Hi complete newbie needing help

Hi, I was tasked with installing freepbx and configuring it to work with voxbeam. The ISO made short work of the install, but I have no ide about the other stuff. Can someone please help me? What steps are required to configure the Voxbeam stuff?

Frankly, a paid support ticket, a knowledgeable consultant, or a lot of reading.

So where should I start reading?

https://wiki.freepbx.org/display/FDT


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

I am a fairly satisfied Voxbeam user (inbound is flawless, but about 1% of outbound calls are mishandled, even on Platinum route). Of course, your experience could be much different, depending on which countries you call.

I suggest that you get outbound working first. The pjsip trunk settings are trivial: Registration None, Authentication None, SIP Server sbc.voxbeam.com, Port 5060.

Then, log into your Voxbeam account, go to Settings and add your public IPv4 address. Adjust other settings as desired.

To make a call, send your caller ID and called number starting with the country code. An initial + is accepted but not required.

If you have trouble, paste the Asterisk log at https://pastebin.freepbx.org and post the link here. It’s useful for the log to include a SIP trace. At the Asterisk command prompt, type
pjsip set logger on
then make your test call. The SIP traffic will appear in the Asterisk log, mixed in with the regular entries.

Thanks for that Stewart, installation and initial config was a no brainer. Having never used anything like this the outbound route and stuff is confusing, and no idea about Voxbeam. Thank you for your input.

Configured the Trunk and working on outbound now

Do you have recommendation for a softphone for Windows? Like I said this is my very first experience with any of this

For testing, any of the common free ones (PhonerLite, Zoiper, Z-Lite) should be fine. If you plan to use a softphone for everyday calling, one of these may have the features you want, or you can look at the paid apps.

The “user Control Panel” (UCP) has a softphone built in, (Works with even Windows :wink: )

Ah, thanks Dicko

Zulu, softphone? $199.99? thanks but I am still trying to get this all working for the first time ever.

No Zulu needed, (I can’t even install it on my systems :wink: ) just the WebRTC phone module and the correct permissions for the user.

How much for one of you guys to configure this instance?

Or cheaper

A couple of hours in

https://wiki.freepbx.org/

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The wiki is pretty good. I would recommend this link as it does a pretty good job of holding your hand right from the beginning when you have a fresh install.

https://wiki.freepbx.org/display/FPG/Configuring+Your+PBX

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I noticed that the two pages are identical. I hope someone would organize the wiki so that it is easy to follow. Hardware requirement >> Installing >> Configuration >> Main and submenu items. The content is there.

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

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My suggestion would be, after getting this all working:

  1. Figure out the settings for your Default Outbound DID/TRUNK, that will route all calls.
  2. Purchase a test DID, so that you can set up an inbound route with your SIP gateway provider. Open up the required ports on your firewall for RTP and any gateways, make sure ONLY SIP and RTP traffic are flowing two and from the gateway, and that ONLY your gateway provider can see your FreePBX via those ports to your PBX, this allows you to gateway out and receive calls from anywhere, protects your FreePBX.
  3. If you are using fortigate, Turn off SIP/ALG management and just have your free PBX manage all of that, for your policy, to and from the gateway to your FreePBX.
  4. This will allow you to make an intial call and it will feel SO good :slight_smile:
  5. Make sure your masking is correct for International and long distance calls, so that no nefarious long distance calls are made. You can tweak things more once you get SIP and RTP flowing and you can take calls both ways. If you get drops etc, or voice only on one site it usually means SIP and RTP are not being handled properly in their signalling and handshaking to and from the FreePBX.

Sorry this was for @samdland

If you are looking for a good Softphone take a look at xtelsio CTI /Phonesuite CTI both are the same. The pro Client costs around 40$