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
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