@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.

@Romo said in Freepbx- Troubleshooting outgoing calls (Call hangups when other side answers):

@scottalanmiller said in Freepbx- Troubleshooting outgoing calls (Call hangups when other side answers):

@Romo said in Freepbx- Troubleshooting outgoing calls (Call hangups when other side answers):

@scottalanmiller said in Freepbx- Troubleshooting outgoing calls (Call hangups when other side answers):

Why is Speex appearing in your voice path? Set all devices to g711 u-Law.

So that did it, just removed speex from the allowed codecs and calls are terminating properly.

That's what I figured. Somehow it was getting priority on one of the legs and that was causing it to need to transcode, which should work in theory, but you don't want in reality.

Why would it take priority if it was the last codec setup. I had it like this

allow=ulaw,alaw,gsm,speex

THe phone might be set to take it first, if allowed.

This is the register process that appears in the cli

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@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

@scottalanmiller fwconsole starts asterisk, and it makes sure it's started and running with the correct UID

Or it should. It did last time I checked 😎

Edit: Saying that, I believe I was the last one that messed with that code, so if it's NOT working, it's probably my fault.

https://github.com/FreePBX/framework/blob/release/13.0/amp_conf/htdocs/admin/libraries/Console/Start.class.php#L147

@tm1000 said in FreePBX 13 No Conference Room App:

This was happening because for some reason the conbridge.conf file (or an include therein) was not being created correctly and thus because conf files were missing confbridge was never loaded. I believe this was because conferences was not originally included in the tarball for 13 but we fixed this so it should be OK???? Not sure...

It was fixed this week? It was just six days ago that we saw it, but it's not been re-run since then.

@thwr said in MC16: VoIP with Jared Busch Session:

@JaredBusch said in MC16: VoIP with Jared Busch Session:

@scottalanmiller said in MC16: VoIP with Jared Busch Session:

@JaredBusch said in MC16: VoIP with Jared Busch Session:

@thwr said in MC16: VoIP with Jared Busch Session:

@scottalanmiller said in MC16: VoIP with Jared Busch Session:

Myths of VoIP:

No 911 Is not secure Requires dedicated Internet Needs special network configuration have to buy all new hardware (or any new hardware!)

Seriously? ...

Serious that those are myths of VOIP? yup. I hear those all the time.

Same here.

@thwr Just happened last night in fact when i gave a presentation to a SpiceCorp.

Really sad. But there are so many myths (and half-knowledge) in and around IT...

Seems to be the majority more often than not 😞