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This thread might help. Not the same NAT problem but a NAT problem. https://www.pbxforums.com/threads/outbound-calls-drop-ack-not-received.3325/

Just throwing spaghetti at the wall here but on your Yealink phone under the advanced account settings try rport = enabled.

Try creating a different profile just for the remote phones using your hard coded ext-sip and ext-rtp settings. You'll most likely need to use a different SIP port but that should straighten things out for you. However are you sure your setting ext-sip and ext-rtp properly in your profile?

You shouldn't have a problem with pfsense and Fusion. I have a very funky test setup and it works when it probably shouldn't (lots of nating, douuble nating, etc), ok so lecture over lets check a few simple things: Check the RTP port range in the phone config to make sure it matches. For that...

I fought with something similar for a long time. The 30 second time out means as you've stated you're not getting the ACK, which means its probably a NAT issue. In my case I was using an ASTPP box as a mediator and was trying to do both external (to carriers) and internal (to fusion PBX) calls...

Are there any transcoder cards fro OPUS?

Looks like transcoding is going to be the deal killer here. Initial tests indicate an unacceptable CPU hit going from Opus to G711. While some carriers are providing Opus support, most aren't yet. Since we don't want to be on the hook for transcoding we'll need to stick with G711

LOL thats the article that prompted us to investigate it. Thanks! We'll probably go in that direction and report back the gritty details.

I have a colleague trying to convince me its a good idea, and it looks good on paper but G722 seems to be winning as a wide band codec. Anyone have any practical experience with Opus?

If you look at the "inbuilt" variable {$row.device_key_type} and you do a little spelunking into the database you'll see it matches a column in table v_device_keys labeled: device_key_type. This is how that particular variable is populated (or at least it matches the form field being populated)...

I'm not sure if I'm answering your question, but basically you can make your own variables. For example under devices and settings, this is my custom variable for a Cisco SPA ring tone So after putting that in the GUI you'd insert the variable {$spa_ringtone} into the appropriate spot in your...

Not sure what optimizations Google makes but because I'm cowboy (and because snapshots) I installed the cloud package. I'd really like to know what optimizations they made so I could compare metrics against an untuned machine. For simple folk like me here's the process: # Add backports to...

Couple of notes on this for Debian; chrony was not installed by default so you'll need to do an apt install chrony. Also chrony.conf is NOT in the root of /etc instead its in /etc/chrony/

These tunables are gold! Thank you! Based on the notes above found this on syncing the clock : https://opensource.com/article/17/6/timekeeping-linux-vms

I'm in the build it and they will come stage but I'm trying to get a handle on what to expect. I'm thinking a server with twenty domains, 500 registered phones and maybe 75 to a hundred concurrent calls at peak. All calls are G711. no video. may also have to mux a couple conferences.

I understand Asterisk is a completely different animal but we used to have a lot of IO and clock problems with it in LXC containers, also at times security issues but that was many moons ago. So now I'm gun shy and will trade a bit of performance for the abstraction I get from KVM. Djzort...

We're in the same boat. We have a testing environment set up with two small customers who have agreed to be guinea pigs. However we still have large knowledge gaps and free documentation is somewhat sparse at times. We are planning on becoming members (as we should to support the project) as...

Old thread but I'm planning on using FusionPBX on Proxmox KVM. I've had issues in the past with leaky containers and choose to go full hypervisor. I haven't put the test boxes under load yet. Has anybody had any experience with this type of setup?

Oh wait there's more. The Linksys SPA will not honour it's own SIP port setting so you have to modify line 415 to the following if you want to use a custom port: <Proxy_1_ group="Ext_1/Proxy_and_Registration">{$server_address_1}:{$sip_port_1}</Proxy_1_> After that It'll work.

If you're still on the ancient Linksys SPA devices you'll probably notice a lot of problems with the template. Here's one that's an easy fix. The Linksys template right now won't honour the SIP transport setting in the gui. To change this modify line 385 from this: <SIP_Transport_1_...