db66cdb56f
Fix mrcp libraries to build right
2014-12-03 12:56:48 -05:00
35ba6a33b1
FS-6766, fix verto caller ringback missing on conference bridge
2014-12-03 10:16:22 +01:00
80649df65c
fix FS-7049 - Count and list agents based on their state
2014-12-02 22:57:43 -03:00
35558993c9
FS-5816 #resolve #comment re-add completion cause to session record stop event
2014-12-02 15:00:54 -05:00
2a7b022733
FS-6980 #resolve don't crash when using native recording on recordstop the redo
2014-12-02 10:51:48 -06:00
72c3df5ed3
FS-6891 FS-6713 #comment revert 1b612fecb6
2014-12-02 16:46:08 -06:00
a8c5a0c87b
add timezone support to mod_say_{de,es,ja,nl,th,zh}
...
This is nothing more than a shameless copy/paste from another mod_say
module, which already had timezone support. It simply checks if the
timezone variable is set and if it contains a valid timezone, then this
timezone will be used when announcing times/dates.
FS-7048 #resolve
2014-12-02 17:08:12 +01:00
bad5dc3022
FS-7037 #resolve
2014-12-01 15:10:00 -06:00
dc9e9042d1
#comment FS-7025 fix compiler warning introduced from e55aee14
2014-12-01 09:40:18 +08:00
e55aee14bb
FS-7025 %FEATURE #comment please test
2014-11-30 16:55:08 -06:00
5bbef7f1e5
FS-7015 #comment please test
2014-11-25 17:02:10 -06:00
79de78a0fb
FS-7021 #resolve
2014-11-19 21:51:09 -06:00
74f243bc3b
Merge pull request #95 in FS/freeswitch from ~ITALOROSSI/freeswitch:master to master
...
* commit 'f87c335e8a49da7b0a2b6f19b45f285f6355e2e0':
Only waiting for agent channel to hangup.
2014-11-19 15:32:19 -06:00
62a2e10e41
remove hack breaking some cross compile builds. if you really need this, you should be using new enough glibc anyways
2014-11-19 16:06:42 -05:00
df423b88d6
improve timerfd implementation to be more accurate
2014-11-18 17:14:04 -06:00
878a04715a
revert
2014-11-18 17:00:44 -06:00
da6043f353
improve timerfd implementation to be more accurate
2014-11-18 16:59:58 -06:00
7b80b6249b
Merge branch 'unimrcp-update-20141117'
2014-11-18 14:28:38 -05:00
8330336e9a
FS-6450 [unimrcp] Update library
...
[apr] Backport APR_RING_FOREACH and APR_RING_FOREACH_SAFE macros to APR for unimrcp compatibility.
[unimrcp] configure.gnu - need full path, not relative path for library paths
[unimrcp] added uni_revision.h - couldn't get it to autogenerate from build
[mod_unimrcp] add better logging and error checking on module load. Currently dumps core on MRCPv1 TTS attempt
[mod_unimrcp] don't configure MRCPv1 session with a connection agent- causes crash
2014-11-18 14:23:54 -05:00
5127b64df8
FS-7014: [mod_sofia] don't touch the tech_pvt when a call has just ended, leaving us with a null tech_pvt
2014-11-18 10:22:13 -06:00
8eaaa083ad
FS-6622: [mod_shout] set buffer size for streams based on the number of channels to avoid buffer starvation
2014-11-18 09:47:43 -06:00
424df19083
FS-6695: fix build on mips
2014-11-17 15:25:39 -05:00
7c0da5cc40
FS-6957 fix regression
2014-11-17 10:36:03 -06:00
250234da76
FS-5800: [mod_curl] add support for additional curl auth methods
2014-11-17 11:01:35 -05:00
f198d82bac
FS-5666: [mod_redis] add ignore_connect_fail config setting to not kill the call when redis is down when using redis backend for limit
2014-11-17 10:29:47 -05:00
6f660c3a10
iksemel - remove support for SSLv23 in iks_proceed_tls (was a FS addition to iksemel). mod_rayo - updated to no longer support SSLv23
2014-11-17 09:46:23 -05:00
311d0766af
Merge pull request #113 in FS/freeswitch from ~FLAVIO/freeswitch:bugfix/FS-7004-mod_sndfile_respect_umask to master
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* commit 'c73afe1c85f42d076c4b71c80251f7c888d47756':
FS-7004 mod_sndfile: respect umask when creating new files
2014-11-17 08:29:08 -06:00
0cf770a836
FS-6996: #resolve fix define change as of glibc 2.20 for _BSD_SOURCE -> _DEFAULT_SOURCE
2014-11-17 09:27:22 -05:00
c73afe1c85
FS-7004 mod_sndfile: respect umask when creating new files
...
Files created by mod_sndfile use sf_open() which uses hardcoded permissions. To
respect the process' umask, manually open files with the correct permissions
and then call sf_open_fd().
2014-11-17 11:26:31 +01:00
f1df8d6096
Allow setting CURL timeout from curl API command
...
