[Freeswitch-users] SRTP

[mitch Johnson]

No.

From: "Don WItt" <witt at cylogistics.com>
To: "'FreeSWITCH Users Help'" <freeswitch-users at 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 at cylogistics.com
http://www.cylogistics.com
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-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Johnson
Sent: Sunday, March 06, 2011 6:50 PM
To: freeswitch-users at 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
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