QUESTION 603 Error: Unavailable - No trunk or registration

[PBXhob]

I've been trying to get an IncredablePBX server running for months now, I set up a fully operational FreePBX server on CrownClowd (thank you Ward for the blog post), inbound, outbound, IVR, extensions, everything so then I wiped the server and used the Incredible PBX 2021 Debian iso. Set everything up the exact same way, only having to change the PJSIP port to 5060, CHANSIP to 5062 (same for FreePBX setup,not using CHAN at all so I don't think that port matters) and whitelisted the entire flowroute FQDN using the /root/add-FQDN command in order to get trunk connectivity.

Outbound works, Internal calls to extensions work, Inbound calls do not come through at all. I have an Inbound route set up the same way as on my FreePBX, no advanced settings edit, just the DID number and gave an extension to it. Tried having it ring to an announcement, lenny, anything. All I get when I call the DID is a long pause followed by blasting *beep* *beep* *beep* tones. No error message.

Not only does it not connect the call, the Asterisk logs don't even show a call coming in and the server side CDN records don't show any incoming calls as well.
The only thing I have to go off are my trunk providers CDN records that show all calls to my DID terminate with error 603. Unavailable - No trunk or registration.
On my trunks side all the settings are exactly the same as when I had it working with FreePBX, down to the IP. I did not change anything.

I really don't know what to do. I googled and could not find a single person having this problem. (Some 603's but they were for completely different issues)
Could anyone point me in the right direction? I don't know how to debug this without log files. The server and trunk are clearly connected, I can see it registered under the settings and I can make calls out. No idea why I can't call in.

Thanks for reading.

 

[kenn10]

First, what company is your incoming provider? Secondly, if the call is not showing on your PBX logs, either the provider is not whitelisted on your PBX, your trunk is set up wrong or there is an error on the setup side at the trunk provider.

 

[PBXhob]

First, what company is your incoming provider? Secondly, if the call is not showing on your PBX logs, either the provider is not whitelisted on your PBX, your trunk is set up wrong or there is an error on the setup side at the trunk provider.

The provider is Flowroute.
I'm not that adept with Linux or Networking, but I ran the whitelist script for the entire flowroute domain and the Trunk looks for a FQDN and not an IP so I thought that would be good. Is there any way to test if my PBX? I can make calls to outside numbers using Flowroutes trunk service.

The setup is exactly the same as what I used for FreePBX and I could call into that.
Could this actually be an error on my trunk providers side?

 

[kenn10]

Also, did you use these instructions to set up the trunk? Note the "registration = send"

FreePBX PJSIP Trunk Setup

FreePBX PJSIP Trunk Setup ——————— IMPORTANT: - Details in this document are for reference only, and are unsupported by the Flowroute support staff. - Please use as pure

support.flowroute.com

If I recall, Flowroute requires inbound registration. PJSIP is the preferred trunking. Not registering the trunk with them would cause your issue.

From the Asterisk CLI, do a sip show registry or pjsip show registrations. If you don't have trunks for Flowroute registered, you'll see it there.

 

[PBXhob]

Also, did you use these instructions to set up the trunk? Note the "registration = send"

FreePBX PJSIP Trunk Setup

FreePBX PJSIP Trunk Setup ——————— IMPORTANT: - Details in this document are for reference only, and are unsupported by the Flowroute support staff. - Please use as pure

support.flowroute.com

If I recall, Flowroute requires inbound registration. PJSIP is the preferred trunking. Not registering the trunk with them would cause your issue.

From the Asterisk CLI, do a sip show registry or pjsip show registrations. If you don't have trunks for Flowroute registered, you'll see it there.

I followed that exact link to set up the trunk. Auth is set to outbound and registration is set to send. Replaced the match IP's with the ones flowroute says to use for my specific exit point.
sip show registry returns nothing.
pjsip show registrations shows the pjsip stunk I set up as registered.
name/sip:us-west-or.sip.flowroute.com:506 name registered

That 506 is strange. I went into the GUI and confirmed I have both the trunk and Asterisk PJSIP settings have the SIP server port set to 5060, not 506. Or is that normal?

 

[kenn10]

Part of the number is cut off when displaying. Not a problem. Did you set the Flowroute end to send the call to your pjsip port number? 5061, 5062 or whatever your settings are.

 

[PBXhob]

Part of the number is cut off when displaying. Not a problem. Did you set the Flowroute end to send the call to your pjsip port number? 5061, 5062 or whatever your settings are.

I rebound it to 5060 and yes. I have it on my IP:5060. Regular CHANSIP is on 5062.

 

[kenn10]

Last thing I can think of is that you should be registered to whatever location the DID is out of. The incoming call is not making it to your system from Flowroute. Either the registration is incorrect, the routing of the DID is incorrect or there is something wrong with your firewall.

Turn off fail2ban and iptables temporarily and see if the call comes through:

systemctl stop fail2ban
systemctl stop iptables

try your test calls.  Then reenable:

systemctl start iptables
systemctl start fail2ban

 

[PBXhob]

Turned fail2ban and iptables off using systemctl.
No effect.
I also tried setting the ports back to default of chan on 5060 and pj on 5061, only killed the trunk connection.
Switched the ports back.
Connection is back, I can call out but inbound still is not functional.
Thanks for trying to help.

 

[kenn10]

when you change sip/pjsip ports, you have to restart Asterisk.

SIP Error 603 means that Flowroute was able to reach your system but the system declined the call. Make sure you have the g711u codec turned on and at the top of the list in the trunk's CODECs tab.

Go to the Asterisk CLI and type pjsip set logger on. When the system is quiet, make an inbound call and look for data from Flowroute's IP's. You should see traffic between your system and theirs. This will give you a clue. When through monitoring the traffic type pjsip set logger off.

 

[kenn10]

Lastly, make sure your inbound routes have the correct format for the incoming number. I can't remember how Flowroute sends the calls but I'm guessing it is 1+10-digit number. Be sure the route points to something that can be dialed within the PBX. If your incoming number is being sent as 12225552323 but your inbound route has only 2225552323, the call will fail.