  7. Making legacy CHAN SIP default by changing its bind UDP port 5060
  8. PJ SIP to secondary by binding it to UDP port 5160
  9. Activating a SIP Station trial and converting to monthly.
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I’ll check out Sangoma University, “PBXact Esentials”. Thank you.

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Luckily not. I made my message too urgent did I? Possibly in a month, maybe a few weeks tact is when we’ll have want the FreePBX installed in our client’s new building. For now I’m asked to focus on learning and creating a test environment before we roll it out to anyone.

The Nortel system is installed in an old building of our Client, which eventually will be retro fitted. The client has a new building in development which my boss and I did the cabling job for. Planning the use of VoIP, using Cat5. Later in the implementation we’ll use SIP from the ISP. To keep things simple we’re sticking to SIP station.

I get no audio. The SIP settings on the phone point to the server and login as an extension user and voicemail as password. Wireshark hasn’t captured any interaction between the server and phone, or softphone. How do I confirm in the logs my phone tried registering to the System? ’

I’ll try to get back to you with detailed observations. I haven’t booted the server in awhile. Tomorrow It’ll be.

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…and voicemail as password…?

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There your problem…use the Ext password…not the VM password…

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I made it all the same. I’ll recreate an extension to be sure.

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Don’t reuse stuff like that, I suggest you learn good and safe behavior from the very start, bad habits are hard to unlearn.

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https://wiki.freepbx.org/display/PPS/FreePBX+Distro+First+Steps+After+Installation

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Yesterday after following along with the wiki, searching up settings, cross referencing with CrossTalk and LawrenceSystems 2hour long video, I can say I have a semi-functioning system!

Created Firewall rules (WAN to LAN) to allow port 5060-5061 (Chan_SIP) and 5160 (CHAN_PJSIP) to be accessible by the SIPStation Trunk gateways as a FQDN entry to the test environment. And vice versa for the test environment to SIPStation thru those SIP protocols above (seems unnecessary since its trusted inside of the network to access anything). With UDP timeout set to 120 seconds - it appears SIP trunks can traverse for awhile? I don’t know the reason why.

I was using the wrong password when I should have been using the secret string generated by creating a extension - while creating the user at the same time.

The SoftPhone (MicroSIP) I’m using registered. One of the two Polycom vxx301 is also registered and can dial to each other. Including Outbound calls!

SIPStation account key made it simple to setup the ordered trunk and DIDs.

The road bump I’m facing now is why is it when I do an outbound call - accept it on my personal phone that’s there’s no voice heard on either ends. To add to the question - when I call from my phone to my main DID it dials, and becomes silence and hangs-up. Are these firewall related issues? I see when I do an external connectivity test the firewall status reads fail.

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What’s a primary way to debugging issues like network connectivity, firewall, received calls?

I’ve come across: asterisks -rvvv
but it’s not truly all debugging or log messages in the system.

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In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set.

If you change these, after Submit and Apply Config, you must restart Asterisk.

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Can confirm Stewert, my External Address and Local Networks are correctly set.

I noticed in CHAN_SIP, NAT configuration > IP Configuration set to “Static IP”. My WAN IP isn’t static. What is Public IP setting? does this mean it updates the WAN IP. Otherwise our WAN IP is set correctly there.

Edit:
I realized just now (WAN-LAN) UDP ports from 10000-20000 weren’t included in Firewall rules. Just added, did another check - failed. Maybe Premature.

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Normally, you set Local Networks and External Address on the General SIP Settings tab and leave the chan_sip specific tab alone (it shows Public IP, even though it’s not).

Confirm that your hardware firewall is not rewriting the UDP source port for outbound RTP packets.

If you still have trouble, at the Asterisk command prompt type
pjsip set logger on
for a pjsip trunk, or
sip set debug on
for a chan_sip trunk.
Make a failing test call, paste the Asterisk log for the call (which will now include a SIP trace) at pastebin.freepbx.org and post the link here.