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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25372?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227456#comment-227456 ] 

Asterisk Team commented on ASTERISK-25372:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> 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 
>            Reporter: Agasthian P
>            Severity: Minor
>
> 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.



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