[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:all-tabpanel ]

Rusty Newton updated ASTERISK-25648:
------------------------------------

    Assignee: Alexey A. Astashov  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

{quote}
username mismatch, have <1101>, digest has <001002>
{quote}

The username that is being matched against is 1101 - that is the peer that you need to check. The digest username should be 1101 if you want it to authenticate against 1101.

If the digest username is correct, then the problem is that Asterisk is matching against the wrong peer - probably due to the IP used since you are using insecure=port,invite.

> 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: Alexey A. Astashov
>         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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