[Freeswitch-users] Old hung calls

[DJB International]

Peter,

Thank you.  Do I also need <param name="session-timeout" value="1800"/> as
well; otherwise, was it the default session-timeout is 0 as I can see it in
sofia status: SESSION-TO 0

-djbinter


On Sat, Mar 5, 2011 at 1:06 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> 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] f&#246;r 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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