I removed the transfer and followed Volodymyr advise and its working now.

Thanks


On Tue, Aug 25, 2015 at 9:03 AM, Michael Jerris <mike@jerris.com> wrote:
Looks like maybe https://freeswitch.org/jira/browse/FS-8037

On Aug 25, 2015, at 2:18 AM, Volodymyr Fedorov <lexxua@gmail.com> wrote:

Hello, maybe transfer makes something bad with your call. Try this before GATEWAY in dialplan:
<action application="set" data="proxy_media=false"/>
<action application="set" data="bypass_media=false"/>
<action application="export" data="zrtp_secure_media=false"/>
<action application="export" data="nolocal:rtp_secure_media=true"/>


On Tue, Aug 25, 2015 at 5:09 AM, Gonzalo Gasca Meza <gascagonzalo@gmail.com> wrote:
I have the following scenario:

SIP Phone (RTP/SIP UDP) ---> Freeswitch ---> SRTP/TLS --> ITSP

Currently when I place a call Freeswitch is not sending SRTP SDES in SDP.
It does the TLS connection but SDP contains only AVP codecs.
I have used the rtp_secure_media set to true and also export but crypto caps are not attached. I can see them listed, but nothing happens.
Logs:


sofia_status
external::itsp-tls gateway   sip:gonzalo@tls.itsp.com:5061:5061 NOREG

dialplan/public.xml 
    <extension name="public_itsp_tls">
      <condition field="destination_number" expression="^(80\d{11})$">
          <action application="set" data="rtp_secure_media=true"/>
          <action application="export" data="rtp_secure_media=true"/>
          <action application="log" data="INFO Processing call to itsp"/>
          <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

dialplan/default.xml

<extension name="US_ITSP_Secure">
        <action application="log" data="INFO Processing REGEX"/>
        <condition field="destination_number" expression="^80(\d{11})$">
                <action application="set" data="dialed_number=+$1"/>
                <action application="log" data="INFO Processing US call to ITSP Secure ${dialed_number}"/>
                <action application="bridge" data="sofia/gateway/itsp-tls/${dialed_number}"/>
         </condition>
    </extension>

sip_profiles/external/sipsitspTrunk.xml 

<include>
<gateway name="itsp-tls">
<param name="tls-version" value="tlsv1"/>
<param name="register-transport" value="tls"/>
  <param name="contact-params" value="transport=tls"/>
<param name="realm" value="itsp-tls"/>
<param name="proxy" value="tls.itsp.com:5061"/> <!-- (your configured Trunk SIP URI) -->
<param name="register" value="false"/>
</gateway>
</include>




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--
Best regards,
Volodymyr
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org