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">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">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><br>Chandrakant Solanki<br>