Cannot place call to feature code - "rejected because extension not found in context 'from-internal'

[elias]

Hi!

I’m trying to setup a FreePBX instance, and have managed to register two phones and place calls between them.

It’s running Asterisk 20.5.2, FreePBX 16.0.40.7.

However, I cannot call any feature codes or misc applications - I always get the following log entry in the console:
[2024-01-14 11:00:49] NOTICE[102876]: res_pjsip_session.c:4027 new_invite: 901: Call (TCP:192.168.8.42:35485) to extension ‘333’ rejected because extension not found in context ‘from-internal’.

“333” in the above log message is configured as a “misc application” that plays the hello world annoncement. But I can’t seem to call any other feature codes either.

Does anyone have any pointers?

[Stewart1 (Stewart)]

From the Asterisk command prompt, what is the output of
dialplan show 333@from-internal

[elias]

minas-tirith*CLI> dialplan show 333@from-internal
There is no existence of 'from-internal' context

[elias]

And this is the SIP-log on the phone:

Request

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 192.168.8.42:5060;branch=z9hG4bK2035319967;rport;alias
From: "901" <sip:[email protected]>;tag=2010729680
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 200 INVITE
Contact: "901" <sip:[email protected]:5060;transport=tcp>
Max-Forwards: 70
User-Agent: Grandstream HT801 1.0.45.2
Privacy: none
P-Preferred-Identity: "901" <sip:[email protected]>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=0C-9D-92-79-71-EC
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-BE-B7-28
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   475


v=0
o=901 8000 8000 IN IP4 192.168.8.42
s=SIP Call
c=IN IP4 192.168.8.42
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97 123 9 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54

Response

SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 192.168.8.42:5060;rport=35485;received=192.168.8.42;branch=z9hG4bK2035319967;alias
Call-ID: [email protected]
From: "901" <sip:[email protected]>;tag=2010729680
To: <sip:[email protected]>;tag=z9hG4bK2035319967
CSeq: 200 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1705233800/56cd41a939153e48808326782cde950f",opaque="7b8ed2207bb48d6e",algorithm=MD5,qop="auth"
Server: FPBX-16.0.40.7(20.5.2)
Content-Length:  0

[elias]

Nevermind - after restarting the whole asterisk/freepbx system, it works.
Apparently something in the config did not stick.

[Stewart1 (Stewart)]

Deleted