Peter,<div><br></div><div>Thank you. Do I also need <span class="Apple-style-span" style="font-family: monospace; font-size: 13px; line-height: 14px; white-space: pre; "><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</span></div>
<div><font class="Apple-style-span" face="monospace"><span class="Apple-style-span" style="line-height: 14px; white-space: pre;"><span class="Apple-style-span" style="font-family: arial; line-height: normal; white-space: normal; "><br>
</span></span></font></div><div>-djbinter</div><div><font class="Apple-style-span" face="monospace"><span class="Apple-style-span" style="line-height: 14px; white-space: pre;"><br></span></font><br><div class="gmail_quote">
On Sat, Mar 5, 2011 at 1:06 PM, Peter Olsson <span dir="ltr"><<a href="mailto:peter.olsson@visionutveckling.se">peter.olsson@visionutveckling.se</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Sorry, totally missed this :)<br>
<br>
You should probably enable SIP Session timers, check out <a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer" target="_blank">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer</a><br>
<div class="im"><br>
/Peter<br>
________________________________________<br>
Från: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] f&#246;r DJB International [<a href="mailto:djbinter@gmail.com">djbinter@gmail.com</a>]<br>
</div>Skickat: den 5 mars 2011 21:46<br>
Till: FreeSWITCH Users Help<br>
Ämne: Re: [Freeswitch-users] Old hung calls<br>
<div class="im"><br>
Just for your info, I am running in media bypass mode.<br>
<br>
In my sip profile:<br>
<param name="inbound-bypass-media" value="true"/><br>
<br>
Thank you,<br>
-djbinter<br>
<br>
<br>
</div><div class="im">On Sat, Mar 5, 2011 at 11:21 AM, Peter Olsson <<a href="mailto:peter.olsson@visionutveckling.se">peter.olsson@visionutveckling.se</a><mailto:<a href="mailto:peter.olsson@visionutveckling.se">peter.olsson@visionutveckling.se</a>>> wrote:<br>
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).<br>
<br>
You probably need to pastebin some logs with examples when this happens, something seems strange here...<br>
<br>
/Peter<br>
________________________________________<br>
</div>Från: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>>] f&#246;r DJB International [<a href="mailto:djbinter@gmail.com">djbinter@gmail.com</a><mailto:<a href="mailto:djbinter@gmail.com">djbinter@gmail.com</a>>]<br>
<div><div></div><div class="h5">Skickat: den 5 mars 2011 18:49<br>
Till: FREESWITCH-USERS MAILING LIST<br>
Ämne: [Freeswitch-users] Old hung calls<br>
<br>
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.<br>
<br>
Here are my questions:<br>
- What is the proper way to clean those calls up, or will FS clean those calls out from database automatically?<br>
- If I decided to restart FS which I know it will clean up those calls, will those calls get written to mod_cdr_csv?<br>
<br>
Here is my sip profile:<br>
<settings><br>
<param name="debug" value="0"/><br>
<param name="sip-trace" value="no"/><br>
<param name="rfc2833-pt" value="101"/><br>
<param name="sip-port" value="5060"/><br>
<param name="dialplan" value="XML"/><br>
<param name="context" value="routing"/><br>
<param name="dtmf-duration" value="2000"/><br>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/><br>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/><br>
<param name="hold-music" value="$${hold_music}"/><br>
<param name="rtp-timer-name" value="soft"/><br>
<param name="local-network-acl" value="localnet.auto"/><br>
<param name="manage-presence" value="false"/><br>
<param name="inbound-codec-negotiation" value="generous"/><br>
<param name="nonce-ttl" value="60"/><br>
<param name="auth-calls" value="$${internal_auth_calls}"/><br>
<param name="rtp-ip" value="$${local_ip_v4}"/><br>
<param name="sip-ip" value="$${local_ip_v4}"/><br>
<param name="ext-rtp-ip" value="auto-nat"/><br>
<param name="ext-sip-ip" value="auto-nat"/><br>
<param name="rtp-timeout-sec" value="300"/><br>
<param name="rtp-hold-timeout-sec" value="1800"/><br>
<param name="auth-all-packets" value="false"/><br>
<param name="disable-transfer" value="true"/><br>
<param name="disable-register" value="true"/><br>
<param name="inbound-late-negotiation" value="true"/><br>
<param name="inbound-bypass-media" value="true"/><br>
<param name="apply-inbound-acl" value="domains"/><br>
</settings><br>
<br>
Thank you,<br>
-djbinter<br>
<br>
<br>
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