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
7802232f16
Merge pull request #65 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/OPENZAP-220-ftmod_libpri-don-t-close-channel to master
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* commit '7ec7c920d1ef37a6b9753db0321dd19b2f3332a9':
OPENZAP-220 fix blocked into read and add cause for a correct hangup
2014-10-07 11:39:15 -05:00
4a5e36d63e
switch_pgsql.c switch_pgsql_next_result_timed() was using switch_time_now() for start time and switch_micro_time_now() for current time. These are different time sources that may not be in sync and could cause the query to timeout prematurely.
2014-10-07 09:33:19 -04:00
7ec7c920d1
OPENZAP-220 fix blocked into read and add cause for a correct hangup
2014-10-07 14:34:39 +02:00
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
da43bdeb12
add some calculations to jitter buffer related to judging the optimal size
2014-10-06 14:08:40 -05:00
397ec5ae1d
fix jb bug where once its full size it will never shrink due to logic err
2014-10-06 09:50:13 -05:00
f7210b2402
some more changes relates to new bypass media controls
2014-10-03 18:43:23 -05:00
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
acd8d74316
cleanup conditions
2014-10-03 12:48:43 -04:00
bde2e2da51
FS-6889 #resolve
2014-10-03 11:34:42 -05:00
6bed5d09a1
change type of int
2014-10-03 10:15:02 -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
01bf42225c
FS-6888 #resolve #comment fix regression from refactoring new feature
2014-10-03 10:17:41 -05:00
d52cb335db
fix trivial vs2010 build errors
2014-10-02 19:47:05 -05:00
ae5d86515a
FS-6884 #comment these were mostly simple warnings
2014-10-02 19:20:35 -05:00
8db31f976f
fix some recovery issues with dynamic payloads
2014-10-02 18:34:00 -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
afb00b2ecc
Force rport on ADTRAN TA Devices
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ADTRAN Total Access devices do not support sending the rport parameter in
the Via header. This allows us to detect the device and force rport when
using the "safe" parameter, enabling the device to be used behind NAT.
FS-6823 #resolve
2014-10-02 13:09:15 -07:00
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
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
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
5653551904
FS-5106 fire an event when a sip client doesn't respond to option-ping
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When all-reg-options-ping is enabled, this adds a new custom event to mod_sofia
(sofia::sip_user_state), which is fired when a client stops responding to such
ping packets (or when it is reachable again).
Add two needed new columns to the sip_registrations table:
- ping_status, which is "Reachable" or "Unreachable" depending on the client
status;
- ping_count, which tracks the number of ping responses received and is used
to provide some kind of hysteresis to avoid firing the event in case of
transitory network failures.
Then ping_count is checked against two threshold values, sip-user-ping-min
and sip-user-ping-max in a similar fashion as the ping-{max,min} options for
the gateways. These two values are configurable in the profile's xml
configuration file.
Also, if unregister-on-options-fail is enabled, the client is unregistered
based on the number of OPTIONS failure which is also checked against the
sip-user-ping-{min,max} values.
2014-10-02 12:34:47 +02:00
9486a645f8
bump
2014-10-01 18:35:04 -05:00
cc44659a7c
bump
2014-10-01 18:34:05 -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
35aeae0170
FS-6822 #comment The code in question appears to have been added by me ( 18f20e24). I think this patch is the correct solution.
2014-10-01 18:11:01 -05:00
b3d71917d2
FS-6870 #comment vs2010 and vs2012 would rather fix it this way
2014-10-01 17:53:51 -05:00
661269a46f
Revert "FS-6870 #vs2012 and vs2010 make download of openssl dependent"
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This reverts commit a39db86863
2014-10-01 17:49:21 -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
a39db86863
FS-6870 #vs2012 and vs2010 make download of openssl dependent
2014-09-30 21:30:48 -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
92a66fb1e7
improve adaptive jitter buffer ascending check
2014-09-30 22:54:46 +05:00
56edfc7062
Merge pull request #76 in FS/freeswitch from ~HRISTO/freeswitch:fix-ptime-on-reinvite-master to master
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* commit 'fbe857e6fafabbca6a64584c51316ccc5e6ba96e':
fix ptime from known broken endpoints on re-invite
2014-09-30 10:53:37 -05:00
0150c862a2
FS-6854 #comment try this patch
2014-09-30 20:35:19 +05:00
6ac26fcc3e
Update timezones
2014-09-30 09:58:42 -05:00
f3473fdb58
Merge pull request #71 in FS/freeswitch from ~RTRELEAVEN/freeswitch-fs-4762-1:fs-4762 to master
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* commit '139b03204550cd394877da882fec49b08eba08fa':
improve regular expression to parse Jerusalem timezone files
2014-09-30 09:55:56 -05:00
4590220b53
Merge pull request #74 in FS/freeswitch from ~DDRAGIC/freeswitch:gsmopen_completion to master
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* commit 'a94fbe807905be714c774f7479936387b31602b2':
mod_gsmopen: add tab completion for api commands
2014-09-30 09:41:28 -05:00
812f9112a5
Fix reference to libressl in makefile on OpenBSD
2014-09-30 09:06:26 -05:00
Mike Jerris
Chris Rienzo
Matteo Brancaleoni
Markus von Arx
Anthony Minessale
Jeff Lenk
Spencer Thomason
Flavio Grossi
Brian West
Anthony Minessale II