[asterisk-bugs] [JIRA] (ASTERISK-22417) RTP ports left open after using click to dial application

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22417?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=209926#comment-209926 ] 

Rusty Newton commented on ASTERISK-22417:
-----------------------------------------

Patrick, after making those changes, does the RTP port leak still appear to exist?
                
> RTP ports left open after using click to dial application
> ---------------------------------------------------------
>
>                 Key: ASTERISK-22417
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22417
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.4.0
>         Environment: Linux 3.2.0-40-virtual #64-Ubuntu SMP i686 i686 i386 GNU/Linux, Ubuntu 12.04
>            Reporter: Patrick Beaumont
>
> Ports in use at the start
> {noformat}
> root at CN1:~# netstat -tunap
> Active Internet connections (servers and established)
> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
> tcp        0      0 0.0.0.0:5038            0.0.0.0:*               LISTEN      428/asterisk    
> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      313/lighttpd    
> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      129/sshd        
> tcp        0      0 10.0.3.101:22           10.0.3.1:60572          ESTABLISHED 339/sshd: steve [pr
> tcp6       0      0 :::22                   :::*                    LISTEN      129/sshd        
> udp        0      0 0.0.0.0:30000           0.0.0.0:*                           428/asterisk    
> udp        0      0 10.0.3.101:30001        0.0.0.0:*                           428/asterisk
> {noformat}
> First I initiate a call using dialer.exe through the application Siptapi.
> My desk phone rings.
> I pick it up.
> My mobile phone then rings (the destination I set in dialer.exe).
> I answer on my mobile phone.
> I place the call on hold using my desk phone.
> I initiate another call to my mobile using dialer.exe through the application Siptapi.
> My desk phone indicates I have a call waiting.
> I hang up the call I have on hold and my desk phone starts ringing.
> I pick up my desk phone and my mobile phone starts ringing.
> I answer the call on my mobile.
> I hang up the call on my mobile.
> I hang up my desk phone.
> Ports in use at the end:
> {noformat}
> root at CN1:~# netstat -tunap
> Active Internet connections (servers and established)
> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
> tcp        0      0 0.0.0.0:5038            0.0.0.0:*               LISTEN      428/asterisk    
> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      313/lighttpd    
> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      129/sshd        
> tcp        0      0 10.0.3.101:22           10.0.3.1:60572          ESTABLISHED 339/sshd: steve [pr
> tcp6       0      0 :::22                   :::*                    LISTEN      129/sshd        
> udp        0      0 0.0.0.0:30000           0.0.0.0:*                           428/asterisk    
> udp        0      0 10.0.3.101:30001        0.0.0.0:*                           428/asterisk    
> udp        0      0 10.0.3.101:30034        0.0.0.0:*                           428/asterisk    
> udp        0      0 10.0.3.101:30035        0.0.0.0:*                           428/asterisk  
> {noformat}
> This is very repeatable and if performed enough times Asterisk will complain about no longing being able to allocate RTP ports and will start rejecting calls.
> Performing a core restart appears to be the only way to release the ports.

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