[asterisk-bugs] [JIRA] (ASTERISK-25372) SIP/2.0 401 Unauthorized for incoming calls.

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25372?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227536#comment-227536 ] 

Rusty Newton commented on ASTERISK-25372:
-----------------------------------------

bq. Thanks a ton for your suggestion. It worked without the insecure=invite,port and removing the secret from the extensions. 

Great! Glad to help out.

bq. This was my first query and i never thought i would ever get a reply for this, but just feel good that the team took time to go through this and reply. I thought this is a bug, i am really sorry for that, below is my explanation. I have worked in other companies as an engineer where getting a reply on bug submission to an engineer from the same company takes weeks. And this is quite awesome as an Opensource thingy. I hope i am able to contribute more to the team.

We are glad to have you in the community! The speed of a response in opensource communities often depends on whether someone is interested or not due to the nature of thing. We of course try to get things taken care of quickly where possible. There are plenty of ways to help out.. if you are looking for something to do please ask on #asterisk-dev on irc.freenode.net or the asterisk-dev mailing list.

bq. Why i felt it is being rejected by the Asterisk server directly, but not by the peer is because it was Asterisk sending "SIP/2.0 401 Unauthorized" directly for the Invite sent by the other PBX.
If this was because of the SIP extension, i would expect it later after communicating with the extension.
Don;t you think, the logs for the sip call is mis-leading and not as per RFC ? Makes us think, the other PBX requires authentication to initiate a call. Or maybe i might be still missing something important.

Asterisk is a Back to Back user agent. https://en.wikipedia.org/wiki/Back-to-back_user_agent. Asterisk is challenging the request directly and it is according to the RFC.

You may have noticed that [~rmudgett] has closed out the issue since there is no bug here.  If you have further questions about the call behavior for chan_sip or for the new chan_pjsip driver (in 13) the appropriate place would be on the Asterisk mailing lists and forums.

http://www.asterisk.org/community/discuss

Thanks again! Feel free to E-mail me directly if you ever have questions about the community or are looking for specific documentation - rnewton at digium dot com! We are looking forward to your contributions.


> SIP/2.0 401 Unauthorized for incoming calls.
> --------------------------------------------
>
>                 Key: ASTERISK-25372
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25372
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.19.0
>         Environment: SHMZ release 6.5 (Final)
> Linux knaufsappdc 2.6.32-431.el6.x86_64 
> Running on a Virtual Environment. VMWare
>            Reporter: Agasthian P
>            Assignee: Unassigned
>            Severity: Minor
>         Attachments: coretracing.txt, sipconffiles1, siptrace.txt
>
>
> Running Freepbx on top of Asterisk 11.19.0.
> SIP trunk between Cisco Call Manager and FreePbx Asterisk.
> All the incoming calls from the Cisco Call Manager through SIP trunk are getting the error "SIP/2.0 401 Unauthorized". The Cisco Call Manager SIP trunks are treated as peer to peer and do not need any kind of registration.
> Have checked all the settings required to not enable any kind of authentication, still it seems to be asking for authentication.
> Tried the forums first, and have taken care of the following settings :-
> 1. insecure=invite,port
> 2.qualify=yes
> 3. type=friend
> 4. allow sip guests= yes
> 5. Allow Anonymous Inbound SIP Calls=yes
> The sip logs and the sip configuration files are attached as files.
> Call Flow :-
> FreePBX(20.1.1.58) - CUCM(20.1.1.170)
> 2723 extension is in both the systems. And 2723(CUCM) is trying to call 2723(Asterisk).



--
This message was sent by Atlassian JIRA
(v6.2#6252)