i tried it and it wont work with rtcachefriend=yes<br><br><div class="gmail_quote">On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson <span dir="ltr"><<a href="mailto:jmr.richardson@gmail.com">jmr.richardson@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">> I am facing an issue with Peer registration in my asterisk server .<br>
><br>
> I am using asterisk version 1.8.5.0 and using SIP real-time<br>
> architecture.when i am doing registration it registered fine on asterisk<br>
> as peer is available in Database.<br>
><br>
> But now i am doing 'sip reload' or 'reload' due to some reason my peer<br>
> registration is going out and i cannot able to call that peer even though<br>
> in SIP client it shows me 'registered'.<br>
><br>
> Can any body elaborate on this issue which settings i need to put in<br>
> sip.conf.<br>
><br>
> I also tried to follow this patch<br>
> <a href="https://issues.asterisk.org/view.php?id=14196" target="_blank">https://issues.asterisk.org/view.php?id=14196</a> But it allready applied in<br>
> code base so why it wont work?<br>
><br>
> Here is my sip.conf settings.<br>
><br>
> [general]<br>
> context=from-internal ; Default context for incoming cal<br>
> rtcachefriends=no<br>
> rtupdate=yes<br>
> rtautoclear=yes<br>
> rtsavesysname=yes<br>
> callcounter = yes<br>
> callevents=yes<br>
> bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)<br>
> srvlookup=yes ; Enable DNS SRV lookups on outbound calls<br>
> pedantic=yes ; Enable slow, pedantic checking for Pingtel<br>
> tos=184 ; Set IP QoS to either a keyword or numeric val<br>
> tos_sip=cs3 ; Sets TOS for SIP packets.<br>
> tos_audio=ef ; Sets TOS for RTP audio packets.<br>
> tos=lowdelay ; lowdelay,throughput,reliability,mincost,none<br>
> maxexpiry=3600 ; Max length of incoming registration we allow<br>
> defaultexpiry=120 ; Default length of incoming/outoing registration<br>
> preferred_codec_only=yes<br>
> disallow=all ; First disallow all codecs<br>
> allow=ulaw ; Allow codecs in order of preference<br>
> allow=alaw<br>
> insecure=invite<br>
> language=en ; Default language setting for all<br>
> users/peers<br>
> rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP<br>
> activity<br>
> useragent=dhaval ; Allows you to change the user agent string<br>
> dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default:<br>
> rfc2833<br>
> qualify=yes<br>
> nat=yes<br>
> ;canreinvite=yes<br>
> directmedia=yes<br>
> directrtpsetup=yes<br>
><br>
> And here is DB fields snapshots.<br>
><br>
> id: 1<br>
> name: 201<br>
> ipaddr: 172.18.100.243<br>
> port: 53624<br>
> regseconds: 1328716180<br>
> defaultuser: 201<br>
> fullcontact: NULL<br>
> regserver: dhaval<br>
> useragent: CSipSimple r1133 / b<br>
> lastms: 554<br>
> host: dynamic<br>
> type: friend<br>
> context: from-internal<br>
> permit: NULL<br>
> deny: NULL<br>
> secret: 201<br>
> md5secret: NULL<br>
> remotesecret: NULL<br>
> transport: NULL<br>
> dtmfmode: NULL<br>
> directmedia: yes<br>
> nat: NULL<br>
> allow: ulaw<br>
> disallow: g729<br>
> insecure: invite<br>
> callerid: NULL<br>
> rfc2833compensate: NULL<br>
> mailbox: NULL<br>
> session-timers: NULL<br>
> session-expires: NULL<br>
> session-minse: NULL<br>
> session-refresher: NULL<br>
><br>
> Kindly help me to resolve this.<br>
><br>
> Thanks<br>
> Dhaval<br>
><br>
<br>
The first thing I would try is 'rtcachefriends=yes', that should do it.<br>
<br>
JR<br>
--<br>
JR Richardson<br>
Engineering for the Masses<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>