No. From: "Don WItt" <witt@cylogistics.com> To: "'FreeSWITCH Users Help'" <freeswitch-users@lists.freeswitch.org> Date: Sun, 6 Mar 2011 20:45:09 -0800 Subject: Re: [Freeswitch-users] SRTP Is your dad's name Jerry? Don Witt Cylogistics 809B Cuesta Dr. #2149 Mountain View, CA 94040 650-694-4949 X 9102 Fax: 650-694-4953 witt@cylogistics.com http://www.cylogistics.com --------------------------------------------------- Suppliers of award winning Ring Carrier -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mitch Johnson Sent: Sunday, March 06, 2011 6:50 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] SRTP 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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org