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Zoiper is nice, but no BLF. 3CX works too, but the BLF only works with their server. X-Lite is filled with ads and feature limited now to push sales for its big brother Bria. I had a license for Bria once upon a time and it worked well, though I don't remember if it had BLF. I use the Bria app...

Trunks seem to be working, but I can't get to their website at all. Just times out from here. Anyone know anything?

I can take a guess and say what's taking up space in /usr would be source files for compilation, but I could be wrong. Can you do another search there to see what's the largest in that directory? Your second screenshot is the same link as the first, but in any case you are correct - "full" in...

Maybe? I'm not 100% sure since I almost always rebuild fresh. But that said, I'm not doing much fancy. If you have two boxes or are migrating to a new box, I'd give it a shot. Technically speaking (and last I heard) if your FreePBX versions match and Asterisk major version (1.8/11/13), it...

You don't need to update IPtables if your external IP address changes (that your PBX is behind). You will need to make Asterisk and select providers aware of it. What provider do you use? Do you register to the provider or do they send calls directly to your IP address? As for updating Asterisk...

Good to know! Don't have any of those phones myself, but I'm filing this away in the memory banks for later. Thanks!

For giggles, I just spun up a brand new Incredible 13 server and the same issue is there, even on PJSIP. I'm calling it a night - still stumped.

Obi 110 should do you just fine

I'll give it a shot. Can't hurt. Box isn't hurting for resources. Edit: No go. Doubled RAM to 4GB and added a CPU core. Nothing's getting banned. If something was getting banned, I wouldn't be able to place a call at all. That, and it'd affect more than just the VVX.

So I tried using an extension that's forced on the Asterisk side to ulaw only and the same issue occurs. The plot thickens... More testing. I don't think it's the firewall (pfSense) that I'm behind. Testing with Bria on my GS6 over Verizon 4G and G722 is clear as day. Ulaw works too. Edit: 4G...

Nope, as I have no reason to. Could I do it for testing? Sure. Edit: Just tested with a subaccount to VoIPms and it works fine without audio issues. So the issue definitely seems to be within Asterisk. Apparently that's more useful than I thought. Thanks for that! It's virtualized on ESXi -...

Welcome to the forums! First, please sanitize the configuration you posted. It's a very bad idea to post credentials (working or not) for anything on such a public forum. Second, It's unfortunate if they require you to have a separate trunk for each DID. Be that as it may, you should be able to...

Yep! Just ask them if they support SIP registration. Most use that by default and use static IP routing only if requested.

Did you create the extension as chan_sip or PJSIP? Do you have the server address set correctly in the HT502? You may also want to try temporarily disabling IPtables to see if it will register. Shouldn't matter for local networks, but worth a shot.

@MGD4me Yep, tried disabling everything but 711 u. Still inbound garbled voice. It's still puzzling me how it's only for calls going outside the PBX though. Everything internal is fine. That's what makes me think it's a codec transcoding problem, but everything I've seen in my log hunting says...

We're going to need more details. Is the HT-502 on the same LAN as the PBX? What version of PIAF/Incredible are you running (output of 'status' is helpful here). Have you checked that your credentials (Ext # and Secret) match between the PBX and the HT502? Does anything show up in the Asterisk...

I bought a VVX 410 new to take with me abroad this summer and while I was over there, all inbound voice was garbled. It wasn't terrible to the point of not being able to hear, but voices are noticeably distorted. "Garbled" is the closest word I can use to describe what's happening. I've been...

Just depends on your DID provider. Anveo Direct, for example, sends all calls to one IP address. If your IP changes and you don't update it in the Anveo portal, your calls go nowhere. Other providers, such as VoIP.ms, use what's called "registration" where your server registers to theirs and if...

If you temporarily set a catch-all route to go to an extension, does it work? Also double check that you have the DID entered properly in FreePBX/Incredible GUI.

Call logs are definitely needed here, and potentially SIP debug too. Start with sanitizing asterisk -vvvr when placing a call that doesn't work, and posting it here.