[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=228935#comment-228935 ] 

Alexey A. Astashov edited comment on ASTERISK-25648 at 1/11/16 10:16 AM:
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 I do not want to check the authenticity of 1101, as end users are on both sides.

The authenticity of the circuit is checked at the server (peer) over IP.
The addresses are correct. And also with another type of protocol - it works.

The point is that, if at the final station, user 1101 is removed, the calls are fine.

The problem is that the external service provider cuts the all CID to 4 digits, and if the CID overlaps with one at my Asterisk 13.5,  the external provider receives a 403 error from my Asterisk.



was (Author: alexey_astashov):
 I do not want to check the authenticity of 1101, as end users are on both sides.

The authenticity of the circuit is checked at the server (peer) over IP.
The addresses are correct. And also with another type of protocol - it works.

The point is that, if at the final station, user 1101 is removed, the calls are fine.

> 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.



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