[asterisk-bugs] [JIRA] (ASTERISK-25648) chan_sip returns forbidden 403, if the incoming number was determined as the present.

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25648?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228845#comment-228845 ] 

Rusty Newton commented on ASTERISK-25648:
-----------------------------------------

{noformat}
[2015-12-29 22:01:15] WARNING[12881][C-00000002]: chan_sip.c:16653 check_auth: username mismatch, have <1101>, digest has <001002>
[2015-12-29 22:01:15] NOTICE[12881][C-00000002]: chan_sip.c:25550 handle_request_invite: Failed to authenticate device "TEST 123" <sip:1101 at 172.16.15.196>;tag=as17bda15c
{noformat}

Looks like an authentication problem. If you don't intend to have authentication happen then you will want to double check your SIP peer/friend configuration.

This doesn't appear to be a bug unless you can demonstrate that it is attempting to authenticate despite proper configuration. You didn't attach configuration as requested so I don't see any configuration to look at. If you post the configuration we can take a quick look at that but otherwise this is a support issue that should be posted to the forums or mailing lists and not the bug tracker.

> chan_sip returns forbidden 403, if the incoming number was determined as the present.
> -------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25648
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25648
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.5.0, 13.6.0
>            Reporter: Alexey A. Astashov
>            Assignee: Rusty Newton
>         Attachments: Debug-GW.txt, Debug-Users-Asterisk.txt, incall.cap, Initial-PBX-call.txt, Truble chan_sip.jpg, Users-asteriskmini.txt
>
>
> I detected a problem with the call processing protocol SIP.
> For example:
> "Some PBX" (num's 1100-1299) --> call came to my GW Asterisk with internal CID "Some PBX" --> then call routed to my PBX Asterisk (num's 1100-1500), but last determine existing number and return Forbidden 403. 
> In configuration TRUNK on My PBX I have insecure=port,invite
> The error is that if the final PBX will see that an incoming call comes CID number that it has, it sends to the gateway error 403. The error was discovered with 13 versions of Asterisk, on Asterisk 11 - everything worked well. At the same time the IAX2 protocol, this is not a problem. Unfortunately, I can not test the functionality of the protocol PJSIP.



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