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This is not necessary, we have fail2ban in place that already does similar.

So where are your tariffs and fair use policy because you certainly won't be giving what you are offering? Unlimited international minutes? To Cuba, Sierra Leone?

Ask them for an example of a good outbound INVITE, then work from there. You also have not mentioned what your carrier actually wants.

Please continue to report them and help keep these forums spam free :)

Oh, and another thing that doesn't really get a mention: reliability. Once you have fusionpbx up and running and you don't mess with it, it will just run for years and years. Freeswitch is super stable.

I'm firmly back in the fusionpbx camp. I actually work a lot now with some commercial PBXs and the more I use them the more I dislike them. There is nothing that matches opens source. At the end of the day, fusionpbx can do anything I want. It may not currently be able to do it but that's...

Forget Debian 9 and follow these instructions... https://fusionpbx.com/download

The billing engine part IS Freeswitch. It uses a number of components. Though yes, asterisk will be there its something you need not touch or use in any way. That's just if you want to use the PBX features. The billing system is actually CGRates/Freeswitch.

Actually reading back, I don't think the above suits your needs as you still want some with pin, this just transfers them to a pinless conference, I'll look later.

I have a dialplan to allow certain outside callers in without a pin, the sip_from_user string should be easily modifiable to suit what you do and don't want let in

I'm not sure how much https://github.com/irontec/ivozprovider would fit your needs but take a look at it, its free. I've only tested it using the wholesale part but you just create a trunk for each PBX tenant. Its also truly prepaid if you want it in that client balances can NEVER go negative...

@dcitelecom I see you chastised a guy earlier in the thread for jumping on your post with a spam advert. that is EXACTLY why the author of the software you are asking about was banned from these forums many years ago, despite the fact that he was politely asked to stop, many times. He also ran...

If you are doing it from the PBX itself there is no need to curl. I would if I were you though, try and solve the problem itself. You may just be hitting NAT timeouts and all restarting the gateway does is refresh the tunnel. Maybe lower your register time down to 120 seconds on the gateway...

Just prefix it on the bridg, eg 12$1 Just use the wizard to add one and use the prefix then look at the generated dialplan, you will get the idea

The xml is generated dynamically from the DB and cache. The caching is very efficient.

Its either star or hash when the message is playing, can't remember which

I'd rather actually use the facility to email voicemails as an attachment rather than exposing this interface to the world. I've seen a good few hacks using remote voicemail.

Did you do this bit: echo "machine freeswitch.signalwire.com login signalwire password $TOKEN" > /etc/apt/auth.conf

Set caller id in from to true in gateway advanced settings. Don't for get to stop and start the gateway after changes.

Use sngrep to see what is being sent by the carrier