Res_pjsip_registrar.c:1166 find_registrar_aor: AOR '' not found for endpoint

pgr · December 21, 2022, 1:37pm

Hi, I am trying to configure a Grandstream HT813 (FXO gateway) to connect a PSTN phone line to my recently installed FreePBX VM. I am a noob with FreePBX and I am mostly trying to do things from the UI, although I have some previous experience with older Asterisks.

In rasterisk I get this message recurrently, approximately every 20 seconds:

WARNING[141870]: res_pjsip_registrar.c:1166 find_registrar_aor: AOR '' not found for endpoint 'LinhaMeo' (10.0.0.3:5062)

I’d like to remove this message, it’s making my terminal scroll while I examine SIP and asterisk debug output, making it really hard to work. And it’s probably an error I need to fix, any way.

Where can I find the place in the UI where that is coming from? I basically configured a few extensions, and a trunk called LinhaMeo. I didn’t define Users. That IP address 10.0.0.3 is the HT813. But I’ve tried a few changes and the error is always there.

Is that an error caused by the HT813 trying to contact Asterisk, or the other way around?

I assume you’ll want more information about my configs, just tell me what you need and I’ll be happy to provide it. Thanks in advance!

david55 · December 21, 2022, 2:00pm

I don’t understand what you mean by the user interface here. The message is coming from the core processing:

github.com

asterisk/asterisk/blob/master/res/res_pjsip_registrar.c#L1166


      
          
          		if (aor_name) {
          			break;
          		}
          	}
          
          	if (ast_strlen_zero(aor_name) || !(aor = ast_sip_location_retrieve_aor(aor_name))) {
          		/* The provided AOR name was not found (be it within the configuration or sorcery itself) */
          		pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 404, NULL, NULL, NULL);
          		ast_sip_report_req_no_support(endpoint, rdata, "registrar_requested_aor_not_found");
          		ast_log(LOG_WARNING, "AOR '%s' not found for endpoint '%s' (%s:%d)\n",
          			aor_name ?: "", ast_sorcery_object_get_id(endpoint),
          			rdata->pkt_info.src_name, rdata->pkt_info.src_port);
          	}
          	ast_free(aor_name);
          	return aor;
          }
          
          static pj_bool_t registrar_on_rx_request(struct pjsip_rx_data *rdata)
          {
          	RAII_VAR(struct ast_sip_endpoint *, endpoint,

I think it is saying that 10.0.0.2 is trying to register without any user part (extension number) in the registration request (in its To header).

jcolp · December 21, 2022, 2:04pm

It’ll also happen if the given AOR in the To doesn’t match a configured AOR on the endpoint. The logic should be reworked to make that clearer, but it hasn’t bubbled up as a huge thing.

I’ve seen this happen when matching incoming requests based on IP address (without a port), so it gets associated with the wrong endpoint.

david55 · December 21, 2022, 2:11pm

Wouldn’t you seen the unknown AOR in the message text, rather than ’ ', or is that what needs reworking?

jcolp · December 21, 2022, 2:16pm

That is what needs reworking.

pgr · December 21, 2022, 3:21pm

Thanks for your replies. I don’t fully understand some of the terminology (endpoints, AOR, … ), but let me try and make things clearer for you so you can help make things clearer for me

What I mean by “place in the UI” is “this error is probably something that I configured incorrectly either in the FreePBX UI, or in the HT813 configuration UI”.

But while it should be straight-forward to go back in the screens and find out, I am actually having trouble finding it.

Here are some more details (I obfuscated my phone number which starts with 21 by overwriting 212212212 manually):

<--- Received SIP request (554 bytes) from UDP:10.0.0.3:5062 --->
REGISTER sip:10.0.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5062;branch=z9hG4bK1989114344;rport
From: <sip:[email protected]>;tag=1868922450
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2030 REGISTER
Contact: <sip:[email protected]:5062>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-C074AD90A2AE>"
Max-Forwards: 70
User-Agent: Grandstream HT813 1.0.13.3
Supported: outbound, path, gruu
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


<--- Transmitting SIP response (321 bytes) to UDP:10.0.0.3:5062 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.3:5062;rport=5062;received=10.0.0.3;branch=z9hG4bK1989114344
Call-ID: [email protected]
From: <sip:[email protected]>;tag=1868922450
To: <sip:[email protected]>;tag=z9hG4bK1989114344
CSeq: 2030 REGISTER
Server: FPBX-16.0.26(18.13.0)
Content-Length:  0

[2022-12-21 15:02:39] WARNING[40600]: res_pjsip_registrar.c:1166 find_registrar_aor: AOR '' not found for endpoint 'LinhaMeo' (10.0.0.3:5062)

Some configuration screens, I don’t know which are relevant. Note that 10.0.0.10 is the FreePBX.

From what I gather from your replies, it seems…

The error is caused by “initiative” of the HT813, trying to register with FreePBX, but FreePBX is not happy and rejects it
I am probably quite confused about what entity (trunk? extension? user?) I should be configuring for this on the FreePBX. I think I only have a trunk. Should I create some other entity so that FreePBX is happy with the registration request?

BTW, I have the calls working in both directions, which is curious…

I also see this old thread, I don’t know if it is related:

pgr · December 21, 2022, 4:15pm

I think I solved this - I changed two things, I don’t know which one made the difference:

Changed trunk name to numeric following this advice.
In trunk PJSIP settings changed registration to None. The calls still go through fine.

I’d still appreciate any further educational advice you have for me.

david55 · December 21, 2022, 4:22pm

You have FreePBX set to send registrations, but whatever is generating the complaint is expecting it to be set to receive them.

You either need to disable registration in the gateway, and set Asterisk for no registration, or set registration to Receive, in Asterisk, and remove the static addresses.

Trunk and extension are FreePBX terms. Extension, for SIP, is something that looks like a phone and trunk is something that looks like a PABX or exchange. I forget the exact significance of user to FreePBX,

If receiving registration, the trunk name needs to be the HT813’s SIP User ID, but that need not be numeric.

system · December 28, 2022, 4:23pm

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.