[asterisk-bugs] [JIRA] (ASTERISK-22314) Failure in canceling a call, sending OK to wrong port

[Karsten Wemheuer (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22314?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=209316#comment-209316 ] 

Karsten Wemheuer commented on ASTERISK-22314:
---------------------------------------------

I've checked the patch from issue [ASTERISK-22071] as Michael told. It solves my problem. The sip communications seems ok (as it was in 1.8.22). 
                
> Failure in canceling a call, sending OK to wrong port
> -----------------------------------------------------
>
>                 Key: ASTERISK-22314
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22314
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.23.0
>         Environment: Debian 7.0, OpenSIPS 1.8.3
>            Reporter: Karsten Wemheuer
>            Assignee: Karsten Wemheuer
>         Attachments: call-asterisk_1.8.22.txt, call-asterisk_1.8.23.txt, debug
>
>
> I've' got a problem with asterisk 1.8.23. The same scenario is working fine in 1.8.22.
> Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on the same LAN, no NAT.
> Phone A (Account fred) calls 456. The call is sent from the phone via the proxy (opensips) to asterisk. Asterisk calls Phone B (Account hans). The call is sent through the proxy to the second phone. The call is accepted. If one user hangs up the phone (e.g. hans), a BYE_Request is sent to asterisk. In version 1.8.22 asterisk sends a 200 OK to the phone. In version 1.8.23 asterisk send the 200 OK to the proxy. The proxy ignores the message, because of a missing second via header. The phone repeats the BYE request several times.
> Attached is the debug output of asterisk and SIP-traces (ngrep) of the scenario, using asterisk version 1.8.22 (which is working) and version 1.8.23, which is not working. I wrote a mark in the traces where the issue happens ("===> This packet is sent from asterisk")
> Asterisk is on 192.168.10.70, port 25060. OpenSIPS is on 192.168.10.70 port 5060. Phone A (Account fred) is on 192.168.10.200, Phone B (Account hans) is on 192.168.10.201.

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