As luck would have it, I do have IPv6 on my test server. I do not have any "network unreachable" errors in my logs and a quick "lsof" of asterisk shows two open IPv6 ports:
lsof -c asterisk | grep -i ipv6
asterisk 30586 asterisk 12u IPv6 3188795947 0t0 TCP *:8089 (LISTEN)
asterisk 30586 asterisk 17u IPv6 3188796008 0t0 UDP *:51967
. So then the question I have would be if I were to use WinSCP to move files from the existing config to my laptop what directories do I find my trunk, route, extensions configs? And can they be just "FTP'd" over? What about the CDR DB? What about the saved VM's? What about my IVR recordings? There are a lot of directories to rummage through on the installation... Thanks!
Not to sound goofy, but in your experience how and what do you use to do "cut and paste". I'm envisioning having the management GUI for two live systems, one old and one new, open at same time and cutting and pasting. But that presumes you remember that I bought a spare Pi 3 - which is unlikely I think
True. Good news @SMTC , the new release of FreePBX 15 (the app IPBX has been based on - after removing nonfree proprietary code etc), is coming with the long anticipated "portable backups" feature. This lets you save massive time migrating to new PBX version. Tap or click, to get all your data out of IPBX 15, and restore to any other IPBX 15, 16, 17, or newer version. You're no longer limited to restoring all yer PBX data to the exact same PBX version. All the db tables stuff, conf file stuff, voicemail recordings locations, IVR recordings locations, CDR, conf files, are no longer a headache of copy pasting or writing programming code to get that data out of PBX version 12 database into PBX version 13 database.
Which would explain why I've never seen the error even with the testing done yesterday - one of the first things we do on every system is disable ipv6 unless needed. No ipv6 enabled would mean no ipv6 attempts and no error message.
Trust me, time at the support desk is our #1 priority and we do almost none of the above, and often the polar opposite. No ICE, no STUN, (almost) no pjsip, no IPv6, no direct media to endpoints (calls are usually managed, recorded, etc.).
If it works for you, great. And granted, ultra super secure encrypted direct media calls are probably less than 10% of all calls around here, they're absolutely required when necessary, yet the 90% majority of calls are your normal managed recorded calls.
Your roaming scenario scenario works probably 95%+ of the time now.
I guess my thinking is, why be a luddite and make users go thru that unnecessary interruption to their business conversation and thought flow, due to an unnecessary technical difficulties interruption. This PBX system is equipped with an advanced software stack specially designed to perform fast relatively seamless handover to a new IP address, in about a second, even when the user's endpoint suddenly auto logs on to a WiFi access point behind a NAT such as you find at all coffee shops retail stores public places of interest etc.Here's what works for me. "Hey, I'm about to walk into Starbucks. Let me call you back."
But you are ignoring the fact this works 95% of the time with the "traditional" chan sip and nat transversal methods. The incremental benefit is minor (or less) for our deployments.
The only thing that matters is how it works for my implementation. Overall histories and other uses are irrelevant.
Just for fun, if you would, reply with detailed step by steps demonstrating the situations you describe above:
Ring groups not working, blfs not working, general weird behavior - call parking/pickup issues if I recall, etc. Some things like the ring group issues definitely got better in FPBX13, we at least still had some weird blf issues IIRC. Had two smaller locations we tried starting out as pure PJSIP(it's the future after all) - that didn't last long. Not all of this (probably most of it) were technically PJSIP issues, more likely FPBX, (and maybe some of our own config), but that doesn't matter, there's no need to waste hours debugging when chan_sip just works. I'm perfectly willing to wait and let others get it worked out.
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