Okay here is an update. I have changed the port back to 5060 and restarted the entire system 4-5 times. After doing this the inbound/outbound calls work as of now but I have a feeling they will stop. I don’t understand why sometimes the calls work and sometimes they don’t, I really hope someone can help. I have captured the inbound call and the outbound call, I will post it in paste bin here for someone smarter than me to take a look at. I have changed a phone number to ########### for privacy. Also, a couple things worth noting. The outbound call doesn’t start ringing for like 15-20 seconds and then one of the outbound calls came in as 1111111111 on my cell phone.
Bumping this to see if anyone has recommendations. I am still facing the same issues.
I am also a newby 
and used Raspbx to get things going. One of the instructions I had to do is forward port 5060 (UDP and I did also TCP) to my system running Asterix but also UDP port 10000 to 20000.
Maybe that helps …
Just an update for everyone here, still looking for help. I was able to get the outbound calls working great now, I was missing the from user: in the trunk settings. But now when I place the inbound call it beeps as if the line is busy. I created a whole new sub account and new number on voip.ms and plugged everything into freepbx and the inbound/outbound calls worked fine for 60 seconds and then the inbound started beeping like the lines are busy and the outbound calls still work fine.
I have already opened up port 5060 and also 10000-20000, I opened up both TCP/UDP. Thank you for the input though I really appreciate it! This whole thing has been stressful and I can’t figure it out for the life of me lol. I am so close though, I really want to get it up and running!
I went to connectivity - trunks Went into edit mode for my trunk then sip-settings → incoming
My settings are:
type=friend
qualify=no
username=<11DIGITS-PHONENUMBER>
secret=<PASSWORD-AS-SUPPLIED-BY-MY-PROVIDER>
insecure=very
host=voip.cheapconnect.net
port=5060
context=from-trunk
and the register string is
MY-11-DIGITS:<PASSWORD-AS-SUPPLIED-BY-MY-PROVIDER>@voip.cheapconnect.net/MY-11-DIGITS
Is yours similar?
Just my 2 cents
This won’t work at all on Asterisk 21, which the OP is using, at least not as supplied by Sangoma, as chan_sip has been changed from deprecated to not present at all.
chan_pjsip does not use a register string.
I wouldn’t expect it to work on other supported versions, as very was changed from deprecated to not supported at all, some time ago. The reason for deprecating was that one rarely needs insecure=port, but everyone was including it, by using very. Unfortunately, they simply did port, invite instead, when they were supposed to go back to basics, and work out the minimum level of insecurity actually needed.
(In almost all cases, friend should be peer, and qualify is often needed to keep router firewalls open.
Thanks David55, I just followed the instructions my provider gave me. It worked for me running RasPBX (FreePBX 15.0.37.9 and Asterisk 16.13.0)
That’s not a good idea. Providers tend to give copy and pastes of other providers’ settings, so they tend to be full of obsolete things and accumulated bad practice.
Do you have a solution idea for my issue? I am able to consistently place outbound/inbound calls for the first 60 seconds after I restart the console and then after that only the outbound calls go through. I am thinking maybe it is something to do with the ports? I have the ports forwarded on my xfinity router to my host and then on my host I have them forwarded to the pbx vm. Also, I have added configurations to the vmnetnat.conf file to open up the ports on vmware NAT just for testing. Let me know any logs or anything you would like me to attach and I would be happy to.
Not without “pjsip set logger on” type output from the call, and accurate timestamps for the log (use the full log file, not a screen scrape).