2051a86df2
FS-6889 #resolve
2014-10-07 13:47:44 -05:00
2514de94d2
fix obvious seg in setting a record file name to every participant and not checking for the recording member which does not have a session
2014-10-07 12:48:58 -05:00
a4f840b947
more jb improvements
2014-10-07 12:48:58 -05:00
6860b41763
Merge pull request #83 in FS/freeswitch from ~MKVONARX/freeswitch-fs-6710:FS-6710 to master
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* commit '490efb7177ddcd3e61018f02c1435362937e8b15':
FS-6710: fix incorrect comparison for Min-SE values between SIP INVITE and local configuration
2014-10-07 11:50:19 -05:00
9fe0956d99
Merge pull request #84 in FS/freeswitch from ~MKVONARX/freeswitch-fs-6897:FS-6897 to master
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* commit 'eaaf9468df366429c56366618df9e9be8457ea52':
FS-6897: uuid_send_info enhancement that allows setting the Content-Type of the SIP INFO message
2014-10-07 11:49:02 -05:00
d4929443f9
Merge pull request #59 in FS/freeswitch from ~SJTHOMASON/freeswitch:FS-5868 to master
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* commit '747322dcc6f4db1bffc985c9bcff0bd32a2682a9':
Remove Contact header from BYE and CANCEL requests.
2014-10-07 11:47:40 -05:00
7802232f16
Merge pull request #65 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/OPENZAP-220-ftmod_libpri-don-t-close-channel to master
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* commit '7ec7c920d1ef37a6b9753db0321dd19b2f3332a9':
OPENZAP-220 fix blocked into read and add cause for a correct hangup
2014-10-07 11:39:15 -05:00
4a5e36d63e
switch_pgsql.c switch_pgsql_next_result_timed() was using switch_time_now() for start time and switch_micro_time_now() for current time. These are different time sources that may not be in sync and could cause the query to timeout prematurely.
2014-10-07 09:33:19 -04:00
7ec7c920d1
OPENZAP-220 fix blocked into read and add cause for a correct hangup
2014-10-07 14:34:39 +02:00
eaaf9468df
FS-6897: uuid_send_info enhancement that allows setting the Content-Type of the SIP INFO message
2014-10-07 10:59:37 +02:00
490efb7177
FS-6710: fix incorrect comparison for Min-SE values between SIP INVITE and local configuration
2014-10-07 10:41:36 +02:00
da43bdeb12
add some calculations to jitter buffer related to judging the optimal size
2014-10-06 14:08:40 -05:00
397ec5ae1d
fix jb bug where once its full size it will never shrink due to logic err
2014-10-06 09:50:13 -05:00
f7210b2402
some more changes relates to new bypass media controls
2014-10-03 18:43:23 -05:00
afd6875d6b
FS-6781: #resolve #comment lets change this to always do confirm to match the other place where we set this
2014-10-03 16:53:38 -04:00
b2ae5f4cc2
few bugs on recent new features
2014-10-03 15:36:23 -05:00
acd8d74316
cleanup conditions
2014-10-03 12:48:43 -04:00
bde2e2da51
FS-6889 #resolve
2014-10-03 11:34:42 -05:00
6bed5d09a1
change type of int
2014-10-03 10:15:02 -05:00
0d1f5d09b3
add way to globally disable system commands by setting global var disable_system_api_commands=true
2014-10-03 12:17:33 -04:00
01bf42225c
FS-6888 #resolve #comment fix regression from refactoring new feature
2014-10-03 10:17:41 -05:00
d52cb335db
fix trivial vs2010 build errors
2014-10-02 19:47:05 -05:00
ae5d86515a
FS-6884 #comment these were mostly simple warnings
2014-10-02 19:20:35 -05:00
8db31f976f
fix some recovery issues with dynamic payloads
2014-10-02 18:34:00 -05:00
d17f14efbd
make sure to pass along appropriate configure flags to sub-configure's when cross compiling
2014-10-02 19:25:50 -04:00
10a3fa55ef
%FEATURE add bypass_media_resume_on_hold and bypass_media_after_hold variables to be set to true to enable these functions on a per channel basis
2014-10-02 17:49:09 -05:00
43733a6166
FS-6886 #comment addition of ignoring unhold as well
2014-10-02 15:48:29 -05:00
747322dcc6
Remove Contact header from BYE and CANCEL requests.
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Per rfc3261 the Contact header is not applicable and MUST not appear in
the request.
