[Freeswitch-users] SRTP

[Mitch Johnson]

When I dial 9664 to test the tls/srtp configuration it says that the call is secure, however, when I dial another phone configured for tls/srtp the call doesn't go through, the automated attendant comes online to say that the extension is not available and then puts the call to voicemail.

I read the Secure RTP wiki, I do see a similar entries in the dialplan under the extension name global, for both inbound and outbound:

<condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
        <action application="set" data="sip_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <action application="export" data="sip_secure_media=true"/>
      </condition>

So if I can make the test call to 9664 on both phones, which I assume is using the inbound part by connecting the call.  

Any help in figuring this out would be greatly appreciated.

BTW, I did buy the book, but there's no mention of SRTP/TLS in there.

Thanks,

Mitch