[asterisk-bugs] [JIRA] (ASTERISK-24773) Hung call in "sip show channels" not listed in "show channels"

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24773?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224837#comment-224837 ] 

Matt Jordan commented on ASTERISK-24773:
----------------------------------------

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.8 branch has ended. For continued maintenance support please move to the 11 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions.  After testing with the latest Asterisk 11 release, if you find this problem has not been resolved, please open a new issue against Asterisk 11.



> Hung call in "sip show channels" not listed in "show channels"
> --------------------------------------------------------------
>
>                 Key: ASTERISK-24773
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24773
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.15.0
>            Reporter: jessup ross
>
> We have a system with 415 active SIP dialogs but only 42 active channels
> 23 active calls.  Clearly we don't have that many calls in setup/tear-down.  These hung calls can't be hung up and they are rapidly eating our bandwidth and RTP ports (1000).  rtptimeout=60 is currently commented out, does this affect our issue or only with user=peer?
> The only way we have found to clean them is an asterisk restart and we can't do that in-production.  Is this a patch or mis-configuration issue?



--
This message was sent by Atlassian JIRA
(v6.2#6252)