As the title says, my trunk fails, but it shows as online on my WebUI and shows registered on my provider site VoIP.MS, it’s very weird, really…
Here’s the log, btw: https://pastebin.com/3BeKQryH
Not really sure what else to say, both inbound and outbound calls are not possible at the moment (Not that I need them to be right now, just future-proofing, I suppose)
You have the paste as private. No one can view it.
Except I don’t?
Oh… it’s pending moderation… so it’s under manual review…
Should I just do a summary thingy…?
Try pasting (with pjsip logger on) at pastebin.freepbx.org (set delete after to 1 Month)
I think that’s what I did prior to posting this thread
What happened at pastebin.freepbx.org that caused you to retry at pastebin.com ?
If an attempted outgoing call does send out an INVITE to VoIP.ms, just post it and the resulting responses in the thread.
This paste has been deemed potentially harmful. Pastebin took the necessary steps to prevent access on July 24, 2025, 7:42 pm CDT. If you feel this is an incorrect assessment, please contact us within 14 days to avoid any permanent loss of content.
Very strange, I’ve never seen an Asterisk log rejected like that. I assume that you took a section of /var/log/asterisk/full , replaced usernames, phone numbers, IP addresses, etc. with redacted values and pasted it otherwise unchanged. Did you add any text or code that might have seemed harmful?
Didn’t know that existed…
Ahh, not sure how to do that though, been ages since I last did this (Like five years…)
No… And that IS very bizarre… they must be using AI… or something close to it…
Here’s this: /var/log/asterisk/full - FreePBX Pastebin But this DOES seem to be privated, and the checkbox is grayed…
I must also apologize for my slow response, I was out and about as I was waiting for responses…
Update, got inbound calls working by setting the authentication to “Outbound”, don’t really wanna set it to “none”, for obvious reasons…
Update 2: Got it working now… was a major pain, but I figured it out… So for those tuning in: Here’s how I fixed it: First and foremost; When creating dial plans, do NOT populate the “Outbound dialing prefix”, it DOES NOT work the way it should, instead keep the “prefix|” bit populated with your desired digit(s) to dial prior to dialing out, VoIP.MS is NOT a Centrex system, treat that as a typical digital POTS line like from your cable company’s phone line, where it doesn’t exactly matter what you hook up to it, it just runs as quick as it can. As of right now, the inbound calls are UNAUTHENTICATED! Not sure if this is important to have it this way, not sure if there are safety risks implied with this, but if not, then great, keep it as “OUTBOUND” Otherwise I imagine that you’d want both ends to be authenticated, but I’m not 100% sure how that’d work. In the “PJSIP Settings → Advanced”, LEAVE IT AS DEFAULT! DON’T BOTHER POPULATING ANY OF THAT! It will happily refuse to authenticate and you will CRY! On the outbound/inbound routes, it’s pretty self-explanatory, just fill those out the typical manner, populating the name and DID/Outbound CID stuff, the usual things. That’s what I’VE found to work.
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