<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Can you join us on #freeswitch on irc.freenode.net? Press F8 and try again and see what it does you might also wanna try SVN Trunk.<div><br></div><div>/b</div><div><br></div><div><br><div><div>On Oct 10, 2008, at 11:20 AM, gary wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0; "><div><font face="Arial" size="2">Hello,</font></div><div><font face="Arial" size="2"> I am new to freeswitch. Just installed freeswitch-1.0.1 with default configurations on CentOS 4.6 with OpenVZ. I registered two Cisco7960 phones using static IP with default users(1001.xml, 1007.xml). The registration are OK, but when I try to call from each other, it immedially send to voicemail. The following is the log info from console: (I replaced the real IP with 10.1.1.117 in the log)</font></div><div><font face="Arial" size="2"></font> </div><div><font face="Arial" size="2">2008-10-10 11:41:02 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes">sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes</a><span class="Apple-converted-space"> </span>[bfe75310-638d-48df-8f3d-262143c15b22]<br>2008-10-10 11:41:02 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes">sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes</a><span class="Apple-converted-space"> </span>[CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]<br>2008-10-10 11:41:02 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE]<br>2008-10-10 11:41:02 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE<br>2008-10-10 11:41:02 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (<a href="mailto:sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes">sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes</a>) Ended<br>2008-10-10 11:41:02 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes">sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes</a><span class="Apple-converted-space"> </span>[CS_HANGUP]<br></font></div><div><font face="Arial" size="2">Is it possible the NAT causing problem? I am using static IP on both server and phones and do not really need NAT. If it is, how can I disable NAT on the FS server.</font></div><div><font face="Arial" size="2"></font> </div><div><font face="Arial" size="2"></font> </div><div><font face="Arial" size="2">Can anybody tell me the cause?</font></div><div><font face="Arial" size="2"></font> </div><div><font face="Arial" size="2">Gary</font></div><font face="Arial" size="2"></font>___________</span></blockquote></div><br></div></body></html>