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
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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Thanks Anthony. I compiled OpenSSL v1.0.0l and re-compiled FreeSWITCH (--with-openssl) but I am still getting "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is answered. Am I missing something? Would switching to Ubuntu 13.10 solve this issue? Regards, Kshitij On Fri, Feb 28, 2014 at 10:42 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You need openssl 1.0.1. On Mon, Mar 3, 2014 at 9:58 PM, Kshitij Saxena <ksaxmail@gmail.com> wrote:
Thanks Anthony.
I compiled OpenSSL v1.0.0l and re-compiled FreeSWITCH (--with-openssl) but I am still getting "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is answered. Am I missing something? Would switching to Ubuntu 13.10 solve this issue?
Regards, Kshitij
On Fri, Feb 28, 2014 at 10:42 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
participants (3)
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Anthony Minessale
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Kristian Kielhofner
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Kshitij Saxena