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
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.
...
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
...
* 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
...
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
...
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.
...
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
...
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
...
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
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
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
7144b25254
obey sip_copy_custom_headers on bye
2014-09-12 00:37:19 +05:00
622e0e1a6f
Check for null hash on increment of mod_hash limit
...
When we specifically release all limits on a channel we destroy the
hash table stored in the "limit_hash" private channel data but we
don't destroy the private data as it will be reclaimed as part of the
session. If limit increment is called after the limit release we can
reuse that channel private, but we need to check whether the hash
table is null first. Fortunately this makes the code look better
anyway.
FS-6775 #resolve
FS-6783 #resolve
2014-09-11 17:47:57 +00:00
bb84b0534c
Check for libpq in core before building mod_cdr_pg_csv
2014-09-10 22:29:24 +00:00
ce5d21106e
FS-6761 #resolve
2014-09-11 03:29:07 +05:00
30283b7f6b
Revert "fix libpq location detection"
...
This reverts commit e2b1ee26ae
2014-09-10 16:27:16 -05:00
e2b1ee26ae
fix libpq location detection
2014-09-10 15:04:50 -05:00
575adecf75
Revert "FS-6566"
...
This reverts commit 99d0f134e7
2014-09-10 14:57:57 -05:00
908dd26559
FS-6815 #comment force 1600hz for native g722
2014-09-10 14:09:32 -04:00
77c99b6306
FS-6806 #resolve #comment off by 1 error in last fix
2014-09-10 20:32:36 +05:00
66302cac89
FS-6811 #resolve
2014-09-09 17:59:29 -05:00
574d19e56e
mod_skinny: fix behavior of transfer when target extension falls through to voicemail - keep bridge from dropping out during that operation
2014-09-09 15:58:56 -05:00
Anthony Minessale