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WRONG CALL STATE usually implies that FreeSWITCH tried to authenticate on the INVITE but couldn't complete. Given the second clue about TCP working and UDP not working, that makes me think that one of the routers is blocking arbitrary UDP ports. Create a firewall rule that allows everything from...

One word of caution, make sure your users stick to US English. I wondered why a few people were getting blank menu bars, and it transpired that people had switched their language to UK English.

No, just during a call press *4, silence will then play, dial 4003, wait for it to connect, finish the transfer by hanging up the call or pressing # to cancel the transfer.

I don't think you want hash at the end. It kills the transfer. I think you just have to wait after entering the digits. Feature Code Channel Variable Name Purpose When to use 0 attxfer_conf_key it will convert the call to three-way conference After the last party answers the call #...

Oh, and I think hash terminates the call from the FreeSwitch docs: https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Modules/mod-dptools/6586414/ Think you are just meant to wait after dialing, speak to the person and then hangup your side.

https://github.com/fusionpbx/fusionpbx/blob/master/app/dialplans/resources/switch/conf/dialplan/085_bind_digit_action.xml Think you are missing the ${bind_action_target} from the actions.

It's so much easier I have found to just create another menu, and assign that to the user/domain in question. That way, you don't have to mess with the default menu.

It's handled by the endpoint mainly. What do you need to know?

Yeh, we do it all the time. The default out of the box config for FreeSwitch is designed for a medium sized PBX. I'd up it to at least the number of extensions you have. The only thing to look out for is performance issues if it's too many, by watching CPU usage.

Looks like it's missing the 'answer' to me.

It's a big blast of invites, for the most part it will probably be fine but you'll really have to test it out yourself. I can't remember if it requires upping the FreeSwitch limits to beyond 30 sessions per second or not. I recommend if your phones have the capability and you are on the same...

Only when you create one, after you pulled that update. If you did at all, you can always compare a working one in the dialplan with a broken one. The IVR is in XML format, so make sure to click the button in the top right for XML

sofia_profile_name in the CDR detail probably if you see one. If you can post a log of an attempted call, that might help. Remember to find and replace any sensitive information like phone numbers and domain names.

It's free CPU usage. It's dumb, and I'm not entirely sure why they added it tbh.

It typically won't be an issue on most routers as the port will be rewritten to another external port during the NAT process. You only really need to change this when an overzealous IT admin has created a custom firewall rule to drop traffic sourced from/to 5060.

If the phones are in the external context, that tends to tell me that they are not registered and are simply sending INVITES to the PBX.

"Thomson Speedtouch ST546 v6 DSL modem/router" Holy shit, a 546! That was my first router I dabbled with back on the support desk days. No WiFi on the bugger, but it did have an SNMP responder!

Possibly the behaviour change is this commit? https://github.com/fusionpbx/fusionpbx/commit/92123052e1921adccc8b48248f48bdb2b3e87412

Probably a question for your SIP trunk provider, I have seen a similar fault where an inbound carrier had the DTMF clipped as some sort of VAD/compression, was only for specific routes into the network though.

It's way easier if your SIP endpoint does this already, most do at least support a 3 way. Alternatively, the att_xfer feature of FreeSWITCH does allow you to make a 3 way upon pressing a designated key. There is already a *4 bound digit: I suspect all that needs adding is the feature code to...