basically the whole scene is such my pc running googletalk with gtalk2voip id added and configured to my freeswitch server --------> freeswitch server -------> gateway server basically i originate calls from my googletalk using the gtalk2voip service, now what happens is im able to make calls and talk also but there is a problem when the destinaion party is called and he rejects the call which is reported by freeswitch as user busy but this isnt passed onto the gtalk2voip server so me using googletalk i can hear the dummy ring music till it timesout. what actually is happening is on user busy, the external bridge is closed but the internal isnt as user busy int passed to gtalk2voip server but this isnt the case if i connect to freeswitch directly using a softphone or ata. when i connect directly using softphone then the log shows as below, its from start of call to call rejected by destination party to softphone being passed message of call rejected so both the ends of the bridge are cleared 2008-09-24 12:46:28 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing bip in->971559270058 in context default 2008-09-24 12:46:29 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/external/971559270058 [7df269f7-cbfc-7248-8688-299fc3d0b04b] 2008-09-24 12:46:37 [NOTICE] switch_channel.c:1426 switch_channel_perform_mark_p re_answered() Ring-Ready sofia/external/971559270058! 2008-09-24 12:46:37 [NOTICE] sofia.c:2226 sofia_handle_sip_i_state() Pre-Answer sofia/external/971559270058! 2008-09-24 12:46:37 [INFO] mod_sofia.c:1085 sofia_receive_message() Asked to sen d early media by sofia/internal/bipin@sip.xbipin.com 2008-09-24 12:46:37 [NOTICE] switch_channel.c:1426 switch_channel_perform_mark_p re_answered() Ring-Ready sofia/internal/bipin@sip.xbipin.com! 2008-09-24 12:46:37 [NOTICE] mod_sofia.c:1129 sofia_receive_message() Pre-Answer sofia/internal/bipin@sip.xbipin.com! 2008-09-24 12:46:55 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup sofi a/external/971559270058 [CS_EXCHANGE_MEDIA] [USER_BUSY] 2008-09-24 12:46:55 [NOTICE] switch_ivr_bridge.c:379 audio_bridge_thread() Hangu p sofia/internal/bipin@sip.xbipin.com [CS_EXECUTE] [USER_BUSY] 2008-09-24 12:46:55 [NOTICE] switch_core_session.c:812 switch_core_session_threa d() Session 20 (sofia/external/971559270058) Ended 2008-09-24 12:46:55 [NOTICE] switch_core_session.c:814 switch_core_session_threa d() Close Channel sofia/external/971559270058 [CS_HANGUP] 2008-09-24 12:46:55 [NOTICE] switch_core_session.c:812 switch_core_session_threa d() Session 19 (sofia/internal/bipin@sip.xbipin.com) Ended 2008-09-24 12:46:55 [NOTICE] switch_core_session.c:814 switch_core_session_threa d() Close Channel sofia/internal/bipin@sip.xbipin.com [CS_HANGUP] when calling from gtalk2voip id in googletalk, below is what happens 2008-09-24 12:44:22 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing 919 825967120->971559270058 in context default 2008-09-24 12:44:23 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/external/971559270058 [358bb7a0-fcda-af41-b62d-3d6b91fbe212] 2008-09-24 12:44:30 [NOTICE] switch_channel.c:1426 switch_channel_perform_mark_p re_answered() Ring-Ready sofia/external/971559270058! 2008-09-24 12:44:30 [NOTICE] sofia.c:2226 sofia_handle_sip_i_state() Pre-Answer sofia/external/971559270058! 2008-09-24 12:44:30 [INFO] mod_sofia.c:1085 sofia_receive_message() Asked to sen d early media by sofia/internal/bipin@sip.xbipin.com 2008-09-24 12:44:31 [NOTICE] switch_channel.c:1426 switch_channel_perform_mark_p re_answered() Ring-Ready sofia/internal/bipin@sip.xbipin.com! 2008-09-24 12:44:31 [NOTICE] mod_sofia.c:1129 sofia_receive_message() Pre-Answer sofia/internal/bipin@sip.xbipin.com! 2008-09-24 12:44:35 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup sofi a/external/971559270058 [CS_EXCHANGE_MEDIA] [USER_BUSY] THEN THE USER BUSY ISNT PASSED TO GTALK2VOIP SERVER SO FOR ME IN GOOGLETALK, THE CALL NEVER ENDS TILL IT TIMEOUTS AND THEN FREESWITCH REPORTS THE BELOW LOG 2008-09-24 12:45:02 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup sofi a/internal/bipin@sip.xbipin.com [CS_EXECUTE] [ORIGINATOR_CANCEL] and also is it possible to open the audio channel to originator as soon as rtp is received from destination either it being call processing or whatever so the originator can hear any provider messages etc that are passed by the destination? -- View this message in context: http://www.nabble.com/call-bridge-not-sending-hangup-or-call-rejected-to-pri... Sent from the Freeswitch-users mailing list archive at Nabble.com.