Thanks to all who have replied - the problem has been found (my fault...).
 
The variables set above (with set/export) work as intended. I simply had a clause in the dial string that sets SRTP only if TLS is active as well.
 
                    __Yehavi: 


2012/11/21 Brian West <brian@freeswitch.org>
you make sure sip_secure_media=true is set inside {} on the originate or export the variable.
--
Brian West
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On Nov 21, 2012, at 7:08 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:

> Hi,
>
>   I am trying to use SRTP without TLS. The initiating side asks and get SRTP (two way) while the other side is not offered SRTP at all.
>
> I've noted that ${sip_has_crypto} is null. I also tried setting unconditionaly sip_secure_rtp (both set and export), but it doesn't help.
>
> Any idea what I am doing wrong?
>
> BTW, I am using today's mornning GIT.
>
>                   Thanks, __Yehavi:


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