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12233 Commits

Author SHA1 Message Date
Mike Jerris
9fe0956d99 Merge pull request #84 in FS/freeswitch from ~MKVONARX/freeswitch-fs-6897:FS-6897 to master 
* 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
Mike Jerris
d4929443f9 Merge pull request #59 in FS/freeswitch from ~SJTHOMASON/freeswitch:FS-5868 to master 
* commit '747322dcc6f4db1bffc985c9bcff0bd32a2682a9':
  Remove Contact header from BYE and CANCEL requests.
 2014-10-07 11:47:40 -05:00
Markus von Arx
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
Michael Jerris
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
Anthony Minessale
b2ae5f4cc2 few bugs on recent new features 2014-10-03 15:36:23 -05:00
Anthony Minessale
bde2e2da51 FS-6889 #resolve 2014-10-03 11:34:42 -05:00
Michael Jerris
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
Jeff Lenk
ae5d86515a FS-6884 #comment these were mostly simple warnings 2014-10-02 19:20:35 -05:00
Michael Jerris
d17f14efbd make sure to pass along appropriate configure flags to sub-configure's when cross compiling 2014-10-02 19:25:50 -04:00
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
Spencer Thomason
747322dcc6 Remove Contact header from BYE and CANCEL requests. 
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
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
0150c862a2 FS-6854 #comment try this patch 2014-09-30 20:35:19 +05:00
Dušan Dragić
a94fbe8079 mod_gsmopen: add tab completion for api commands 2014-09-29 13:25:30 +02:00
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
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
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
Nathan Neulinger
1f5bb3470d mod_skinny: avoid truncation of non-null-terminated strings in protocol 2014-09-17 11:13:15 -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
b2917e06db improve ssl errors 2014-09-17 02:14:43 +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
Nathan Neulinger
04269fdf19 mod_skinny: additional logging 2014-09-15 16:42:31 -05:00
Dušan Dragić
f262dbce94 FS-6821 mod_gsmopen: fix interface name in log 
Fix interface name for logs emitted from mod_gsmopen.cpp during startup
 2014-09-14 13:06:31 +02:00
Dušan Dragić
9423953e02 FS-6820 mod_gsmopen: fix total interfaces count when executing gsm reload 2014-09-14 12:24:19 +02:00
Anthony Minessale
efe0ebd318 FS-6818 #resolve 2014-09-12 18:49:58 +05:00
Dušan Dragić
d5f9de4fa3 mod_gsmopen: add AT+COPS support to get operator name. 
For now expose the info in gsmopen_dump and events.
 2014-09-11 22:33:28 +02:00
Travis Cross
5bd35471f7 Add var to suppress Privacy: none header 
Apparently the MetaSwitch guys incorrectly interpret `Privacy: none`
as `Privacy: id`.

ref: RFC 3325

Reported-by: Stéphane Alnet <stephane@shimaore.net>

FS-6817 #resolve
 2014-09-11 19:56:19 +00:00
Anthony Minessale
7144b25254 obey sip_copy_custom_headers on bye 2014-09-12 00:37:19 +05:00