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12262 Commits

Author SHA1 Message Date
Artur Kraev
33cb950500 mod_managed: Added pure CreateStateHandlerDelegate in ManagedSession for native api usage 2014-11-03 00:42:00 +03:00
Artur Kraev
10ebebaae0 mod_managed: added console log level 2014-11-03 00:34:04 +03:00
Artur Kraev
4037e782a5 mod_managed: support per-module references directory 2014-11-03 00:32:30 +03:00
Artur Kraev
f3d089a998 mod_managed: not crash when cannot remove shadow directory (this sometimes happens when restarting from FS console) 2014-11-03 00:24:49 +03:00
Artur Kraev
7c0cf506d8 mod_managed: managedlist command must return value to api stream instead of log 2014-11-03 00:17:57 +03:00
Peter Olsson
7faf9f4c25 FS-6767 #comment Add initial support for mod_verto on Windows 2014-10-18 09:53:57 +02:00
Brian West
f1ee4ba4d7 Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch 2014-10-16 17:03:57 -05:00
Brian West
15e9e68064 FS-6927 #resolve #comment This display option ping times in the gateway status on sofia status gateways or individual gateway status output 2014-10-16 17:03:37 -05:00
Anthony Minessale
2e10407336 actual fix for commit cff5209ca3582994dae1353372e2f91b345ab959 which was in the wrong place 2014-10-16 16:04:15 -05:00
Anthony Minessale
3bdbdd4abf revert cff5209ca3582994dae1353372e2f91b345ab959 2014-10-16 14:39:59 -05:00
Mike Jerris
8b1fad813b Merge pull request #89 in FS/freeswitch from ~TMK/freeswitch:master to master 
* commit 'f5f6d15709427a1f8125b538fd4f753188bb3e16':
  add command 'action' with types 'reboot', 'reset', 'dialeddel', 'misseddel', 'receiveddel'
  Fix url encoding for snom remote commands (required to make # key work)
 2014-10-16 12:00:14 -05:00
Anthony Minessale
6b42e5a231 FS-6849 #comment change to 'import freeswitch' to only load it 2014-10-16 11:36:59 -05:00
Anthony Minessale
3cd0400d38 FS-6849 #resolve #comment scripts need to have 'from freeswitch import *' at the top. I added it explicitly to compensate. 2014-10-16 10:58:55 -05:00
Thomas Kleffel
f5f6d15709 add command 'action' with types 'reboot', 'reset', 'dialeddel', 'misseddel', 'receiveddel' 2014-10-14 23:26:48 +02:00
Thomas Kleffel
412f214809 Fix url encoding for snom remote commands (required to make # key work) 2014-10-14 22:31:40 +02:00
Mike Jerris
e898770e69 Merge pull request #64 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/FS-6400-improve-sip-ping-generation-by-distributing to master 
* commit 'beb1d1792134f61a252538d45af909ee50771017':
  FS-6400 Improve sip ping generation by distributing them across an interval
 2014-10-14 11:59:43 -05:00
Marc Olivier Chouinard
2ca349a3f8 FS-6910 #resolve Multiple entry with the same first, last name or extension in the directory would only return 1 entry. Fix issue where group by would produce multiple row of count(*) result. Using distinct instead wouldn't solve the issue in SQLITE because of a bug, so solution is to use a subselect. 2014-10-14 09:53:12 -04:00
Matteo Brancaleoni
beb1d17921 FS-6400 Improve sip ping generation by distributing them across an interval 2014-10-14 14:24:21 +02:00
Anthony Minessale
cff5209ca3 fix leak of nua handle due to reference counting that must be between 3 to 7 years old. Effects all calls with auth/challenge on INVITE 2014-10-13 18:06:32 -05:00
Anthony Minessale
e4e9b1b9f9 have resume media on hold not send invite back out at the holder but rather enable media in the 200ok 2014-10-10 16:09:43 -05:00
Travis Cross
b5294c53d6 Fix crash on transport=tls with non-TLS profile 
We use the transport of the Contact header of the remote UAC to decide
which of our own Contact addresses we should use when replying to a
SUBSCRIBE or sending a presence NOTIFY.

If TLS is not enabled on a Sofia profile, then the TLS Contacts for
that profile are NULL.  Unfortunately we were using these NULL values
uncritically when the remote UAC sent us a Contact header with a TLS
transport and our own Sofia profile did not have TLS enabled.

With this commit we fall back to our TCP Contact address when the
remote Contact is TLS and our Sofia profile does not have TLS enabled.
 2014-10-10 18:36:37 +00:00
Michael Jerris
855cc4b4e0 add 908-retry-seconds gateway param to set reg retry time when getting a 908 for backup interfaces to connect quickly 2014-10-09 14:43:23 -04:00
Chris Rienzo
28bc992fce mod_rayo: fix error in SRGS grammar parser... <one-of><item>7</item><item>715</item></one-of> will return MATCH_END with input of 7 instead of MATCH since 715 is a potential match with further input. 2014-10-09 11:41:22 -04:00
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
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
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
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
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
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