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

[Michael Keuter (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26523?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Michael Keuter updated ASTERISK-26523:
--------------------------------------

    Description: 
Asterisk 13.12.1 with chan_sip cuts incoming calls (coming from my Berofix ISDN/SIP-gateway) 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. "rtptimeout" in sip.conf is set to 120 secs (the default is commented out ("off")).

{code}
   -- Called SIP/28_yeal52p
   -- Connected line update to SIP/berofix-pstn-00000017 prevented.
   -- SIP/28_yeal52p-0000001c is ringing
   -- 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}



  was:
Asterisk 13.12.1 with chan_sip cuts incoming calls (coming from my Berofix ISDN/SIP-gateway) 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. "rtptimeout" in sip.conf is set to 120 secs.

{code}
   -- Called SIP/28_yeal52p
   -- Connected line update to SIP/berofix-pstn-00000017 prevented.
   -- SIP/28_yeal52p-0000001c is ringing
   -- 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}




> 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, Linux 3.2.80 64-bit, Asterisk 13.12.1, Beronet Berofix ISDN/SIP-gateway (FW: 3.0.12), ISDN BRI line
>            Reporter: Michael Keuter
>            Assignee: Unassigned
>         Attachments: Asterisk-13.12.1-rtp-timeout.pcap, asterisk-full-log.txt, dialpan-part.txt, sip-conf-part-updated.txt
>
>
> Asterisk 13.12.1 with chan_sip cuts incoming calls (coming from my Berofix ISDN/SIP-gateway) 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. "rtptimeout" in sip.conf is set to 120 secs (the default is commented out ("off")).
> {code}
>    -- Called SIP/28_yeal52p
>    -- Connected line update to SIP/berofix-pstn-00000017 prevented.
>    -- SIP/28_yeal52p-0000001c is ringing
>    -- 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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