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
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
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
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
bde2e2da51
FS-6889 #resolve
2014-10-03 11:34:42 -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
ae5d86515a
FS-6884 #comment these were mostly simple warnings
2014-10-02 19:20:35 -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
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
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
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
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
0150c862a2
FS-6854 #comment try this patch
2014-09-30 20:35:19 +05:00
a94fbe8079
mod_gsmopen: add tab completion for api commands
2014-09-29 13:25:30 +02:00
4ce990504e
Merge pull request #52 in FS/freeswitch from ~DDRAGIC/freeswitch:gsmopen_feature_additions to master
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* commit 'a9b2e061dcd1d95322d27e169ac2f0016aa628a3':
mod_gsmopen: clean up "gsm list" output a little
mod_gsmopen: convert reported RSSI from AT+CSQ to dBm.
mod_gsmopen: get device manufacturer, model and firmware version info.
mod_gsmopen: add support for reading own number from ON phonebook using AT+CNUM
mod_gsmopen: add AT+COPS support to get operator name.
2014-09-26 10:17:14 -05:00
9e3a375c36
Merge pull request #54 in FS/freeswitch from ~DDRAGIC/freeswitch:bugfix/FS-6820-mod_gsmopen-executing-gsm-reload to master
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* commit '9423953e028f8dd319a790ba1e5fdca37ff0cb2f':
FS-6820 mod_gsmopen: fix total interfaces count when executing gsm reload
2014-09-26 10:14:46 -05:00
0d538cd7b1
Merge pull request #42 in FS/freeswitch from ~DDRAGIC/freeswitch:FS-6799_fix_msg_index_check to master
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* commit '9cf72b541e8184b2911b0bd78f9aee71cd6d44b4':
FS-6799 fix reading sms in index 0
2014-09-26 10:13:44 -05:00
7c89c21153
FS-6860 #resolve this was fixed once but was lost in the last sync
2014-09-26 09:00:09 -05:00
f5b9bef319
Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch
2014-09-25 15:10:51 -05:00
0767191769
FS-6803 try this, less is more
2014-09-25 15:10:11 -05:00
f7de058acd
FS-6854 #resolve
2014-09-25 21:44:02 +05:00
7d7223e931
FS-6842 #resolve mod_graylog2: added send-uncompressed-header param- set to true for logstash support
2014-09-23 16:40:46 -04:00
9e72c8477f
fix possible buffer overrun in websocket uri and sync the ws.c between sofia and verto (missing code from last commit)
2014-09-24 01:09:44 +05:00
0cc7bc8db6
Add missing CURLOPT_NOSIGNAL options
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To work correctly in a multi-threaded environment, curl needs to be
used with CURLOPT_NOSIGNAL set to 1. If it's left at zero, the
default, then curl will use signals to deal with timeouts which will
often result in a crash.
ref: http://curl.haxx.se/libcurl/c/libcurl-tutorial.html#Multi-threading
ref: http://curl.haxx.se/libcurl/c/CURLOPT_NOSIGNAL.html
ref: http://stackoverflow.com/questions/9191668/error-longjmp-causes-uninitialized-stack-frame
ref: https://bugzilla.redhat.com/show_bug.cgi?id=539809
ref: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=570436
2014-09-23 00:04:21 +00:00
1bb0b8e16d
fix leak in lua when script does not execute properly in xml_binding handler
2014-09-23 03:57:04 +05:00
a9b2e061dc
mod_gsmopen: clean up "gsm list" output a little
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Replace tabs with spaces and add two columns, operator and imei.
2014-09-21 20:14:13 +02:00
4aa7c98d5a
mod_gsmopen: convert reported RSSI from AT+CSQ to dBm.
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Add to gsmopen_dump and events.
2014-09-21 20:14:12 +02:00
13a595a15e
mod_gsmopen: get device manufacturer, model and firmware version info.
2014-09-21 20:14:05 +02:00
79d962f38e
mod_gsmopen: add support for reading own number from ON phonebook using AT+CNUM
2014-09-21 20:04:04 +02:00
1f5bb3470d
mod_skinny: avoid truncation of non-null-terminated strings in protocol
2014-09-17 11:13:15 -05:00
295fcce8a8
add buffer_seconds param to shout filehandles to override the original default of 1 and remove previous code to attempt to buffer several seconds of audio in the open routine. Any experiencing jittery playback from slow shout destinations should add {buffer_seconds=N} to the file path to increase the amount of time allotted for buffering when no audio is discovered on the wire
2014-09-17 04:54:38 +05:00
b2917e06db
improve ssl errors
2014-09-17 02:14:43 +05:00
36addd5b61
bytes is signed
2014-09-16 19:15:12 +08:00
f78007766b
don't reset when video floor is locked
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when video floor is locked by a member, changing audio floor on del_member
will cause the video floor lock cleared unexpectedly, this commit fixes that.
2014-09-16 19:15:12 +08:00
04269fdf19
mod_skinny: additional logging
2014-09-15 16:42:31 -05:00
f262dbce94
FS-6821 mod_gsmopen: fix interface name in log
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Fix interface name for logs emitted from mod_gsmopen.cpp during startup
2014-09-14 13:06:31 +02:00
9423953e02
FS-6820 mod_gsmopen: fix total interfaces count when executing gsm reload
2014-09-14 12:24:19 +02:00
efe0ebd318
FS-6818 #resolve
2014-09-12 18:49:58 +05:00
Anthony Minessale