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. /b On Oct 10, 2008, at 11:20 AM, gary wrote:
Hello, 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)
2008-10-10 11:41:02 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/ 1001@10.1.1.117:51029;transport=udp;fs_nat=yes [bfe75310-638d-48df-8f3d-262143c15b22] 2008-10-10 11:41:02 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 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] 2008-10-10 11:41:02 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/ 1001@10.1.1.117:51029;transport=udp;fs_nat=yes) Ended 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1001@10.1.1.117 :51029;transport=udp;fs_nat=yes [CS_HANGUP] 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.
Can anybody tell me the cause?
Gary ___________