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I think it's a bit different because it's also user based. This is much more limiting being only a single user.

@VoIP_n00b said in IPsec Site-to-Site:

Is using a IPsec Site-to-Site VPN safe?

Using a site-to-site VPN is not safe in the first place. It completely bridges two disparate networks. Allowing an attacker on one network to attack the other network, without anything blocking it.

If you are fully prepared to trust everything on both networks, then sure, use it.

Yes, you need to use only known good ciphers. That is no different than any encrypted communications.

@JaredBusch said in iOS Sip Client?:

Scott had me using Zoiper for years. Paid and free work as advertised. What issue do you get?

@VoIP_n00b said in Ring Paging Group When Extention is Called:

Crap. FreePBX.

LOL, that'll make it easier to figure out.

If your nginx reverse proxy is on Fedora, confirm that SELinux isn't denying anything.

sudo tail -f /var/log/audit/audit.log | grep 'denied'

@JaredBusch said in VoIP Drama Sangoma vs. Clearly IP:

Please also recall that the Rob being discussed as the co-owner of the key, is the person that prompted me to start the thread Revisiting ZFS and FreeNAS in 2019 after seeing him post about it on Facebook.

I had no idea what kind of a shit show that thread was going to turn in to.

Yeah, holy hell, I'd not trust anything that ClearlyIP produces. I'd be pretty worried to depend on them as a service provider!

Seems I got it fixed 🙂

set interfaces ethernet eth0 dhcpv6-pd no-dns
set interfaces switch switch0 ipv6 router-advert name-server '<IPv6 Address>'

@Alex-Jones said in FreePBX UCP behind a NGINX reverse proxy:

@scottalanmiller You can't login. I am able to get the the Web GUI

So it accepts the username and password, but it just doesn't work?

@JaredBusch said in Incoming Call Issue:

@VoIP_n00b said in Incoming Call Issue:

Running FreePBX v14... No resent changes made, except to change the name on an extension.

When some calls in, and let's say they dial 555 then end up getting 511, 511 can then forward the call to 555 no issue, but I have no idea what is causing this.

Because they released a horrible fucking update to IVR.

Edit your IVR and set strict dial time out to No - legacy

That fixed it! Thank you so much!!!

I'm going to assume you used a chan_sip trunk.

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Fist look at the chan_sip settings for NAT and the SIP signalling. Though nothing on the signaling side is set to 15 minutes. But since you need to check the NAT setting, check the other settings.
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Now look at the RTP settings, this is set to 15 minutes by default
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