[asterisk-bugs] [JIRA] (ASTERISK-14731) [patch] sip session timer: Does not work if initial INVITE min-se timer is too small

[Corey Farrell (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-14731?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223428#comment-223428 ] 

Corey Farrell commented on ASTERISK-14731:
------------------------------------------

I believe this is related to ASTERISK-23373.  The channel/RTP leaks should be resolved by this.  Is the REINVITE issue resolved?  Note 1.6 is out of support, so you need to be running 1.8.26.1, 11.8.1, 12.1.1 or higher.

> [patch] sip session timer: Does not work if initial INVITE min-se timer is too small
> ------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-14731
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-14731
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/General
>            Reporter: Johann Steinwendtner
>            Severity: Minor
>         Attachments: full_sip_session-timer.gz, session_timer_422.patch
>
>
> Asterisk sip with session timer enabled.
> sip.conf:
> session-timers=accept
> session-expires=600
> session-minse=180
> Patton box connected to asterisk. Patton sends INVITE with session timer 90
> asterisk responds with 422 session interval too small
> patton reinvites with the proposed session timer.
> asterisk send 200 ok, nothing happens. no tones or anything.
> When patton sends BYE, asterisk sends ACK
> But
> sip channels remains, audio ports are not released
> voip-1*CLI> sip show channels
> Peer             User/ANR    Call ID          Format           Hold     Last Message   
> 91.128.104.50    (None)      302e3db1464e650  0x0 (nothing)    No       Rx: OPTIONS               
> 91.128.104.50    test_user   9e2ec18f1622d61  0x8 (alaw)       No       Rx: BYE                   
> 2 active SIP dialogs
> voip-1*CLI> 
> voip-1*CLI> core show channels
> Channel              Location             State   Application(Data)             
> SIP/test_user-b7     01229922640 at from_sip Down    (None)                        
> 1 active channel
> 0 active calls
> 0 calls processed
> voip-1*CLI> 



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