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

Alexey A. Astashov (JIRA) noreply at issues.asterisk.org
Thu Jan 7 14:48:32 CST 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25648?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228886#comment-228886 ] 

Alexey A. Astashov edited comment on ASTERISK-25648 at 1/7/16 2:47 PM:
-----------------------------------------------------------------------

Hmm, I do not understand why the Asterisk 11 does not require re-examination, and it requires Asterisk 13.5(13.6) .. just enough to remove the extension (for examle 1101), and everything works well.
in this case are not important setting is "insecure" - is of particular importance CID party, and as soon as the final Asterisk receives the value corresponding to the extension number - an error 403.


was (Author: alexey_astashov):
Hmm, I do not understand why the 11 th Asterisk does not require re-examination, and it requires Asterisk 13th .. just enough to remove the extension (for examle 1101), and everything works well.
in this case are not important setting is "insecure" - is of particular importance CID party, and as soon as the final Asterisk receives the value corresponding to the extension number - an error 403.

> 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: Unassigned
>         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.



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



More information about the asterisk-bugs mailing list