Hi all i use freeswitch to register to sip server the domain is not real: sip.yeah.com the outbound proxy : 118.69.145.5 i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder </include> <gateway name="sip.yeah.com"> <param name="username" value="071234"/> <param name="from-user" value="071234"/> <param name="proxy" value="sip.yeah.com"/> <param name="username" value="071234"/> <param name="password" value="1234"/> <param name="register-proxy" value="118.69.145.5"/> <param name="register" value="true"/> <param name="retry-seconds" value="10"/> </gateway> </include> so i use xlite register to freeswitch with user 1000 i start freeswitch , freeswitch start ok, freeswitch register to sip server successfull with username 071234 xlite register to freeswitch successfull so i edit the public.xml for incomming call from sip server, i will forward to extension 1000 <extension name="sip.yeah.com"> <condition field="destination_number" expression="071234"> <action application="transfer" data="1000 XML default"/> </condition> </extension> but when i use xlite make call out from freeswitch to sip server and get no routing nta: received INVITE sip:071234@192.168.1.90;user=phone SIP/2.0 (CSeq 3621) nta: INVITE (3621) going to a default leg nua(0xb6e20f90): adding session usage nta: sent 100 Trying for INVITE (3621) nua(0xb6e20f90): event i_invite 100 Trying nua(0xb6e20f90): call state changed: init -> received, received offer nua(0xb6e20f90): event i_state 100 Trying 432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000@192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006] 446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default 455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000@192.168.1.90 to enum[071234@default] 566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting 566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000@192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION] nua(0xb6e20f90): recv signal r_respond 404 Not Found 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 please help, Thank you Hãy cùng chat với bạn bè trên Yahoo! Pingbox mới toanh! Tạo một không gian riêng để chat trên blog là chuyện nhỏ. http://vn.messenger.yahoo.com/pingbox/