<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Again remove <div><br></div><div><anti-action application="set" data="sip_exclude_contact=${network_addr}"/> </div><div><br></div><div>/b</div><div><br></div><div><div><html>On Apr 17, 2008, at 3:16 AM, Pete Kay wrote:</html><br class="Apple-interchange-newline"><blockquote type="cite">Hi,<br><br>I am working on some test on seeing how I can port my exist Asterisk stuff to Freeswitch. I am just getting started and I am hoping someone can give me some help to get started. <br><br>I installed with all the default config and xml setting. Then, I bring up two SiP clients - one in the same machine as freeswitch (<a href="http://192.168.1.104">192.168.1.104</a>) and the other one on another machine ( <a href="http://192.168.1.102">192.168.1.102</a>).<br> <br>When I dial an extension ( 1001, or 1002... etc ) from my SIP client on <a href="http://192.168.1.102">192.168.1.102</a>, I can make the call to the other client no problem. <br>2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/<a href="http://1002@192.168.1.104:5060">1002@192.168.1.104:5060</a> [9e6d6146-0c98-11dd-bb92-f1e303d528b1]<br> 2008-04-18 00:09:12 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1002-><span style="color: rgb(255, 0, 0);">81001@default</span><br>2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a> [9e93e564-0c98-11dd-bb92-f1e303d528b1]<br> 2008-04-18 00:09:12 [NOTICE] sofia.c:1603 sofia_handle_sip_i_state() Ring-Ready sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a>!<br><br>However, when I dial extension from the other SIP client, the one on <a href="http://192.168.1.104">192.168.1.104</a>, the call can't be routed. <br> <br>2008-04-18 00:10:34 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a> [cf710b58-0c98-11dd-bb92-f1e303d528b1]<br>2008-04-18 00:10:34 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1001-><span style="color: rgb(255, 0, 0);">1002@public</span><br> 2008-04-18 00:10:34 [NOTICE] switch_core_state_machine.c:198 switch_core_standard_on_execute() Hangup sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a> [CS_EXECUTE] [NORMAL_CLEARING]<br>2008-04-18 00:10:34 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 20 (sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a>) Ended<br> <br>It seems like the the call is being routed to the wrong context. How come this happens? I am using the standard default config xml files. Can anyone please help me?<br><br>With freeswitch, is there anyway to debug/trace the processing of the call so I can see which condition it is in, and where it is routed to? That way, I can debug the config easier? <br> <br>Thanks alot for all your inputs and help.<br><br>Regards,<br>Pete<br> _______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></body></html>