So I’m not sure why I am getting this error when calling. I was able to call my domestic number just fine but when I tried to call a toll-free number I got that error. When I look into the asterisk log, I found this:
[2023-06-06 14:25:23] VERBOSE[1886] res_pjsip_logger.c: <--- Received SIP response (807 bytes) from UDP:143.244.44.217:57391 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147:5060;rport=5060;received=147;branch=z9hG4bKPjdb1976eb-e456-4728-ac81-ea3aaad66596
Call-ID: d9dba0f1-d78a-4864-af21-099ac3ef7dae
From: <sip:4001@147>;tag=c76a6ed0-a063-43bd-a2ac-de3adca6556f
To: <sip:[email protected];ob>;tag=z9hG4bKPjdb1976eb-e456-4728-ac81-ea3aaad66596
CSeq: 4847 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.21.3
Content-Length: 0
[2023-06-06 14:25:24] VERBOSE[1886] res_pjsip_logger.c: <--- Received SIP request (1008 bytes) from UDP:143.244.44.217:57391 --->
INVITE sip:8888051752@domain SIP/2.0
Via: SIP/2.0/UDP 10.171.166.246:57391;rport;branch=z9hG4bKPja17f0f0706bc45138f654fba903b6897
Max-Forwards: 70
From: "Name" <sip:4001@domain>;tag=f8a638648a0c4d85b2775505c95aa619
To: <sip:8888051752@domain>
Contact: "Judy Fox" <sip:[email protected]:57391;ob>
Call-ID: e24d18dd87984a72996598133d86755e
CSeq: 8134 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.3
Content-Type: application/sdp
Content-Length: 345
v=0
o=- 3895035931 3895035931 IN IP4 10.171.166.246
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 8 0 101
c=IN IP4 10.171.166.246
b=TIAS:64000
a=rtcp:4009 IN IP4 10.171.166.246
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:178860796 cname:14bf32a66b1e5a9f
[2023-06-06 14:25:24] VERBOSE[8145] res_pjsip_logger.c: <--- Transmitting SIP response (579 bytes) to UDP:143.244.44.217:57391 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.171.166.246:57391;rport=57391;received=143.244.44.217;branch=z9hG4bKPja17f0f0706bc45138f654fba903b6897
Call-ID: e24d18dd87984a72996598133d86755e
From: "Name" <sip:4001@domain>;tag=f8a638648a0c4d85b2775505c95aa619
To: <sip:8888051752@domain>;tag=z9hG4bKPja17f0f0706bc45138f654fba903b6897
CSeq: 8134 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1686061524/b02231ce72062f96e4db8e706363ebfa",opaque="3d7f6c8e3f21890b",algorithm=md5,qop="auth"
Server: FPBX-16.0.10.40(18.6.0)
Content-Length: 0
I use BulkVS for my trunking/DIDs.