Failed to create authenticated REGISTER request to server

I’m brand new to FreePBX. I have previously used Asterisk (not FreePBX) with sip.conf and was able to register my SIP provider. However, with FreePBX (brand new install on Debian OS Google Cloud instance), I’m using pjsip by default (port 5060? I never changed anything), and no matter what I try, I can’t register. After I apply settings in from the trunk module, in the asterisk console:

res_pjsip/pjsip_configuration.c:2211 ast_sip_retrieve_auths_vector: Auth object 'sip' could not be found

res_pjsip_outbound_registration.c:988 handle_registration_response: Failed to create authenticated REGISTER request to server 'sip:voip.provider.com:5060' from client 'sip:[email protected]:5060'

Endpoint from-trunk is now Reachable

This is saying there is a broken pjsip.conf configuration. Without the relevant parts of pjsip.conf and the files it includes, we are not going to be able to work out exactly what you have done wrong, but it looks like you are missing a type=auth section called “sip”, or more likely you have the wrong name in the reference to it.

Here’s what I did:

Connectivity > Trunks > Add SIP > Add SIP (pjsip_chan) Trunk

General >
Trunk name: from-trunk
Outbound caller ID: org
Maximum channels: 3

pjsip Settings:
Username: 15551235555
Secret: hackme
SIP Server: voip.provider.com
SIP Server Port: 5060
Context: from-trunk

Codecs >
Unchecked everything except ulaw

I left everything else as default. Also, I looked at pjsip.conf, and this is all that’s in there:

type=global
user_agent=FPBX-15.0.17.62(16.22.0)
use_callerid_contact=no
keep_alive_interval=90
taskprocessor_overload_trigger=pjsip_only

Very little is done in the top level file in FreePBX.

I’m no longer getting the error. It seems that I needed to put in my username twice. Once in username and once in Auth username. Now I’m getting activity in the Asterisk console. However, when I put sip show peers, it’s showing 0 peers, which is very confusing.

All I want to do is set it up so that inbound calls are automatically placed on hold, and moh (from an internet radio station; an Icecast server that the made) plays.

Try pjsip show aors

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