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

[Alexey A. Astashov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25648?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228771#comment-228771 ] 

Alexey A. Astashov edited comment on ASTERISK-25648 at 12/29/15 1:45 PM:
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Files containing debug information, I made a call from PBX 172.16.10.173 to 172.16.15.194 through 172.16.15.196, вut everything else like in the picture.

Oh, I almost forgot. Unless the PBX (for which a call comes in) remove the subscriber number which can come at CID, the calls begin to work normally.


was (Author: alexey_astashov):
Files containing debug information, I made a call from PBX 172.16.10.173 to 172.16.15.194 through 172.16.15.196, вut everything else like in the picture.

> 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
>
>
> 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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