nobody facing any issue with this or nobody using real time architecture<br><br><div class="gmail_quote">On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA <span dir="ltr"><<a href="mailto:dhaval.it01034@gmail.com">dhaval.it01034@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">Hi Group.<br><br>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 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 registration is going out and i cannot able to call that peer even though in SIP client it shows me 'registered'.<br>
<br>Can any body elaborate on this issue which settings i need to put in sip.conf. <br><br>I also tried to follow this patch <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 code base so why it wont work?<br>
<br>Here is my sip.conf settings.<br><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 users/peers<br>rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity<br>
useragent=dhaval ; Allows you to change the user agent string<br>dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: 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><br>Kindly help me to resolve this.<br><br>Thanks<span class="HOEnZb"><font color="#888888"><br>Dhaval<br><br>
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