Ext 200 always returns busy when called by 203, any suggestions?
I need to sanitize the logs before posting
Asterisk Full log
It looks like the extension was really called, answered and huing up. You need to enable protocol logging with the CLI command “pjsip set logger on”, but one cause is incompatible codecs.
Disallowed Codecs /Allowed Codecs is blank in the extension settings
Nobody would be doing that
Phones do, though, and will do if they are presented with an unacceptable offer too late to reject it without answering.
That sounds wrong to me. You should disallow everything and then allow just the 1, 2, or 3 that it is sensible to use; Note that allowing all codecs can result in truncated packets which could also cause problems.
I have never used those fields but anyway it would not explain this problem as all the other extensions are identical and the problem only occurs when 203 dials 200.
SIP/2.0 486 Busy Here
Any other extension can call 200
Please recreate the logs, but with “pjsip set logger on” in effect.
Pastebin here
Check lines 205 to 214
[0K<— Transmitting SIP response (493 bytes) to UDP:TEST.EXT.IP.ADD:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.198:5060;rport=5060;received=TEST.EXT.IP.ADD;branch=z9hG4bK29ed53c75449dfe545d63
Call-ID: [email protected]
From: “203” sip:[email protected];tag=657916278
To: sip:[email protected];tag=z9hG4bK29ed53c75449dfe545d63
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1734236550/5ee7d3f343e46c6877ae88fa95406120”,opaque=“33fa6de3438b5dec”,algorithm=md5,qop=“auth”
Server: FPBX-16.0.40.11(18.9)
Content-Length: 0
Why would that “nonce” be occurring?
And BTW thanks for taking the time to look at this
Because it is part of the authentication protocol. It ensures that password hashes cannot be re-used at a later time.
If you are asking why is 401 being sent, it is either because it recognizes the requestor, and requires proof of knowledge of the password from them, or because it doesn’t recognize them.
[0K<--- Received SIP response (565 bytes) from UDP:EXT.200.IP.ADD:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP PBX.SRV.IP.ADD:5060;rport=5060;branch=z9hG4bKPj1fe85ffe-6355-4f1b-a2da-7a2852d4699b
From: "Management Office 2" <sip:[email protected]>;tag=9f5c2ac6-30cc-499c-bc85-559bf2616532
To: <sip:[email protected]>;tag=2625488010
Call-ID: f528d15c-244b-4b99-9103-2be7ca2cde81
CSeq: 14989 INVITE
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T44U 108.86.0.90
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0
The phone itself is returning a busy.
I’m guessing that 203 was accidentally blocked in the 200 phone.
Yealink SIP-T46U – Blocking Calls With the Blocklist Directory : Teleboard | Support Center.
I was starting to think of that possibility late last night, ironically the person at 203 is in charge of the people who are using 200
Now to see if can fix that remote
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