[Freeswitch-users] External Profile Problem

[juan camilo ospina quintero]

Hi, 

im trying to establish a simple conference using freeswitch and sailfin, sailfin is
and application server that works with SipSevlets.
the all thing works as follow.

two softphone register with freeswitch, extension 1000 and 1001
1000 sends and invite to 1001, this invite goes to sailfin, i use this

    <extension name="redirect">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
      <condition field="destination_number" expression="^1(\d+)$">
    <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
      </condition>
    </extension>

this is in /freeswitch/conf/dialplan/default.xml


this far all goes well, the servlet receives the invite, and sends back the invite to freeswitch, i put this

    <extension name="receive">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>
      <condition field="destination_number" expression="^1(\d+)$">
        <action application="transfer" data="$0 XML default"/>
      </condition>
    </extension>

in /freeswitch/conf/dialplan/public.xml, but freeswitch returns this


------------------------------------------------------------------------
2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000 at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
send 632 bytes to udp/[192.168.2.9]:5070 at 01:14:29.517927:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
   From: <sip:1000 at 192.168.2.9>;tag=g4xfbi12-3
   To: <sip:1001 at 192.168.2.9:5080>;tag=4r91165pvcycB
   Call-Id: 192.168.153.1_3_3990383226484831353
   Cseq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1086 Session 9 (sofia/external/1000 at 192.168.2.9) Ended
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000 at 192.168.2.9 [CS_DESTROY]

i dont understand why i doesnt work if in public.xml, i tell to transfer the call if extension starts with 1 and the caller ip address is 192.168.2.9

please if someone can help me
 		 	   		  
_________________________________________________________________
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