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

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25648?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228769#comment-228769 ] 

Matt Jordan commented on ASTERISK-25648:
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Thanks for the report and debug. However we also need protocol specific debug captured at the time of the issue. Please include the following:

* Asterisk log files generated using the instructions on the Asterisk wiki [1], with the appropriate protocol debug options enabled, e.g. 'pjsip set logger on' if the issue involves the chan_pjsip channel driver.
* Configuration information for the relevant channel driver, e.g. pjsip.conf.
* A wireshark compatible packet capture, captured at the same time as the Asterisk log output.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> 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
>         Attachments: Truble chan_sip.jpg
>
>
> I detected a problem with the call processing protocol SIP.
> For example:
> "Some Station" (num's 1100-1299) --> call came to my GW Asterisk with internal CID "Some Station" --> then call routed to my station Asterisk (num's 1100-1500), but last determine existing number and return Forbidden 403. 
> In configuration on My station 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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