Re: [Freeswitch-users] SRTP call between two extensions.

10 Jan 2014

      Hi,

Can you pastebin a console log with siptrace enabled when placing a
call?

Regards,
António 

On Fri, 2014-01-10 at 19:50 +0200, Iskren Hadzhinedev wrote:
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Hello Daniel,
Here's the version according to freeswitch:
freeswitch@internal> version
FreeSWITCH Version 1.5.8b+git~20131210T191701Z~9ec1e86574~64bit (git
9ec1e86 2013-12-10 19:17:01Z 64bit).
I suppose I should build the 1.2 branch then?
Thanks,
--
Iskren Hadzhinedev
System Administrator
The Idea Factory | 20 Mearns Street | Aberdeen | AB11 5AT | UK
T: 01224 607500
VAT Reg No: 982 4936 74. Company registered in Scotland, SC237116
On Friday 10 January 2014 18:15:30 Daniel Ivanov wrote:
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which FS version is this? we've had this problem with 1.5 tree.
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On Fri, Jan 10, 2014 at 1:26 PM, Iskren Hadzhinedev <
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iskren.hadzhinedev@ikiji.com> wrote:
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Hello, everyone!
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I'm having trouble with SRTP calls between two internal
extensions.
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Whenever I configure the phones to require SRTP calls fail and the
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freeswitch console shows:
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[INFO] mod_dptools.c:3201 Originate Failed. Cause:
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INCOMPATIBLE_DESTINATION
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If I select "allow SRTP" so that encryption is not mandatory I get
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encryption on a single leg in a call.
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Here's an example of what's happening:
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Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)
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Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)
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Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail:
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INCOMPATIBLE_DESTINATION)
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I've setup the sip_secure_media=true variable as instructed from
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https://wiki.freeswitch.org/wiki/Secure_RTP
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but it's still not working. Any help is greatly appreciated!
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Kind regards,
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--
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Iskren Hadzhinedev
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_________________________________________________________________________
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Re: [Freeswitch-users] SRTP call between two extensions.

Antonio