That’s what I did even shorten the name because of an apparent bug according to that site I copied the setting from for SSL/TLS above:
Surprisingly, there was nowhere for the key…when I tried to import it, it’s just rejected.
How do I go about getting them up and bridging them? I thought that was done automatically and presented as FreePBX.
Have you resolved the port conflict yet?
They key should never be on the client for obvious reasons.
You will need the extension to be successfully registered and the trunk to be also functional, either by registration or direct IP authentication, given these two being successful, then Asterisk will indeed be able to send a call from/to each other.
It has not presented that message for the pass two or three reboot; so I take it that has been resolved.
Yes, I am waiting on voip.ms to respond as to why their server showing rejected…the password is the same exact as on freepbx. I rather wait than to merely change password as something is not right.
When you get that resolved with the carrier and you fix your own network so the phone is registered, I think you will have made progress. Until both of those happen you are kinda stuck.
Sorry, I don’t speak Htek.
Maybe you should start over using baby steps, it works for most folks without such drama. both with your carrier and basic pjsip extensions, get that working then layer on your particular spin on things.
I have been doing just that the entire month of February…I have configured HAproxy but won’t enable until the phone is working. I have a feeling it could have something to with the service isn’t registered…but that doesn’t explain the send/ack missing segment every time I do a capture resulting in an unregistered phone to freepbx. I will walk away for a few days even if I heard from voip.ms…I didn’t have very much trouble with Twillio with the same phone except receiving calls before I had to shut it down because of war with ISP. Cresstalk make it look so easy although he was dealing with port 5060.
I think you are missing the advice to start with baby steps, then there would be no problem setting up your trunk with voip.ms because lots of folks use them.
Similarly, no-one has a problem setting up pjsip extensions, so get your trunk working, get your extensions working initially on the default UDP/5060.
Step by step add a valid certificate and make sure TLS is working in the big wide world this will need a public URL to resolve that certification, next add media encryption for your extensions, and make sure that also works. If your carrier supports TLS then step by step, set that up as per their instructions .
At this point you should have a system that call in and out.
Once you have done that I suggest then you can be as clever as you want with pfsense and/or haproxy but know you will be seriously off the reservation here, so please don’t expect too much further advice here.
I wish you well though., you are ‘boldly going where no-one has gone before’ but it would behoove you to actually “Understand the FreePBX/Asterisk” first 
Your screenshot shows
Port to Listen On for both tcp and tls set to 5061.
This is incorrect (the ports must be different) and also strange; the defaults are tcp:5060 and tls:5061. If you need to use a non-standard port for tcp, choose something that doesn’t conflict. If you must use 5061 for tcp, first change the tls port to something else.
Note that if you change any of these settings, after Submit and Apply Config, you must restart Asterisk.
Of course, whatever ports you choose, your phones or other SIP devices must be configured to connect to those ports.
BTW, It will not fixe your issue, but I subgest you swap to Asterisk 18 or 19, because 17 is EOL.
Try yo use asterisk-version-switch and select a valid version for asterisk.
It’s normal to use a random source port number for TCP (TLS uses TCP). It looks to me as though the phone may have reused a port number combination before the end of the TCP TIME-WAIT state on the previous connection using those ports.
Web → Account → 1 → Profile: 1 → Use Random Port: Yes
SIP-URI Host:
David55, I think you hit the nail on its head…I was wondering why that German setup mentioned above had that as a requirement. I was wondering looking at Wireshark’s output why it was reusing a port and whether that could be my issue… I just looked up the SIP-URI and will need to learn more since it was a 3CX example I saw…thank you!
How do I go about doing that? I like living on the cutting edge; so, I would go with Asterisk 19…can I upgrade from the CLI or I would need to reinstall FreePBX which has Asterisk 18 or download a new instance that has Asterisk 19?
He just gave you the command to use.
what…this: asterisk-version-switch?
Just heard from voip.ms and they claimed that they had been monitoring the sip server and saw no registration attempts from my end…so there is a problem with Asterisk 17.
Cool, thanks!
