Problem with inbound calls from

I have been searching and trying various different solutions I have found but they are all for older versions of freepbx.

I just installed a new load with the following:

Release Date: 2016
FreePBX 13 • Linux 6.6 • Asterisk 13

I have a current version of trixbox that is currently working and I am trying to upgrade to latest freepbx.

I have dedicated IP for new freepbx.
I have setup extensions and calling between works fine, and outbound calls work with no issues.

Problem I am having is inbound calls don’t route at all. Below is what I am getting.

[2017-03-08 13:02:14] NOTICE[2051]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request ‘INVITE’ from ‘"XYZ Company " sip:[email protected]’ failed for ‘’ (callid: [email protected]) - No matching endpoint found

From what I can see the system is trying to route from the callid not the to:... in the signaling.

I have had remove the +1 for incoming and outgoing.

Any ideas, or if you need anything else let me know

What is answering that connection ?

The 216. Ip is bandwidth sending the call to the freepbx.

From the log it looks like freepbx is using the callid from to determine the inbound DID not the (to:…) like our existing pix does.

Again, which process in FreePBX do you have answering udp/5060?_

I have the trunk setup as follows:


and there is an inbound route setup with the phone number setup in the incoming DID routing to one of the phones that has a extension that works between other phones and is able to make outbound calls.

If that is not what your looking for how do I find the process?

I’m pretty sure that


won’t work out, make two trunks each with it’s own host , but apart from that you have pjsip and sip running, they can’t run on the same port, but you previous posts indicated that you might have misconfigured them, for the longest time SIP over UDP would listen on 5060 and your provider does that, BUT pjsip doesn’t yet work fully especially with some providers, although it now the default in FreePBX ( go figure :slight_smile: ) So I suggest you use chan-sip on 5060 and chan-pjsip on something else, if your extensions need the added facility of pjsip, adjust the port the use, if not just stay with chan-sip all round for now.

you were right…

I didn’t realize that pjsip defaulted to port 5060 and they moved chan_sip to 5160…

once I changed it inbound calls started working.

I want to assign multiple extensions to the same phone and with pjsip that was working once I switched everything to chan_sip the extra extension on the phone stopped working so I will have to see if I need pjsip or something else…

It looks like from the logs that the extra extension that was working just fine when I was using pjsip and switched it to chan_sip is having a password error…

So maybe that is something else that got messed up when switching between pjsip and chan_sip

If you have an idea about that let me know.


Multiple extension in chan-sip, not a problem if you have the keys, one ‘endpoint’ connected to multiple endpoints concurrently, then yes for that to work you will need PJSIP, adjust the extensions to use the port your pjsip is running on, you should be hunky-dory