[Freeswitch-users] Old hung calls

[Peter Olsson]

Sorry, totally missed this :)

You should probably enable SIP Session timers, check out http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer

/Peter
________________________________________
Från: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för DJB International [djbinter at gmail.com]
Skickat: den 5 mars 2011 21:46
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] Old hung calls

Just for your info, I am running in media bypass mode.

In my sip profile:
<param name="inbound-bypass-media" value="true"/>

Thank you,
-djbinter


On Sat, Mar 5, 2011 at 11:21 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Since RTP timeout is set to 5 minutes (300 sec) FS should clear the call after this (and of course the record in core.db as well).

You probably need to pastebin some logs with examples when this happens, something seems strange here...

/Peter
________________________________________
Från: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r DJB International [djbinter at gmail.com<mailto:djbinter at gmail.com>]
Skickat: den 5 mars 2011 18:49
Till: FREESWITCH-USERS MAILING LIST
Ämne: [Freeswitch-users] Old hung calls

I am running FS git-deec244 2011-02-24 10-22-47 -0600.  I ran into a problem the other day when our ISP was down badly for couple hours (fortunately after midnight).   The problem was that after the internet went back up; there were so many old hung calls before the internet got cut off in the core.db (calls & channels table) database.

Here are my questions:
- What is the proper way to clean those calls up, or will FS clean those calls out from database automatically?
- If I decided to restart FS which I know it will clean up those calls, will those calls get written to mod_cdr_csv?

Here is my sip profile:
 <settings>
   <param name="debug" value="0"/>
   <param name="sip-trace" value="no"/>
   <param name="rfc2833-pt" value="101"/>
   <param name="sip-port" value="5060"/>
   <param name="dialplan" value="XML"/>
   <param name="context" value="routing"/>
   <param name="dtmf-duration" value="2000"/>
   <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
   <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
   <param name="hold-music" value="$${hold_music}"/>
   <param name="rtp-timer-name" value="soft"/>
   <param name="local-network-acl" value="localnet.auto"/>
   <param name="manage-presence" value="false"/>
   <param name="inbound-codec-negotiation" value="generous"/>
   <param name="nonce-ttl" value="60"/>
   <param name="auth-calls" value="$${internal_auth_calls}"/>
   <param name="rtp-ip" value="$${local_ip_v4}"/>
   <param name="sip-ip" value="$${local_ip_v4}"/>
   <param name="ext-rtp-ip" value="auto-nat"/>
   <param name="ext-sip-ip" value="auto-nat"/>
   <param name="rtp-timeout-sec" value="300"/>
   <param name="rtp-hold-timeout-sec" value="1800"/>
   <param name="auth-all-packets" value="false"/>
   <param name="disable-transfer" value="true"/>
   <param name="disable-register" value="true"/>
   <param name="inbound-late-negotiation" value="true"/>
   <param name="inbound-bypass-media" value="true"/>
   <param name="apply-inbound-acl" value="domains"/>
 </settings>

Thank you,
-djbinter


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