did you set it first in your dialplan before anything else and make sure your dialplan does nothing at all that reqires media. Ideally nothing but setting the var and the bridge line. <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/1004"/> if you do anything at all that requires media from FS it will about the bypass and establish media from FS. It works, once you tell yourself that and look for how you have misconfigured it, then you will have success. On Tue, Apr 24, 2012 at 10:55 AM, Cucu Florian <koralu@gmail.com> wrote:
Thank you for the replies but didn't work.
I uncomment late-negotioation parameter in sip_profile/internal.xml <param name="inbound-late-negotiation" value="true"/>
I put <action application="set" data="bypass_media=true"/> in my dial plan.
I set for both Xlite only iLBC codec. The logs are:
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [iLBC:98:8000:30:13330]/[G7221:115:32000:20:48000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [iLBC:98:8000:30:13330]/[G7221:107:16000:20:32000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [iLBC:98:8000:30:13330]/[G722:9:8000:20:64000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [iLBC:98:8000:30:13330]/[PCMU:0:8000:20:64000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [iLBC:98:8000:30:13330]/[PCMA:8:8000:20:64000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [iLBC:98:8000:30:13330]/[GSM:3:8000:20:13200] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [telephone-event:101:8000:30:0]/[G7221:115:32000:20:48000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [telephone-event:101:8000:30:0]/[G7221:107:16000:20:32000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [telephone-event:101:8000:30:0]/[G722:9:8000:20:64000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec Compare [telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200] 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4948 Set 2833 dtmf send/recv payload to 101 2012-04-24 18:48:58.207474 [DEBUG] switch_channel.c:2848 (sofia/internal/1001@floflo85.zapto.org) Callstate Change DOWN -> HANGUP 2012-04-24 18:48:58.207474 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1001@floflo85.zapto.org) Running State Change CS_HANGUP 2012-04-24 18:48:58.207474 [NOTICE] sofia.c:5805 Hangup sofia/internal/1001@floflo85.zapto.org [CS_NEW] [INCOMPATIBLE_DESTINATION]
When I set for just one Xlite iLBC and also G711 the call is ok.
Ty.
On 4/24/12, Anthony Minessale <anthony.minessale@gmail.com> wrote:
To use bypass from the bypassl_media=true variable you must enable inbound-late-negotiation in your sofia profile.
On Tue, Apr 24, 2012 at 8:55 AM, Ken Rice <krice@freeswitch.org> wrote:
You still need to have the codecs enabled in your FreeSWITCH configuration, also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a list of other codecs that are just not enabled by default...
K
On 4/24/12 5:19 AM, "Cucu Florian" <koralu@gmail.com> wrote:
Hello,
I tried to test the behavior of "bypass media" in FS with the following scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
On each Xlite I set one single codec (that is currently unsupported by FS) as it said in wiki page: "This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSWITCH "
As the FS tutorial said "Before executing the bridge action you must set the "bypass_media" flag to true. " so I put the following line in my dial plan.
<extension name="extension-intercom"> <condition field="destination_number" expression="1001"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/1001"/> </condition> </extension>
<extension name="extension-intercom"> <condition field="destination_number" expression="1000"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/1000"/> </condition> </extension>
When I put in Xlite also G711ulaw/alaw as codecs preference the calls was ok. But when I put only codecs that FS do not support(ex. iLBC,BroadVoice32) the call failed. What am I missing?
I made this scenario because I need to route via FS some calls with G729r8 codec and I need no codec negotiation.
Ty
________________________________ _________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900