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[bipin at xbipin.com]

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-priginating-party-when-destinaion-cancels-call-tp19644777p19644777.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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