sir i will call a FS(extension 600) from my sip phone(extension 423)  i used a gateway  when the call is ringing i will hear a short msg ,
this msg  is a pcap file  ,when i changed it to wav file automatically in freeswitch it has  changed to the first codec selected in my soft phone :

my pcap file has us codec g711
my phoner lite i have selected 1/opus 2/ g711A 3./.........

but when i play this wav  i have in tshark this line:
8533 960.577294  192.168.3.5 -> 192.168.3.1  RTP 109 PT=opus, SSRC=0x57A054EC, Seq=2363, Time=424320

my dialplan/default.xml:
<extension name="600">
       <condition field="caller_id_number" expression="^(10[01][0-9]|1500|423|400|1011)$">
            <action application="set" data="continue_on_fail=true"/>
            <action application="ring_ready"/>
            <action application="set" data="ringback=${us-ring}"/>
            <action application="set" data="transfer_ringback=$${hold_music}"/>
            <action application="set" data="ignore_early_media=false"/>
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="bridge"  data="[leg_timeout=20,absolute_codec_string=PCMA ]sofia/gateway/gateway_to_600/600"/>
            <action application="answer"/>

      </condition>
  </extension>

     <extension name="600">
       <condition field="destination_number" expression="^(600)$">
             <action application="export" data="nolocal:codec_string=PCMA"/> 
            <action application="answer"/>
           <action application="info"/>
            <action application="sleep" data="1000"/>
            <action application="playback" data="/home/amani/projet/PFE/gg711.pcap.0xd2bd4e3e.wav"/>
         </condition>
</extension>



i have us tcpdump :
14:59:31.659246 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600@192.168.3.5 SIP/2.0
14:59:31.659773 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying
14:59:31.661679 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 407 Proxy Authentication Required
14:59:31.662289 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600@192.168.3.5 SIP/2.0
14:59:31.662726 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600@192.168.3.5 SIP/2.0
14:59:31.663177 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying
14:59:38.508070 IP 192.168.3.5.sip > 192.168.3.1.35200: Flags [.], ack 2002, win 789, length 0
14:59:38.508188 IP 192.168.3.1.35200 > 192.168.3.5.sip: Flags [.], ack 2260, win 253, length 0
14:59:41.823972 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 180 Ringing
14:59:52.073381 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 183 Session Progress
15:00:02.027890 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 200 OK
15:00:02.028681 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600@192.168.3.5:5060;transport=udp SIP/2.0
15:00:02.034884 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: BYE sip:423@192.168.3.1:5060 SIP/2.0
15:00:02.035367 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: SIP/2.0 200 OK

my supervisor tel me that when i change the file to .wav  nrmally FS code and decode the file using unknown codec
can you help me  please sir ???



best regards 

amani