Commit Graph

637 Commits

Author SHA1 Message Date
Armen Babikyan 6ed86abf9f FS-9300: Add support for disabling sofia's 100 Trying via configuration, and sending 100 Trying from dialplan 2017-02-13 14:49:05 -08:00
Sergey Safarov df1ab07ca4 FS-9924: Removed extra space in source files 2017-02-09 23:59:49 -05:00
Anthony Minessale 9dba32410f FS-10015: [freeswitch-core] Add variable to allow firing of text events #resolve 2017-02-06 16:37:59 -06:00
Luis Azedo 7c60be14c7 [core] add channel hold/unhold verbosity 2016-11-16 19:30:50 -06:00
Mike Jerris 17444343c0 Merge pull request #914 in FS/freeswitch from feature/FS-9367-log-session-creation to master
* commit '0fd6667fd2e30122ce7facf74a2f26ee7dfb26fc':
  [FS-9367] log session counts at channel state change
2016-11-11 15:48:21 -06:00
Anthony Minessale 9b8a5edd3d FS-9638 2016-11-10 12:09:00 -06:00
nneul at mst.edu 0fd6667fd2 [FS-9367] log session counts at channel state change 2016-07-19 13:03:12 -05:00
Thomas Weber 346e044daf Buffer overflow in switch_channel_expand_variables_check and switch_event_expand_headers_check fixed (FS-8757) 2016-04-19 11:44:35 -05:00
Ken Rice e18c12b609 FS-8953 [core] white space clean up. 2016-03-17 08:55:00 -05:00
Anthony Minessale 2cf9962f61 FS-8914 2016-03-09 00:02:59 -06:00
Artur Zaprzała e24c393979 FS-8812 Respect in_thread_only parameter in switch_channel_check_signal() 2016-02-08 15:04:02 +01:00
Anthony Minessale c01a8849a0 FS-8720 #resolve [Segmentation Fault when switch_channel_str2cause is called] 2016-01-12 16:29:44 -06:00
Anthony Minessale 0da5d8a350 FS-8716 #resolve [recording offset is delayed few seconds for rtmp stream] 2016-01-07 16:39:06 -06:00
Anthony Minessale 56a68e3ad9 FS-8677 #resolve [Crash (possible memory corruption) after codec change] 2016-01-06 10:10:14 -06:00
Michael Jerris 9f52ebf257 FS-8658: [mod_verto] windows fixes for mod_verto 2015-12-14 16:37:09 -06:00
Anthony Minessale 85f8bca628 FS-8642 #resolve [CF_VIDEO_READY set on non-video calls] 2015-12-09 19:02:23 -06:00
Anthony Minessale 6a08424fa7 FS-8566 #resolve [Call fails when put on hold in bypass media mode with inbound late neg set to false] 2015-11-25 13:53:14 -06:00
Mark Lipscombe 6288af5ef1 FS-8413: Segfault calling session:getVariable(nil) in lua script
script calling session:getVariable() with a null variable
name will cause FreeSWITCH to segfault.

