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Author SHA1 Message Date
Jeff Lenk
6146efd446 FS-6870 #comment please reopen if this doesnt fix vs2010 2014-10-07 22:28:53 -05:00
Mike Jerris
34bc98cafa Merge pull request #47 in FS/freeswitch from ~FLAVIO/freeswitch-fs-5106:master to master 
* commit '56535519043201c723467c66c772d7519a2b6f62':
  FS-5106 fire an event when a sip client doesn't respond to option-ping
 2014-10-07 14:06:34 -05:00
Anthony Minessale
2051a86df2 FS-6889 #resolve 2014-10-07 13:47:44 -05:00
Anthony Minessale
2514de94d2 fix obvious seg in setting a record file name to every participant and not checking for the recording member which does not have a session 2014-10-07 12:48:58 -05:00
Anthony Minessale
a4f840b947 more jb improvements 2014-10-07 12:48:58 -05:00
Mike Jerris
6860b41763 Merge pull request #83 in FS/freeswitch from ~MKVONARX/freeswitch-fs-6710:FS-6710 to master 
* 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
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
Mike Jerris
7802232f16 Merge pull request #65 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/OPENZAP-220-ftmod_libpri-don-t-close-channel to master 
* commit '7ec7c920d1ef37a6b9753db0321dd19b2f3332a9':
  OPENZAP-220 fix blocked into read and add cause for a correct hangup
 2014-10-07 11:39:15 -05:00
Chris Rienzo
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
Matteo Brancaleoni
7ec7c920d1 OPENZAP-220 fix blocked into read and add cause for a correct hangup 2014-10-07 14:34:39 +02:00
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
Markus von Arx
490efb7177 FS-6710: fix incorrect comparison for Min-SE values between SIP INVITE and local configuration 2014-10-07 10:41:36 +02:00
Anthony Minessale
da43bdeb12 add some calculations to jitter buffer related to judging the optimal size 2014-10-06 14:08:40 -05:00
Anthony Minessale
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
Anthony Minessale
f7210b2402 some more changes relates to new bypass media controls 2014-10-03 18:43:23 -05:00
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
Michael Jerris
acd8d74316 cleanup conditions 2014-10-03 12:48:43 -04:00
Anthony Minessale
bde2e2da51 FS-6889 #resolve 2014-10-03 11:34:42 -05:00
Anthony Minessale
6bed5d09a1 change type of int 2014-10-03 10:15:02 -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
Anthony Minessale
01bf42225c FS-6888 #resolve #comment fix regression from refactoring new feature 2014-10-03 10:17:41 -05:00
Jeff Lenk
d52cb335db fix trivial vs2010 build errors 2014-10-02 19:47:05 -05:00
Jeff Lenk
ae5d86515a FS-6884 #comment these were mostly simple warnings 2014-10-02 19:20:35 -05:00
Anthony Minessale
8db31f976f fix some recovery issues with dynamic payloads 2014-10-02 18:34:00 -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
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
Flavio Grossi
5653551904 FS-5106 fire an event when a sip client doesn't respond to option-ping 
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
Anthony Minessale
9486a645f8 bump 2014-10-01 18:35:04 -05:00
Anthony Minessale
cc44659a7c bump v1.5.14 2014-10-01 18:34:05 -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
Anthony Minessale
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
Jeff Lenk
b3d71917d2 FS-6870 #comment vs2010 and vs2012 would rather fix it this way 2014-10-01 17:53:51 -05:00
Jeff Lenk
661269a46f Revert "FS-6870 #vs2012 and vs2010 make download of openssl dependent" 
This reverts commit a39db86863f17fd82e578c05da935f28604b6bc5.
 2014-10-01 17:49:21 -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
Jeff Lenk
a39db86863 FS-6870 #vs2012 and vs2010 make download of openssl dependent 2014-09-30 21:30:48 -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