@tim_g said in Installing Wordpress on CentOS 7 Minimal:

@jaredbusch said in Installing Wordpress on CentOS 7 Minimal:

setenforce permissive;

Setup SELinux permissions

semanage fcontext -a -t httpd_sys_content_t '/opt/yourdomain(/.*)?' restorecon -R /opt/yourdomain semanage fcontext -a -t httpd_sys_rw_content_t '/opt/yourdomain/wp-content(/.*)?' restorecon -R /opt/yourdomain/wp-content

I run my servers with SELinux running in Enforcing mode, not permissive.

So if it's in enforcing, WordPress will not run correctly. Example, you can't update, and the Add Plugins page gives a connection error (it's because httpd can't write correctly due to selinux enforcing).

Instead of running SELinux in permissive system-wide, you can keep it in Enforcing mode, and issue the following command:

semanage permissive -a httpd_t

The reason I use that one instead of by directory, is because I don't know how to do that... well I do, but I didn't have time yet to see what all directories need to be written to for updating and plugins pages and such to work correctly. I'll add that here some other time when I get a chance to look in to it more.

I only said that for install message you remove it back to enforcing

@brianlittlejohn said in Outbound call routing with FreePBX 13:

Is it best to setup a second route for e911 usage?

I always do. Because then I can mark the route as such.

@JaredBusch I see that now. I'm not making plans to make any kind of changes with Asterisk's behavior. I was simply geeking out on the questions of why and how. 🙂

Justy a note, you need to create an OpenCNAM account and fund it if you do not have one.
https://www.opencnam.com/register

I recommend OpenCNAM because it is generally cheaper than your carrier CID charges.
If you recall from the VoIP.ms setup, they charge $0.008 per lookup.
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OpenCNAM is half that at $0.0039 per lookup. https://www.opencnam.com/pricing

Often sales people and management will ask, "How do I check my voicemail from outside?" I always recommend that they enable voicemail to email, but for those that want it, you can set up a silent option in the IVR.

What I mean by silent option is that you will add another option to the IVR, but you will not announce that option in the recorded audio.

Add another entry and choose the Destination category of Feature Code Admin and then select Dial Voicemail <*98>
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It should be noted, that when you create a new extension, FreePBX will create user conference rooms automatically if you create a new User Manager user at the same time.

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@u2communications said in Setting up a SIP trunk in FreePBX 13:

If you find yourself still trying to get incoming calls working after several hours (like me), be advised that the default DID settings on voip.ms will not work. You merely have to set up a sub-account and make sure to set the authentication type as user/password authentication. Add your DID's to the sub-account and authenticate with the accountnumber_subaccount convention and use the password you created when setting up the sub-account.

I have never experienced an issue using the primary account on VoIP.ms. I highly recommend against it, always. But, I have never had a problem using it.

Also, welcome to ML.

The last step there, "Networks" is still functionally the same, but the GUI has been updated.
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@JaredBusch said in Using Vultr for FreePBX 13:

Then, what providers support OPUS? Most only support 711 and sometimes 722.

Hence why we transcode to g722 for Skyetel. But the g722 path is from a datacenter to datacenter, so not really worried about questionable call path qualities.

Opus seems to help on everything from wifi to mobile to offices with low quality WANs. We have people on fractional T1 still, so fight some tight bandwidth battles.

@JaredBusch

We need proxy guide for shiity countries

I think I am unable to complete call due to blocking.

I configured 2 extenstions on 2 smartphones, and they both can ring each other, but never connect