[Freeswitch-users] codec changed

[Michael Jerris]

https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation <https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation>

> On Mar 31, 2016, at 10:31 AM, amani mansour <amani.mansour2 at gmail.com> wrote:
> 
> 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 at 192.168.3.5 <mailto:sip%3A600 at 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 at 192.168.3.5 <mailto:sip%3A600 at 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 at 192.168.3.5 <mailto:sip%3A600 at 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 at 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 at 192.168.3.1:5060 <http://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 
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