<div><span class="Apple-style-span" style="color: rgb(51, 102, 255);">Hello <span class="Apple-style-span" style="border-collapse: collapse; font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px; ">Chandrakant,</span></span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;"><br></span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;">Problem in your Asterisk configuration of sip.conf, </span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;">Please check the Trunk creation in sip.conf with Asterisk, </span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;">Where Kamailio trunk is registered in Asterisk and See whether you getting option Message to Kamailio Server from Asterisk Server. </span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;"><br></span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;">It should works fine, both scenarios </span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;"><br></span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse; color: rgb(51, 102, 255); font-size: 11px; white-space: pre-wrap; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;"><br></span></div>
<div><br></div><br clear="all">Thanks &Regards<br>Ravi Prakash Sunkara<br>VoIP Architect & JAVA-SIP Developer<br>+91-9999882776<br><br><a href="http://www.brainyquote.com/quotes/authors/b/bette_davis.html" target="_blank">Bette Davis</a> - "Brought up to respect the conventions, love had to end in marriage. I'm afraid it did."
<br><br><div class="gmail_quote">2009/5/29 Chandrakant Solanki <span dir="ltr"><<a href="mailto:solanki.chandrakant@gmail.com">solanki.chandrakant@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
hi<br><br>I am using.. <br><br>Asterisk : 1.6.0.5<br>Kamailio : 1.5.0<br><br><br>Here, is network diagram ...<br><br><a href="http://172.18.100.10/20/30" target="_blank">172.18.100.10/20/30</a> ============ 192.168.1.68 ========================== 192.168.1.70<br>
<br>(SIP Phone Register on this IP) (Kamailio IP) (Asterisk Server)<br><br>And here link for kamailio file<br>
<br><br>I have registered 111 and 222 user on asterisk (192.168.1.70)... and call to kamailio user (<a href="mailto:1212@domain.com" target="_blank">1212@domain.com</a>)... call established successfully.. but sip phone is not hangup..<br>
<br>As well as i call from kamailio user like 1212 to 2121 ... call established .. but sip phone not hangup..<br><br><br>Help me out....<br><br><br>Thanks in advance<br><br>-- <br>Regards,<br><font color="#888888"><br>Chandrakant Solanki<br>
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