Posts made by bigbear

@aaronstuder said in Correct Settings For Hosted FreePBX 13:

@bigbear said in Correct Settings For Hosted FreePBX 13:

Running a vultr $20 instance.

Why? o_0

Cause? What do you suggest?

@aaronstuder said in Correct Settings For Hosted FreePBX 13:

Have you tried disabling the firewall temporary and testing?

Then you at least know the issue is the caused by the firewall.

Yup, everything works and its a grand ol time...

Running a vultr $20 instance. I am using the responsive firewall even though my inclination is to configure the vultr firewall for the explicit privileges necessary, mostly I want to do it the "FreePBX" way so I need some guidance.

I have configured everything as best as I can figure based on what I think the developers are intending to manipulate ipchains. My remote yealink thinks its registered, but it shows offline in reports - asterisk info - peers.

I am just going to attach a bunch of screenshots of my settings, hoping someone will see somethnig I have done wrong and point it out. Thanks in advance!

//well, I cant upload photos as it says I dont have a permission, so I created a google photos library...

PHOTO LIBRARY OF MY FREEPBX FIREWALL SETTINGS

When it does register under CHAN_SIP I get this in the event log. Like it registers but is immediately unreachable...

"[2017-03-06 19:10:51] VERBOSE[1903] chan_sip.c: Registered SIP '8001' at 192.168.15.141:5060
[2017-03-06 19:10:55] NOTICE[1903] chan_sip.c: Peer '8001' is now UNREACHABLE! Last qualify: 0"

Of course since the extension is setup CHAN_SIP I guess that would made sense - DERP.

Just wandering if everyone is using openvpn. On FreeBPX I didnt use it, just had to put in some settings for the LAN.

Looking at your settings, do you just connect OTT of the firewall or do you use OpenVPN on the remote phones?

Using these settings my Yealink 48G shows registered on port 5160, but in the Asterisk Report I run under CHAN_SIP peers it shows unreachable instead of a ping status.

When I switch my Yealink to port 5060 registration fails pretty quickly.

Asterisk error logs shows a lot of this...

"[2017-03-06 19:08:20] NOTICE[16244] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"Andy" sip:[email protected]' failed for '70.60.148.110:5060' (callid: [email protected]) - Failed to authenticate"

Hi all,

Andy Myers. I am about 15 years a WISP engineer and have spent the last 5 or 6 years expanding into BTS/VSAT/Cellular projects.

Currently working at a WISP who provides business internet and voice primarily, some residential service in rural areas where big cable still doesn't reach. We do some speciality cellular work for military training facilities throughout the U.S. and some Maritime ISP stuff due to a large client we brought on a few years back.

Working on expanding my knowledge of Python, Ruby, VoIP switches such as FreePBX and Freeswitch.

Follwed link from SW to Jared's FreePBX Guide. Had some questions...

1.) Jared mentioned in another post using config files to manual provision the phone system. Is he hosting them on the client freepbx install or a random FTP/Server?

2.) Is the dynamic firewall safe to use or hosted PBX and remote phones? I assume if you get locked out you can always access the direct console from vultr's and disable the firewall.

3.) Is it best to still use CHAN_SIP or is PJSIP viable for hosted pbx?

4.) Has anyone used the Remote FreePBX RMS module. There is no indication of the monthly cost per install so I am reluctant to waste time setting up a demo.

Thanks!