PBX receiving traffic on random ports

Hi All,

my a bit rusty with FreePBX but getting back in the game, however, I hit a wall. I seem to be getting the following errors on the PBX

Request ‘INVITE’ from ‘"#MobileNumber# <sip:MobileNumber"at" RTP Address>’ failed for "‘SIP Signalling address:58914’ (callid: ########) - No matching endpoint found

I can’t seem to find anything relating to this on the internet however the forum posts with similar issues list the port as 5060. I understand how it all pieces together so I have come to the conclusion that the problem is this port not translating properly.

That being said, I sure the inbound NAT and firewall rules are correct and the sessions show that the firewall has passed NAT correctly to the PBX yet the PBX is still showing the traffic as the original source port. I have also played around with the outbound NAT but still can’t get the port to translate properly.

Am i right in thinking this is the problem or am i barking up the wrong tree?

for clarification, using:
FreePBX firmware 12.7.5
pfsense firewall
Simwood SIP Trunk

hoping this is something simple

NOTE: Getting an alert to say i can’t add links as a number user so had to sub 2 ip addresses for “RTP Address” and SIP Signalling Address"

Thanks in advance people

Update: for all who might hit the same issue. I was using a CHAN_SIP trunk and not PJSIP trunk. there was no endpoint found as all phones connected were using PJSIP.

No matching endpoint found usually indicates that the connection that failed either is set up for “the other” SIP driver (going to the Chan-SIP port instead of PJ-SIP) or that the actual extension they are trying to connect to doesn’t exist so the phone can’t register.

If the addresses and phone numbers all make sense, you are just sending them to the wrong channel driver. You can either point them to “the other” channel driver or drop and recreate the extension in the correct one.

If the addresses and phone number don’t make sense, it’s a problem with your incoming firewall. Since you’re using pfSense, you should probably take a look at some of the pfSense posts on the forum and make sure you are as up-to-date as you can be for everything.

Couple things. First, for future reference the RTP address is never in the INVITE. RTP information lives in the SDP body. The INVITE would be the IP of the PBX or the destination of the other side (trunks).

Two, Chan_SIP does not have the concept of “endpoints” it has the concept of “peers”. So if you are seeing errors such as “Not Matching Endpoint Found” that means it is hitting your PJSIP port and the PJSIP driver is trying to handle that request.

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