This change checks whether varname parameter to
switch_channel_get_variable_dup is non-NULL.
2015-11-04 17:33:11 +11:00
Corey Burke 3a9e7f08b4 FS-8286: Minor debug log level tweaks
Adjust some DEBUG and INFO log lines, reducing log verbosity at the INFO level while increasing call debugging info.
2015-10-02 08:41:41 -07:00
Anthony Minessale 43ef01fbbe correct version of proposed patch 2015-09-23 11:58:57 -05:00
Artur Zaprzała 8ea99bfccf FS-7344: Fix race condition for INVITE right after ACK in 3PCC mode. 2015-09-03 22:32:32 +02:00
Michael Jerris 5afdc24bf7 add a debug log 2015-06-24 17:43:38 -04:00
Anthony Minessale 8aea72c825 FS-7602 FS-7499 FS-7587 modify dtls init function placement 2015-06-04 20:37:15 -05:00
Anthony Minessale 2188358832 FS-7500 FS-7499 refactoring while battling chrome 2015-05-28 12:47:34 -05:00
Anthony Minessale dc2e98e536 FS-7500: start media thread one answer/pre_answer 2015-05-28 12:47:34 -05:00
Anthony Minessale 40484fce58 FS-7499 FS-7500 mods for interop against latest chrome builds 2015-05-28 12:47:34 -05:00
Anthony Minessale dc4c38dab5 FS-7499 FS-7508 FS-7501 some more general improvements for initial call setup 2015-05-28 12:47:29 -05:00
Anthony Minessale a8a2c32ac3 FS-7499 FS-7500: combat black screen disease 2015-05-28 12:47:28 -05:00
Anthony Minessale 6522dbdffb FS-7500: wait for video ready on answer 2015-05-28 12:47:17 -05:00
Brian West 379950f523 FS-7500: video introp tweaks 2015-05-28 12:47:15 -05:00
Anthony Minessale 84ca513353 FS-7500: fix some regressions regarding passthru video 2015-05-28 12:47:14 -05:00
Anthony Minessale fdcfcaece9 FS-7500: don't wait for video ready from inside video thread that sets that flag 2015-05-28 12:47:11 -05:00
Anthony Minessale 24254bb1fd FS-7500: revert 2015-05-28 12:47:11 -05:00
Anthony Minessale d3359ff9f0 FS-7500: don't wait for video ready from inside video thread that sets that flag 2015-05-28 12:47:11 -05:00
Anthony Minessale 9a7a33fb55 FS-7500: block in flag set for wait for video ready 2015-05-28 12:47:07 -05:00
Chris Rienzo 3c2afc6a2c FS-7406 #resolve #comment Added DTMF-Source header to DTMF event.
DTMF-Source may have the following values:
   APP : injected by application (send_dtmf, etc)
   ENDPOINT : detected by endpoint signaling (like SIP INFO)
   INBAND_AUDIO : detected by start_dtmf, spandsp_start_dtmf, etc
   RTP : detected as 2833/4733 telephone event
   UNKNOWN : unknown source

One possible use of this header is to determine telephone events
are being received, and if so, disable inband detection.
2015-04-01 11:00:43 -04:00
Anthony Minessale 2c92ef31e3 FS-7386
Conflicts:
	src/switch_core_media.c
2015-03-26 23:52:53 -05:00
Brian West 4b87056625 remove debugging printf 2014-11-03 14:17:24 -06:00
Anthony Minessale 8d720d5bcc FS-6940 #resolve #comment %FEATURE use the variable digits_dialed_filter to set regular expressions with () captures and anything matched will be replaced with X's in the CDR 2014-10-23 12:47:27 -05:00
Anthony Minessale 01bf42225c FS-6888 #resolve #comment fix regression from refactoring new feature 2014-10-03 10:17:41 -05:00
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
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 c018c28738 FS-6851 #resolve 2014-09-24 20:40:27 +05:00
William King 47760e2d75 Silence a warning in clang-3.5 dealing with implicit conversion from 64bit to 32bit in a function call to switch_ivr_sleep() 2014-08-02 19:41:44 -07:00
Anthony Minessale 388d980b86 FS-6701 #resolve 2014-07-29 22:20:31 +05:00
Brian West ac265ce495 FS-6682: fix arg order on switch_channel_export_variable called from switch_channel_export_variable_printf #resolve 2014-07-24 08:28:38 -05:00
Anthony Minessale 91d405a2c1 call recovery_track on recovering channels once the recovery has completed and fix race condition with repeated recovery 2014-07-16 21:40:23 +05:00
Kathleen King 2d85726ecd Fixed dead code.
While reviewing code I noticed some dead code. It was not possible to
default to the channel variable because the parameter could not be
both full and empty.

If the parameter is not a non zero length string then the code looked
like it was intending to default to the channel variable
'presence_data_cols'. If neither of these are the case it is a noop.

By enabling the dead code, you now have access to set the
'presence_data_cols' in the dialplan or scripts like lua.
2014-07-03 17:03:01 -07:00
Travis Cross 7406be6927 Relay cause of hangup on SRTP failure
We hangup the channel after receiving 10 SRTP packets in a row with a
bad auth tag or that are replayed.  Prior to this commit we were
indicating a normal clearing.  When doing interop and looking first at
packet traces, this made freeswitch's behavior look surprising.  With
this commit we'll indicate more loudly what's happening.
2014-06-28 01:18:50 +00:00
Anthony Minessale 495db48f5a make device state code more accurate 2014-05-31 00:30:59 +05:00