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>We use FusionPBX for testing, so we need to generate a wide range of scenarios including how calls might arrive from a carrier, hence the request above. Look at https://github.com/SIPp/sipp for this task. You build a porfolio of SIP scenarios and can replay them at will. Very flexible As for...

I think you have to slot an action of application: 'ring_ready' pretty early in the inbound routes. It will break anything that uses playback without an answer though, so make sure you test any ring_groups, IVRs, voicemails, etc. I have to ask, what's the problem with early media?

https://github.com/fusionpbx/fusionpbx/commit/d796eee8a020af5cf7d7048e7e6f18fe2bcb865a This commit seems to explain it, I think Mark implemented some sort of user security to the sync/reboot commands. I can't quite grasp it, but it looks like it's checking to what extensions the user you're...

Keepalived is using VRRP, so everything has to be on the same layer 3 network. You have to remember that SIP the protocol uses FQDNs for a lot more than just the initial DNS lookup. If at any point you find yourself using IP addresses in SIP configuration, you know it's not going to float around...

/etc/cron.daily/fusionpbx-maintenance Have a look at that file.

Anyone else on a relatively recent release notice the Call Forward on No Answer doesn't work? For example, here's an extension that has it enabled. (No other follow mes or forwards for simplicity) 6b3c3dc0-1517-41f6-84e8-eaf19ca99810 Dialplan: sofia/external/442071111111@example.com Action...

1. As much as possible for your use case. Personally, I would love to firewall the entire PBX off to a specific network and make all the users connect via a VPN, but this won't always be available. For what it's worth, I've had several FusionPBX servers sat on the internet for approximately 5...

FreeSwitch won't let you register to a domain it doesn't know about, in this case presumably the VIP. It should end up, ideally with domain names: Server Address: sip.example.com (configured on both fusionpbx servers, probably via pgsql mirroring) Outbound Proxy: proxy.example.com (your...

That is to do with the Cisco 7970 there, not the PBX. You'll need to log on to it to check what the dialplan is configured as. Sometimes they'll come with a default that cuts off digits and such.

You might have to create a SIP profile on each PBX that uses the floating IP as the SIP/RTP address. I imagine there's freeswitch logs which state registrations on domains that don't exist. I've never used keepalived (well, for anything other than http), so I'm not entirely sure myself.

Sure, why are you forwarding 5060 to 5090 though? By default, the only internal port is 5060, and 5061 if you have set up TLS.

SIP is definitely designed for multi-tenancy. That's what the domain is for!

You'll probably want to start freeswitch in the foreground to get a better idea of what is going on: sudo -u www-data freeswitch

Hm? Video?

Sounds like some provider nonsense to be honest. Probably their version of fraud detection.

Not from the GUI. SSH to the machine, then use 'fs_cli -q' from a priviledged user. This puts you into the freeswitch CLI fsctl loglevel DEBUG Then quit after: /quit or \quit, I can't remember which.

That's essentialy saying the endpoint is rejecting your SIP Options pings, so Sofia won't choose to send calls to it. If your gateways match the providers documentation, it might be a question for them. Maybe they have ACLs in place, I'm not sure.

Yeh, I suppose that could be a pull-request in the destination page. It's weird that SignalWire's guide doesn't pick up on it either: https://signalwire.com/resources/guides/fusionpbx

It's the only thing I can think that would break both playback and upload. The file ends up nicely in my recording folder: # locate 1d6f8 /var/lib/freeswitch/recordings/pbx.example.com/1d6f8e21-3038-4334-9302-81ed165ef213.mp3 # file...

Think you might want an optional plus in there to handle E164: ^\+?(0123456789)$