[asterisk-bugs] [JIRA] (ASTERISK-26523) Asterisk 13.12.1 cuts incoming calls after 2 minutes

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26523?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233088#comment-233088 ] 

Asterisk Team commented on ASTERISK-26523:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> Asterisk 13.12.1 cuts incoming calls after 2 minutes
> ----------------------------------------------------
>
>                 Key: ASTERISK-26523
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26523
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.12.0, 13.12.1
>         Environment: AstLinux 1.2.7 64-bit, Asterisk 13.12.1
>            Reporter: Michael Keuter
>
> Asterisk 13.12.1 with chan_sip cuts incoming calls after 2 minutes. This does not happen with 13.11.2. This commit is related to that problem:
> [http://git.asterisk.org/gitweb/?p=asterisk/asterisk.git;a=commit;h=93332cb1d0eea18021ea6538237297e627d6e2fc]
> After reverting this commit the problem is fixed.
> {code}
>   -- Connected line update to SIP/berofix-pstn-00000017 prevented.
>    -- SIP/28_yeal52p-0000001c answered SIP/berofix-pstn-00000017
>    -- Channel SIP/28_yeal52p-0000001c joined 'simple_bridge' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>
>    -- Channel SIP/berofix-pstn-00000017 joined 'simple_bridge' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>
> 2 minutes later:
> [2016-10-29 12:51:08] NOTICE[25945]: chan_sip.c:29402 check_rtp_timeout: Disconnecting call 'SIP/berofix-pstn-00000017' for lack of RTP activity in 121 seconds
> [2016-10-29 12:51:08] NOTICE[25945]: chan_sip.c:29402 check_rtp_timeout: Disconnecting call 'SIP/28_yeal52p-0000001c' for lack of RTP activity in 121 seconds
>    -- Channel SIP/28_yeal52p-0000001c left 'native_rtp' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>
>    -- Channel SIP/berofix-pstn-00000017 left 'native_rtp' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>
>  == Spawn extension (privat_standard, s, 17) exited non-zero on 'SIP/berofix-pstn-00000017'
> {code}



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