[Freeswitch-users] Audio Handshake Failure 1 in switch_rtp.c

[Anthony Minessale]

Its related to dtls make sure you are on the latest version which requires
openssl not supported by that older ubuntu so you need to build your own.
 Use wireshark captures filtered on dtls to see the root of your problem.




On Fri, Feb 28, 2014 at 1:06 AM, Kshitij Saxena <ksaxmail at gmail.com> wrote:

> Summary: SIPml / JSSIP calls over WS to FreeSWITCH are terminated with
> "Audio Handshake Failure 1" in switch_rtp.c
>
> I am running FreeSWITCH Version 1.5.8b on an Ubuntu 12.04 server with a
> Sangoma A101DE PRI card running on Wanpipe 7.0.10. The PRI card is
> connected to the ISP's network.
>
> When I try connecting to FreeSWITCH from the SIPml or JSSIP demo pages,
> (over websocket, RTCWebBreaker enabled, port 5066 open on server for
> tcp/udp along with 5060-5081 and 16383-32768), the client is successfully
> registered, but any attempt to make a call is failing. In fs_cli I can see
> an error "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is
> attempted from the client.
>
> When using X-lite, the calls are successful. Originate from within fs_cli
> is also successful.
>
> I have been splitting my hair for 2 days but cannot begin to decipher what
> has gone wrong! What is worse is that I have used the exact same
> configuration successfully at 2 previous locations!
>
> Any help would be very greatly appreciated.
>
> Full Version: FreeSWITCH Version
> 1.5.8b+git~20140208T085053Z~4fa68fcd75~64bit (git 4fa68fc 2014-02-08
> 08:50:53Z 64bit)
> Originate: originate freetdm/1/a/99xxxxxxxx &bridge(freetdm/1/a/99xxxxxxxx)
>
> Regards,
> Kshitij Saxena
>
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-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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