- 04 1月, 2017 9 次提交
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由 Anthony Minessale 提交于
FS-9906: [mod_conference] Member join/part in conference shows webcam briefly during slide transition #resolve
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由 Brian West 提交于
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由 Mike Jerris 提交于
Merge pull request #1137 in FS/freeswitch from ~KIDPLUG/freeswitch-kp:bugfix/FS-9917-switch_rtp-loses-the-current-packet to master * commit '879c04b3': FS-9917 [switch_rtp/core] Fix in do_flush to handle the current packet (if RFC2833) rather than discard it. only print flushed if we flushed something
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由 chapman 提交于
FS-9917 [switch_rtp/core] Fix in do_flush to handle the current packet (if RFC2833) rather than discard it. only print flushed if we flushed something
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由 Mike Jerris 提交于
Merge pull request #1124 in FS/freeswitch from ~TCULJAGA/freeswitch_tc:bugfix/FS-9873-a1-hash_for_mod_verto to master * commit '1f7a7e33': FS-9873 mod_verto a1-hash - squashed
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由 Anthony Minessale 提交于
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由 Brian West 提交于
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由 Tihomir Culjaga 提交于
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由 Anthony Minessale 提交于
FS-9206: [mod_sofia] proxy media with enable-3pcc=proxy does not properly pass audio after 3pcc re-invite #resolve
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- 03 1月, 2017 15 次提交
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由 Mike Jerris 提交于
Merge pull request #1128 in FS/freeswitch from ~ANDYWOLK/freeswitch:feature/FS-9896-mod_v8-implement-serving-configuration to master * commit 'a3a8d7e6': FS-9896 [mod_v8] Implement Serving Configuration with JavaScript
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由 Mike Jerris 提交于
* commit '7248a4f3': FS-9910 [mod_sofia]: Set SIP reason header for BYE events.
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Josh Allmann 提交于
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由 Mike Jerris 提交于
Merge pull request #1118 in FS/freeswitch from ~ANTONIO/freeswitch:bugfix/FS-9877-mod_loopback-no-audio to master * commit '5dfc63c1': FS-9877 [mod_loopback] fix no audio
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由 Mike Jerris 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Mike Jerris 提交于
Merge pull request #1129 in FS/freeswitch from ~ANDYWOLK/freeswitch:bugfix/FS-9897-mod_v8-fix-visual-studio-2015-build to master * commit '31fd8f36': FS-9897 [mod_v8] Fix Visual Studio 2015 build
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由 Mike Jerris 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale II 提交于
* commit 'c1abfaf4': FS-9903 #resolve #comment fix server side TLS and add client TLS support
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由 Seven Du 提交于
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- 02 1月, 2017 1 次提交
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由 Seven Du 提交于
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- 31 12月, 2016 1 次提交
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由 Seven Du 提交于
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- 30 12月, 2016 1 次提交
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由 Anthony Minessale 提交于
FS-9206: [mod_sofia] proxy media with enable-3pcc=proxy does not properly pass audio after 3pcc re-invite #resolve
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- 29 12月, 2016 4 次提交
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由 Andrey Volk 提交于
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由 Italo Rossi 提交于
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由 Italo Rossi 提交于
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由 Andrey Volk 提交于
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- 28 12月, 2016 5 次提交
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由 Brian West 提交于
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由 Mike Jerris 提交于
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由 Italo Rossi 提交于
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由 Italo Rossi 提交于
[mod_callcenter] FS-9347: Do not log as error when the member is gone just before we assigned an agent, now logging as DEBUG
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由 Italo Rossi 提交于
[mod_callcenter] FS-9689 Fixed issue when agent rejects the call while hearing cc_outbound_announce making member (caller) waiting on queue forever
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- 27 12月, 2016 4 次提交
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由 Italo Rossi 提交于
[mod_callcenter] FS-9723: Fixing cc_warning_tone, using switch_ivr_play_file instead of queue private event
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由 Italo Rossi 提交于
Merge pull request #1126 in FS/freeswitch from ~STEFAN_YOHANSSON/freeswitch:feature/change-wss-server to master * commit '10646ab9': [verto-communicator] - Added change server feature
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由 Mike Jerris 提交于
Merge pull request #1125 in FS/freeswitch from ~ANDYWOLK/freeswitch:feature/FS-7933-v8-allow-non-blocking-socket-access to master * commit '3f3cd734': FS-7933 [mod_v8] Implement timeout property for Socket() class in javascript.
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由 Mike Jerris 提交于
* commit '8c1ed38d': FS-9792: Set channel variable based on the sip phone Accept Language SIP message
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