[Freeswitch-users] Audio Handshake Failure 1 in switch_rtp.c
ksaxmail at gmail.com
Fri Feb 28 10:06:50 MSK 2014
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
Originate: originate freetdm/1/a/99xxxxxxxx &bridge(freetdm/1/a/99xxxxxxxx)
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