[Freeswitch-users] SRTP
Don WItt
witt at cylogistics.com
Mon Mar 7 07:45:09 MSK 2011
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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