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I can respectfully disagree, I don't believe the upgrade process is intentionally painful. The way the system was designed certainly has warts when it comes to upgrades. All mature software projects I have ever worked with have their problems and the upgrade process in Fusionpbx could certainly...

I'm sorry I don't really have time to manage anyone else's upgrades. I really do think that Mark and his team would do a good job for your upgrade.

Look at post My steps are earlier in the thread. https://www.pbxforums.com/threads/teams-integration.4876/post-25749

I don't have 15 second delays in call connection. I would check your javascript console/sip.js logs for problems.

tyryko, that page has a pretty good explanation of the process. You should take a backup, and the guide clearly tells you that you should also take a backup of your provisioning templates explicitly as well. My upgrades are usually a mixture of command line (for the git and backup portions)...

I updated my steps to where I am more likely to land, and for sure the freeswitch gateway pings don't keep the gateways alive. You are better off using SIPP for sure.

leandrodes, The value you get out of my steps may vary based upon your knowledge of FusionPBX and of freeswitch call routing, but I will endeavor to flesh out the instructions I started earlier in this thread. Then maybe someone will turn it into a sticky or a wiki article later.

I had a gateway mark itself out of service because of the PING problem you mentioned last week, but because of how many times I had messed with those configurations I wasn't sure whether it was something I did or not. I'll keep an eye on it, it is certainly easy enough to go back to SIPP if...

gflow, I am not aware of it being able to do an attended transfer. It requests a destination number and only transfers blind. That said adding an attended transfer feature should be possible as the underlying sip.js library does have support for refer headers and can do it. Park has been my...

Daz, First of all, Thanks! So I was going to migrate to SIPP next, I probably should have just started there instead of wasting the time with kamailio. Using a modified OPTIONS request in SIPP I was able to get the SBC to mark itself as active in Teams. YAY!? However, calls to Teams are still...

A SIP trace from a functioning freeswitch would be nice as well...?

Yes Daz, If someone has a functioning freeswitch gateway and profile with all of the options set that would probably be helpful. I'm not sure whether the From and the To actually make a difference. I actually tweaked my kamailio config and was able to have it send OPTIONS pings over to my...

Well, in a fit of frustration I went an got Kamailio sort of working. Meaning at least I was able to get my SBC in direct routing to be marked as active. I used the exact same certificate from letsencrypt as well. It doesn't make sense why freeswitch/fusion isn't working. Kamailio Options...

As for my troubleshooting steps. I can see my gateway make an outbound SIP OPTIONS ping every 60 seconds to microsoft it looks like this: OPTIONS sip:sip.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS sbc.example.com:5083;rport;branch=z9hG4bKe6HgFv5vDcB1a Max-Forwards: 70...

Some rough steps for posterity's sake: Microsoft Side: Add domain and validate domain Assign domain to at least 1 user (otherwise you can't add an sbc for that user) Create sbc under voice->direct routing (once fusionpbx is configured and pinging properly you will need to validate that tls...

So I made some progress for anyone who is interested. In order to make tlsv1.2 work with freeswitch it was necessary to set tls-verify-depth in my sip profile to a value higher than 2 in order to make letsencrypt certificates work. I now have calls from MSTeams -> freeswitch/fusionpbx working...

Ughh.. still not working. In all of my messing I found that the Microsoft servers communicate ONLY via tlsv1.2. My freeswitch/fusionpbx profile is running tlsv1.3 and ONLY tlsv1.3. After some digging I found that by altering my /etc/ssl/openssl.cnf file I could allow tlsv1 to run, but tlsv1.2...

Ironic that someone on the forum would be the first to mention about the contact-in-ping parameter. https://www.pbxforums.com/threads/new-gateway-parameter.5657/ I made a quick hack to add contact-in-ping to my gateway but I doubt that is going to work on its own?? If that is the case then...

Did this work for you? By adding the contact-in-ping option to your Gateway were you able to communicate with Teams?

DigitalDaz, so if I understand you correctly you have an SBC in Microsoft Direct Routing that connects directly to a TLS enabled port on a Fusionpbx system. And in order to keep the Microsoft side happy you are using SipP to send regular "Options" updates to sip.pstnhub.microsoft.com:5061?