So I tried changing the PJSIP port associated with my SIP trunking provider from 5060 to 6060. There were some issues on their end handling the change. So instead I limit my public port 5060 to just their network. And switched the PJSIP port back in FreePBX.
I can make outbound calls just fine, but inbound aren’t even making it to the FreePBX. Nothing in the pjsip debug or in the logs as to inbound routes.
And when I look at the results of a pjsip show endpoints command, here is what I see for the SIP trunk. Note the udp 0.0.0.0:6060 transport that’s listed. Where is this coming from? I’ve rebooted the FreePBX after switching back to port 5060, and my SIP trunking provider is configured for port 5060 as well.
Endpoint: Twilio_Trunking Not in use 0 of inf
OutAuth: Twilio_Trunking/myfreepbx
Aor: Twilio_Trunking 0
Contact: Twilio_Trunking/sip:myfreepbx@diamondcella e3329b4bf2 Avail 1.009
Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:6060
Identify: Twilio_Trunking/Twilio_Trunking
Match: 54.172.60.0/32
Match: 54.172.60.3/32
Match: 54.172.60.2/32
Match: 54.172.60.1/32
I did, although there still were problems. Now if I have my desk phones using PJSIP port 6060, but wanted to only have Twilio use PJSIP 5060 for trunking will this cause an issue?
Did you do this the first time? Changing IP or port requires fwconsole restart or a reboot.
And before anyone wants to come in with a million other ays, yes there ae. but these are the “normal” methods. Normal users should either reboot or use standard fwconsole commands.
Sorry about that. I was too vague. Yes, I restarted via fwconsole restart.
The root cause of the “weirdness” was the SIP provider could send me inbound calls via the non-standard SIP port I wanted and configured. But when I’m sending out calls to them, the standard SIP port 5060 is hard-coded on their end.
That’s what had me stumbling for a bit. So I configured PJSIP under Asterisk SIP settings to define the port as the non-standard one. But in the SIP trunk settings I had to flip it back to port 5060. Then inbound as well as outbound calls are working.
You can only define and control half of a connection.
You can tell them that you are listening on any port you want. That is what you defined in the settings in FreePBX and then told the provider as much on the provider’s side.
Again, that is where you are listening. So that is where they sent you calls.
When it comes to sending calls to them, we’ll you have to send it where they are listening. You are not flipping anything back.
Yes this makes perfect sense now. My mind was stuck in the days of T1’s where both sides had to align as to SF/ESF, B8ZS/AMI, etc. In this case I didn’t realize I could receive on a totally different SIP port than I’m sending to the SIP trunking provider. Thanks for the insight!
