All Circuits are Busy

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.

The 401 Unauthorized is not an error – it’s a request for authentication. Please paste the complete Asterisk log for the call (including pjsip logger) at pastebin.com and post the last eight characters of the URL.

So I enable the logger and the log is at /var/log/asterisk correct?

Yes, just like you already posted, but it will likely be a few hundred lines.

HJnF4kTn

Line 2251: SIP/2.0 404 Not Found

While something is wrong (you should hear ‘number not in service’ or ‘call cannot be completed as dialed’ rather than ‘all circuits are busy’), it does appear that the number is invalid. Calling from my own system, I hear “This number is not in use. Thank you for calling. Goodbye.” Looking up 8888051752 at Search Who Owns, I get an error. Calling from my mobile (on Mint), the call is silently dropped.

Try calling another toll-free, such as 800-437-7950 .

Yeah that worked

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