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16529 Commits

Author SHA1 Message Date
Anthony Minessale
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
Anthony Minessale
43733a6166 FS-6886 #comment addition of ignoring unhold as well 2014-10-02 15:48:29 -05:00
Anthony Minessale
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
Anthony Minessale
9e9175321a FS-6886 #resolve 2014-10-02 11:30:13 -05:00
Anthony Minessale
eeedb8683e the other way works better revert 91ffe171b6e76f60f1e94f148176ce8556d460e6 to use high quality on stereo calls 2014-10-02 10:41:59 -05:00
Anthony Minessale
91ffe171b6 use OPUS_APPLICATION_VOIP always to get FEC and filtering 2014-10-01 18:33:33 -05:00
Anthony Minessale
8258180735 start jb at one frame since it now has better adaptation 2014-10-01 18:21:50 -05:00
Michael Jerris
5e11744632 fix makefile syntax errors 2014-10-01 17:52:01 -04:00
Anthony Minessale
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
Brian West
8e408e9abe FS-6865 #resolve add XMPP priority to dingaling 2014-10-01 10:40:57 -05:00
Brian West
644b41f792 FS-6874 #resolve 2014-09-30 17:05:06 -05:00
Anthony Minessale
24084adf77 %FEATURE Add new feature to filter the SDP on bypass_media calls to remove or limit codecs. 
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
Anthony Minessale
92a66fb1e7 improve adaptive jitter buffer ascending check 2014-09-30 22:54:46 +05:00
Anthony Minessale II
56edfc7062 Merge pull request #76 in FS/freeswitch from ~HRISTO/freeswitch:fix-ptime-on-reinvite-master to master 
* commit 'fbe857e6fafabbca6a64584c51316ccc5e6ba96e':
  fix ptime from known broken endpoints on re-invite
 2014-09-30 10:53:37 -05:00
Anthony Minessale
0150c862a2 FS-6854 #comment try this patch 2014-09-30 20:35:19 +05:00
Mike Jerris
4590220b53 Merge pull request #74 in FS/freeswitch from ~DDRAGIC/freeswitch:gsmopen_completion to master 
* commit 'a94fbe807905be714c774f7479936387b31602b2':
  mod_gsmopen: add tab completion for api commands
 2014-09-30 09:41:28 -05:00
Hristo Trendev
fbe857e6fa fix ptime from known broken endpoints on re-invite 
Freeswitch tries to fix timing issues (wrong ptime) on re-invite the same way
it does for the initial invite. This results in small audio glitches, while it
sends a couple of packets with different ptime, before the timing detection
logic figures out the remote (broken) endpoint true ptime.

In order to avoid unnecessary timing changes, this patch overwrites the
advertised ptime from known broken endpoints with the ptime, which was detected
by freeswitch. It does this by checking if the sip_h_X-Broken-PTIME (1.2.x) or
rtp_h_X-Broken-PTIME (master) variables are set.

FS-6644 #resolve
 2014-09-30 11:19:35 +02:00
Anthony Minessale
da51603a2c FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 #resolve #comment 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d 2014-09-29 19:26:32 +05:00
Anthony Minessale
e94af49e1e revert 2014-09-29 19:26:01 +05:00
Anthony Minessale
d619017621 FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d 2014-09-29 19:21:01 +05:00
Dušan Dragić
a94fbe8079 mod_gsmopen: add tab completion for api commands 2014-09-29 13:25:30 +02:00
Michael Jerris
dac4afbfdb this was alraedy in there, whoops 2014-09-28 10:40:57 -04:00
Darren Schreiber
c1e9b0d414 expose apr socket put 2014-09-27 15:02:41 -07:00
Giovanni Maruzzelli
4ce990504e Merge pull request #52 in FS/freeswitch from ~DDRAGIC/freeswitch:gsmopen_feature_additions to master 
* 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
Giovanni Maruzzelli
9e3a375c36 Merge pull request #54 in FS/freeswitch from ~DDRAGIC/freeswitch:bugfix/FS-6820-mod_gsmopen-executing-gsm-reload to master 
* commit '9423953e028f8dd319a790ba1e5fdca37ff0cb2f':
  FS-6820 mod_gsmopen: fix total interfaces count when executing gsm reload
 2014-09-26 10:14:46 -05:00
Giovanni Maruzzelli
0d538cd7b1 Merge pull request #42 in FS/freeswitch from ~DDRAGIC/freeswitch:FS-6799_fix_msg_index_check to master 
* commit '9cf72b541e8184b2911b0bd78f9aee71cd6d44b4':
  FS-6799 fix reading sms in index 0
 2014-09-26 10:13:44 -05:00
Brian West
7c89c21153 FS-6860 #resolve this was fixed once but was lost in the last sync 2014-09-26 09:00:09 -05:00
Brian West
f5b9bef319 Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch 2014-09-25 15:10:51 -05:00
Brian West
0767191769 FS-6803 try this, less is more 2014-09-25 15:10:11 -05:00
Anthony Minessale
f7de058acd FS-6854 #resolve 2014-09-25 21:44:02 +05:00
Anthony Minessale
c018c28738 FS-6851 #resolve 2014-09-24 20:40:27 +05:00
Chris Rienzo
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
Anthony Minessale
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
Anthony Minessale
e8d6866899 use the more reliable offset_pos counter in file position parsing for seek in scripts 2014-09-23 21:01:25 +05:00
Travis Cross
0cc7bc8db6 Add missing CURLOPT_NOSIGNAL options 
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
Anthony Minessale
1bb0b8e16d fix leak in lua when script does not execute properly in xml_binding handler 2014-09-23 03:57:04 +05:00
Dušan Dragić
a9b2e061dc mod_gsmopen: clean up "gsm list" output a little 
Replace tabs with spaces and add two columns, operator and imei.
 2014-09-21 20:14:13 +02:00
Dušan Dragić
4aa7c98d5a mod_gsmopen: convert reported RSSI from AT+CSQ to dBm. 
Add to gsmopen_dump and events.
 2014-09-21 20:14:12 +02:00
Dušan Dragić
13a595a15e mod_gsmopen: get device manufacturer, model and firmware version info. 2014-09-21 20:14:05 +02:00
Dušan Dragić
79d962f38e mod_gsmopen: add support for reading own number from ON phonebook using AT+CNUM 2014-09-21 20:04:04 +02:00
Anthony Minessale
372455c30a FS-6829 #resolve 2014-09-19 02:28:47 +05:00
Jeff Lenk
8f85b5204c vs2010 trival compiler warnings 2014-09-17 18:11:20 -05:00
Nathan Neulinger
1f5bb3470d mod_skinny: avoid truncation of non-null-terminated strings in protocol 2014-09-17 11:13:15 -05:00
Anthony Minessale
d2f8fca18a FS-6825 #resolve #comment caused by regression in commit 0732c0b0 pertaining to FS-6825 2014-09-17 20:32:18 +05:00
Anthony Minessale
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
Anthony Minessale
16d947dd7a can't have asserts here after all 2014-09-17 02:14:54 +05:00
Anthony Minessale
b2917e06db improve ssl errors 2014-09-17 02:14:43 +05:00
Anthony Minessale
47ae1837d5 add some asserts 2014-09-16 20:44:10 +05:00
Seven Du
36addd5b61 bytes is signed 2014-09-16 19:15:12 +08:00
Seven Du
f78007766b don't reset when video floor is locked 
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