FS-5868 #resolve
2014-10-02 12:24:46 -07:00
6bfc05b81e
FS-6887 #resolve #comment new bug flag always_auto_adjust (also implicitly sets accept_any_packets)
2014-10-02 11:55:53 -05:00
9e9175321a
FS-6886 #resolve
2014-10-02 11:30:13 -05:00
eeedb8683e
the other way works better revert 91ffe171b6 to use high quality on stereo calls
2014-10-02 10:41:59 -05:00
9486a645f8
bump
2014-10-01 18:35:04 -05:00
cc44659a7c
bump
2014-10-01 18:34:05 -05:00
91ffe171b6
use OPUS_APPLICATION_VOIP always to get FEC and filtering
2014-10-01 18:33:33 -05:00
8258180735
start jb at one frame since it now has better adaptation
2014-10-01 18:21:50 -05:00
35aeae0170
FS-6822 #comment The code in question appears to have been added by me ( 18f20e24). I think this patch is the correct solution.
2014-10-01 18:11:01 -05:00
b3d71917d2
FS-6870 #comment vs2010 and vs2012 would rather fix it this way
2014-10-01 17:53:51 -05:00
661269a46f
Revert "FS-6870 #vs2012 and vs2010 make download of openssl dependent"
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This reverts commit a39db86863
2014-10-01 17:49:21 -05:00
5e11744632
fix makefile syntax errors
2014-10-01 17:52:01 -04:00
789e1481ed
FS-6880 #resolve #comment I would think that in real life once the call agreed on a codec it would only offer the negotiated codecs but we can fix this to always filter for good measure. I am not sure what the ramifications are of filtering responses but I think this patch will do so as well.
2014-10-01 13:03:50 -05:00
8e408e9abe
FS-6865 #resolve add XMPP priority to dingaling
2014-10-01 10:40:57 -05:00
a39db86863
FS-6870 #vs2012 and vs2010 make download of openssl dependent
2014-09-30 21:30:48 -05:00
644b41f792
FS-6874 #resolve
2014-09-30 17:05:06 -05:00
24084adf77
%FEATURE Add new feature to filter the SDP on bypass_media calls to remove or limit codecs.
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VARIABLE: bypass_media_sdp_filter
Can be set globally or per leg on the inbound side of a bypass_media bridge.
VALID FILTERS:
remove(): Removes the specified codec if it exists in the SDP.
only(): Removes all codecs besides the one specified (providing that it exists in the sdp) (will not remove telephone-event))
EXAMPLE 1 (remove everything leaving only g729):
<action application="set" data="bypass_media_sdp_filter=only(g729)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 2 (remove everything leaving only g729 and also remove dtmf):
<action application="set" data="bypass_media_sdp_filter=only(g729)|remove(telephone-event)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 3 (remove alaw and speex):
<action application="set" data="bypass_media_sdp_filter=remove(pcma)|remove(speex)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
2014-10-01 01:28:10 +05:00
92a66fb1e7
improve adaptive jitter buffer ascending check
2014-09-30 22:54:46 +05:00
56edfc7062
Merge pull request #76 in FS/freeswitch from ~HRISTO/freeswitch:fix-ptime-on-reinvite-master to master
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* commit 'fbe857e6fafabbca6a64584c51316ccc5e6ba96e':
fix ptime from known broken endpoints on re-invite
2014-09-30 10:53:37 -05:00
0150c862a2
FS-6854 #comment try this patch
2014-09-30 20:35:19 +05:00
6ac26fcc3e
Update timezones
2014-09-30 09:58:42 -05:00
f3473fdb58
Merge pull request #71 in FS/freeswitch from ~RTRELEAVEN/freeswitch-fs-4762-1:fs-4762 to master
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* commit '139b03204550cd394877da882fec49b08eba08fa':
improve regular expression to parse Jerusalem timezone files
2014-09-30 09:55:56 -05:00
4590220b53
Merge pull request #74 in FS/freeswitch from ~DDRAGIC/freeswitch:gsmopen_completion to master
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* commit 'a94fbe807905be714c774f7479936387b31602b2':
mod_gsmopen: add tab completion for api commands
2014-09-30 09:41:28 -05:00
Anthony Minessale
Mike Jerris
Chris Rienzo
Matteo Brancaleoni
Markus von Arx
Jeff Lenk
Spencer Thomason
Brian West
Anthony Minessale II