What does it mean by LOSE_RACE
Hi Guys, I have got this error lately, while trying to bridge 2 calls. ==== 2008-08-20 16:27:48 [NOTICE] switch_ivr_originate.c:1311 switch_ivr_originate() Hangup sofia/external/01160163472932 [CS_CONSUME_MEDIA] [LOSE_RACE] 2008-08-20 16:27:48 [DEBUG] switch_channel.c:1362 switch_channel_perform_hangup() Kill sofia/external/01160163472932 [KILL] 2008-08-20 16:27:48 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/01160163472932 [BREAK] 2008-08-20 16:27:48 [DEBUG] sofia_glue.c:1756 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/1001@192.168.253.101] 192.168.253.101 port 24120 -> 192.168.253.101 port 8880 codec: 8 ms: 20 2008-08-20 16:27:48 [DEBUG] switch_rtp.c:813 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/01160163472932) State CONSUME_MEDIA going to sleep 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/01160163472932 Running State Change CS_HANGUP 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:393 switch_core_session_run() (sofia/external/01160163472932) State HANGUP 2008-08-20 16:27:48 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/external/01160163472932 hanging up, cause: LOSE_RACE 2008-08-20 16:27:48 [DEBUG] mod_sofia.c:328 sofia_on_hangup() Sending CANCEL to sofia/external/01160163472932 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/external/01160163472932, cause: LOSE_RACE ==== One phone rang not the other, as we can see the cause from the error log. Below is the way I am bridging calls. ==== <extension name="con-1019" continue="true"> <condition field="destination_number" expression="^1019$"> <action application="set" data="call_timeout=20"/> <action application="bridge" data="sofia/gateway/gizmo1/011xxxxxxxx,sofia/gateway/gizmo2/011yyyyyyyy" /> </condition> </extension> ==== Thanks for your support. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari
My intention is, say using 1001 --- voip user1 -------> hardphone1 using 1001 --- voip user2 -------> hardphone2 and then bridge both, so they can talk to each other. Am I making sense here? I mean, can we bridge one or more session using bridge? Or do we need to do it via conference? Thanks. On Wed, Aug 20, 2008 at 4:37 PM, Adeel Ansari <adeel.gnome@gmail.com> wrote:
Hi Guys,
I have got this error lately, while trying to bridge 2 calls.
==== 2008-08-20 16:27:48 [NOTICE] switch_ivr_originate.c:1311 switch_ivr_originate() Hangup sofia/external/01160163472932 [CS_CONSUME_MEDIA] [LOSE_RACE] 2008-08-20 16:27:48 [DEBUG] switch_channel.c:1362 switch_channel_perform_hangup() Kill sofia/external/01160163472932 [KILL] 2008-08-20 16:27:48 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/01160163472932 [BREAK] 2008-08-20 16:27:48 [DEBUG] sofia_glue.c:1756 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/1001@192.168.253.101] 192.168.253.101 port 24120 -> 192.168.253.101 port 8880 codec: 8 ms: 20 2008-08-20 16:27:48 [DEBUG] switch_rtp.c:813 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/01160163472932) State CONSUME_MEDIA going to sleep 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/01160163472932 Running State Change CS_HANGUP 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:393 switch_core_session_run() (sofia/external/01160163472932) State HANGUP 2008-08-20 16:27:48 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/external/01160163472932 hanging up, cause: LOSE_RACE 2008-08-20 16:27:48 [DEBUG] mod_sofia.c:328 sofia_on_hangup() Sending CANCEL to sofia/external/01160163472932 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/external/01160163472932, cause: LOSE_RACE ====
One phone rang not the other, as we can see the cause from the error log. Below is the way I am bridging calls.
==== <extension name="con-1019" continue="true"> <condition field="destination_number" expression="^1019$"> <action application="set" data="call_timeout=20"/> <action application="bridge" data="sofia/gateway/gizmo1/011xxxxxxxx,sofia/gateway/gizmo2/011yyyyyyyy" /> </condition> </extension> ====
Thanks for your support.
-- Best, Adeel Ansari
-- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari
This line mans you're calling the two numbers at once. The LOSE_RACE means one made it to someone before the other... thus one leg was hung up. /b On Aug 20, 2008, at 3:37 AM, Adeel Ansari wrote:
<action application="bridge" data="sofia/gateway/ gizmo1/011xxxxxxxx,sofia/gateway/gizmo2/011yyyyyyyy" />
Brian West sip:brian@freeswitch.org
you probably want to prepend {ignore_early_media=true} to the dial string to make it wait until someone actually answers. On Wed, Aug 20, 2008 at 9:11 AM, Brian West <brian@freeswitch.org> wrote:
This line mans you're calling the two numbers at once. The LOSE_RACE means one made it to someone before the other... thus one leg was hung up.
/b
On Aug 20, 2008, at 3:37 AM, Adeel Ansari wrote:
<action application="bridge" data="sofia/gateway/ gizmo1/011xxxxxxxx,sofia/gateway/gizmo2/011yyyyyyyy" />
Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Actually, my intention is to act as an agent between the two numbers I have mentioned in the bridge clause. And yes, I want to ring both at the same time, and let both of them answer the call, so they can talk to each other. Should I need to follow some other way i.e. conference or something? Bridging like this will not do that? Thanks again. On Wed, Aug 20, 2008 at 10:57 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
you probably want to prepend {ignore_early_media=true} to the dial string to make it wait until someone actually answers.
On Wed, Aug 20, 2008 at 9:11 AM, Brian West <brian@freeswitch.org> wrote:
This line mans you're calling the two numbers at once. The LOSE_RACE means one made it to someone before the other... thus one leg was hung up.
/b
On Aug 20, 2008, at 3:37 AM, Adeel Ansari wrote:
<action application="bridge" data="sofia/gateway/ gizmo1/011xxxxxxxx,sofia/gateway/gizmo2/011yyyyyyyy" />
Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari
participants (3)
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Adeel Ansari
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Anthony Minessale
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Brian West