Extension getting 401 error when calling another extension

Hello All,

I have this issue where an external extension gets registered and can receive a call with no issues, however when they call another extension they get a 401 error.
Note this worked for several months.

We tried several different things including deleting the extension and rebuilding it but no luck.
I decided to capture a debug, but I’m not seeing anything of value.

Any assistance appreciated!

<--- SIP read from UDP:134.23.6.211:37276 ---> INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 134.23.6.211:37276;branch=z9hG4bK454078054;rport Route: From: "system-201" ;tag=2086520258 To: Call-ID: [email protected] CSeq: 90 INVITE Contact: "system-201" X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.7.97 Privacy: none P-Preferred-Identity: "system-201" P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=D4-CA-6D-0B-D8-54 P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-5F-D8-00 Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 437

v=0
o=201 8000 8000 IN IP4 134.23.6.211
s=SIP Call
c=IN IP4 134.23.6.211
t=0 0
m=audio 36992 RTP/AVP 0 8 4 18 9 97 2 123 101
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:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:123 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (20 headers 20 lines) —
Sending to 134.23.6.211:37276 (NAT)
Sending to 134.23.6.211:37276 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘201’ for ‘201’ from 134.23.6.211:37276

<— Reliably Transmitting (NAT) to 134.23.6.211:37276 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 134.23.6.211:37276;branch=z9hG4bK454078054;received=134.23.6.211;rport=37276
From: “system-201” sip:[email protected]:5060;tag=2086520258
To: sip:[email protected]:5060;tag=as3a46cf9d
Call-ID: [email protected]
CSeq: 90 INVITE
Server: FPBX-13.0.192.9(13.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4b699320"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:134.23.6.211:37276 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 134.23.6.211:37276;branch=z9hG4bK454078054;rport
Route: sip:162.17.96.246:5060;lr
From: “system-201” sip:[email protected]:5060;tag=2086520258
To: sip:[email protected]:5060;tag=as3a46cf9d
Call-ID: [email protected]
CSeq: 90 ACK
Content-Length: 0

<------------->