<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">its looking in context default... not public.<div><br></div><div>/b</div><div><br><div><div>On Mar 29, 2010, at 11:33 PM, false wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><table cellspacing="0" cellpadding="0" border="0" style="position: static; z-index: auto; "><tbody><tr><td valign="top" style="font: inherit;">Hi all<br><br>i use freeswitch to register to sip server<br>the domain is not real: <a href="http://sip.yeah.com">sip.yeah.com</a><br>the outbound proxy : 118.69.145.5<br><br>i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder<br></include><br> <gateway name="<a href="http://sip.yeah.com">sip.yeah.com</a>"><br> <param name="username" value="071234"/><br> <param name="from-user" value="071234"/> <br> <param name="proxy" value="<a href="http://sip.yeah.com">sip.yeah.com</a>"/><br> <param name="username" value="071234"/><br> <param name="password" value="1234"/><br> <param name="register-proxy"
value="118.69.145.5"/><br> <param name="register" value="true"/><br> <param name="retry-seconds" value="10"/><br> </gateway><br></include><br><br>so i use xlite register to freeswitch with user 1000<br>i start freeswitch , freeswitch start ok,<br>freeswitch register to sip server successfull with username 071234<br>xlite register to freeswitch successfull<br><br>so i edit the public.xml<br>for incomming call from sip server, i will forward to extension 1000<br> <extension name="<a href="http://sip.yeah.com">sip.yeah.com</a>"><br> <condition field="destination_number" expression="071234"><br> <action application="transfer" data="1000 XML default"/><br> </condition><br> </extension><br><br>but when i use
xlite make call out from freeswitch to sip server and get no routing<br><br>nta: received INVITE <a href="sip:071234@192.168.1.90;user=phone">sip:071234@192.168.1.90;user=phone</a> SIP/2.0 (CSeq 3621)<br>nta: INVITE (3621) going to a default leg<br>nua(0xb6e20f90): adding session usage<br>nta: sent 100 Trying for INVITE (3621)<br>nua(0xb6e20f90): event i_invite 100 Trying<br>nua(0xb6e20f90): call state changed: init -> received, received offer<br>nua(0xb6e20f90): event i_state 100 Trying<br>432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000@192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006]<br>446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default<br>455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000@192.168.1.90 to enum[071234@default]<br>566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting<br>566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000@192.168.1.90 [CS_ROUTING]
[NO_ROUTE_DESTINATION]<br>nua(0xb6e20f90): recv signal r_respond 404 Not Found<br><br>i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch<br><br>please help, <br><br>Thank you<br><br></td></tr></tbody></table><br>
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