[Freeswitch-users] Call from Secure RTP to non-secure RTP

[Mark Campbell-Smith]

Hi!

How do I setup FS so that placing a call from an extension that only
support SRTP (1002) to an extension that only supports RTP (1000)?

I put this dialstring, from the wiki
http://wiki.freeswitch.org/wiki/Tls, into the users xml file under
directory/default

 <param name="dial-string"
value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@${dialed_domain})}|transport=tls)},
  presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"
/>

I have also put a <action application="export"
data="sip_secure_media=true"/> when 1000 is dialing 1002.
      <condition field="destination_number" expression="^(1002)$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="export" data="sip_secure_media=true"/>
        <action application="bridge" data="user/${dialed_extension}@${domain}"/>

However I never see crytpo sent in the RTP to 1002 and it responds
with Bad Security Level

What have I missed?

Thanks