Re: [Freeswitch-users] SIP TLS (and SRTP)

thanks allot, this was my mistake. /Tzury

It tells you to edit conf/directory/default.xml not dialplan/ default.xml and put <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})}" />

as the dial-string.

/b