<div>Hi all</div>
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<div>Since long iam running with CDROUTER (0.814)</div>
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<div>Now i have got another server for testing with new Version of SER</div>
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<div>So i took the stable version from <a href="http://onsip.org">onsip.org</a></div>
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<div>and copied all the users and config files from old server to new server</div>
<div>and modified accoring to the new setup</div>
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<div>when the user not available or busy iam sending to voice mail ( asterisk another server)</div>
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<div>but i have new problem here with new version of ser is</div>
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<div>when ever i call from user X to user Y ( once its start ringing)</div>
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<div>if i disconnect the call on X Side, its going to voice mail. ( and other side iam able to see Y user connection hangup)</div>
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<div>same setup working with my old server with out any problem, </div>
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<div>and i ahve changed many things in the config, but no luck, </div>
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<div>so posting in group for the suggestions.</div>
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<div>OLD SERVER</div>
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<div>version: ser 0.8.12 (i386/linux)<br>flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT<br>ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
<br>@(#) $Id: main.c,v 1.168 2003/10/12 15:09:08 andrei Exp $<br>main.c compiled on 19:40:54 Jul 19 2004 with gcc 3.2<br> </div>
<div>-------------------------------------------</div>
<div>NEW SERVER</div>
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<div>version: ser 0.9.7-pre1 (i386/linux)<br>flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT<br>ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
<br>@(#) $Id: main.c,v <a href="http://1.197.2.1">1.197.2.1</a> 2005/07/25 16:56:24 andrei Exp $<br>main.c compiled on 15:49:21 Nov 3 2006 with gcc 3.4.6<br> </div>
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<div>here is my config</div>
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<div> # does the user wish redirection on no availability? (i.e., is he<br> # in the voicemail group?) -- determine it now and store it in<br> # flag 4, before we rewrite the flag using UsrLoc<br> if (is_user_in("Request-URI", "voicemail")) {
<br> setflag(4);<br> };</div>
<div>
<p> # native SIP destinations are handled using our USRLOC DB<br> if (!lookup("location")) {<br> # handle user which was not found<br> route(4);<br> break;
<br> };</p>
<p> # if user is on-line and is in voicemail group, enable redirection<br> if (method == "INVITE" || method=="BYE" || method=="ACK" && isflagset(4)) {<br> setflag(1); # for accounting
<br> t_on_failure("1");<br> };<br> t_relay();<br>}</p>
<p># ------------- handling of unavailable user ------------------</p>
<p>route[3]{<br> log(1,"route[3]:no user location: foward to voicemail");<br> rewritehostport("asteriskip:5090");<br> t_relay();<br>}<br> break;<br> }</p>
<p><br>route[4] {</p>
<p> # non-Voip -- just send "off-line"<br> if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) {<br> sl_send_reply("404", "Not Found");
<br> break;<br> };</p>
<p> # not voicemail subscriber<br> if (!isflagset(4)) {<br> sl_send_reply("404", "Not Found and no voicemail turned on");<br> break;<br> };</p>
<p> # forward to voicemail now<br> rewritehostport("asteriskip:5090");<br> t_relay_to_udp("asteriskip", "5090");<br>}</p>
<p>failure_route[1] {<br> revert_uri();<br> rewritehostport("asteriskip:5090");<br> append_branch();<br> t_relay_to_udp("asteriskip", "5090");<br>}<br></p>
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<p>any suggestion will be appriciated</p>
<p>Ram</p></div>