Previously the `timeout` option to the curl API command set only
`CURLOPT_CONNECTTIMEOUT` -- the maximum amount of time that curl will
wait to connect to the server. If the server accepted the connection
but then never replied, curl would wait essentially forever. There
was no way to set `CURLOPT_TIMEOUT` -- the maximum amount of time the
entire request operation is allowed to take.
With this change, the `timeout` option sets `CURLOPT_TIMEOUT`. We've
earlier added a `connect-timeout` option to set
`CURLOPT_CONNECTTIMEOUT`.
This is a change to existing behavior. However, it's likely that this
is what people expected it to do all along. The curl application
call, for example, accepts both `curl_connect_timeout` and
`curl_timeout` channel variables, with the latter setting
`CURLOPT_TIMEOUT`.
If people really were relying on this odd behavior, we'll rename the
option with the new behavior to something else and come up with a
transition plan.
2014-11-16 19:31:00 +00:00
1ee325df48
Add `connect-timeout` option for curl API command
...
This option sets the maximum number of seconds that curl will wait to
connect with the server.
Right now this is a synonym for the `timeout` option.
2014-11-16 19:31:00 +00:00
07c5cc18ba
FS-7003 #resolve #comment mod_rayo: fix infinite loop when output sent to server without SSML configured and repeat-times=0
2014-11-14 14:20:48 -05:00
e1c0ef5008
mod_rayo: new configuration parameter, add-variables-to-offer (default=false). When true, all channel variables are included in the offer to rayo client
2014-11-14 13:22:53 -05:00
826d428741
FS-6992 [mod_opus] global configuration or maxplaybackrate and maxaveragebitrate
...
from opus.conf.xml
2014-11-14 10:31:32 +01:00
dd61232163
FS-6979 #resolve #comment mod_http_cache: added base-domain config to s3 profiles so mod_http_cache can access self hosted s3 compatible service.
...
Example configuration:
<profiles>
<profile name="s3">
<!-- Credentials for AWS account. -->
<aws-s3>
<!-- 20 character key identifier -->
<access-key-id><![CDATA[AKIAIOSFODNN7EXAMPLE]]></access-key-id>
<!-- 40 character secret -->
<secret-access-key><![CDATA[wJalrXUtnFEMI/K7MDENG/bPxRfiCYEXAMPLEKEY]]></secret-access-key>
<base-domain><![CDATA[example.com]]></base-domain>
</aws-s3>
<!-- Domains that this profile applies to -->
<domains>
<domain name="bucket.example.com"/>
<domain name="bucket2.example.com"/>
</domains>
</profile>
</profiles>
2014-11-13 10:20:02 -05:00
75473a70b6
FS-6531: #resolve set to tag on uuid_phone_event notify to make grandstream happy, even tho they could have matched the dialog fine off the from tag like every other phone does.
2014-11-12 21:55:31 -06:00
82aa33140e
FS-6531: #resolve set to tag on auto answer notify to make grandstream happy, even tho they could have matched the dialog fine off the from tag like every other phone does.
2014-11-12 21:37:14 -06:00
65502293cf
FS-6890 #comment revert
2014-11-12 13:09:39 -06:00
1cbbc14724
Merge pull request #110 in FS/freeswitch from ~JCHAVANTON/freeswitch-opus:FS-6947-opus to master
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* commit '0eefdca47b127f8151c9b1a0b12eaf2d7e99def8':
FS-6947 Opus RTP payload fmtp settings ( maxaveragebitrate / maxplaybackrate )
2014-11-12 12:57:31 -06:00
0eefdca47b
FS-6947 Opus RTP payload fmtp settings ( maxaveragebitrate / maxplaybackrate )
2014-11-12 17:51:56 +01:00
07030c63f0
fix compiler warning on unmatched return type
2014-11-12 13:31:30 +08:00
dd629c1516
add external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media
2014-11-12 08:44:20 +08:00
fada4b893a
FS-6977 #resolve
2014-11-11 18:18:32 -06:00
dbc5571594
FS-6983 wrap new curl TLS macro usage with ifdefs
2014-11-11 16:26:44 -06:00
09abee2492
Merge pull request #108 in FS/freeswitch from ~NNEUL/freeswitch:feature-dialplan-tstamps to master
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* commit 'f175c7118879b882343da9b6f15075161923fcca':
FS-6805 add support for logging full timestamps with dialplan, defaults to old behavior unless requested
2014-11-11 15:19:21 -06:00
33d37ce0f5
PLIV-13 #resolve
2014-11-11 14:51:19 -06:00
0c68bb6d89
FS-6957 #resolve
2014-11-11 13:37:46 -06:00
f175c71188
FS-6805 add support for logging full timestamps with dialplan, defaults to old behavior unless requested
2014-11-11 13:25:47 -06:00
a279bf38af
FS-6890 #comment please test
2014-11-11 12:56:40 -06:00
c79360c596
reduce logging when level is below 9 for less important messages
2014-11-11 09:11:44 -06:00
ab24bde262
FS-5533 fix issue with busy signal being sent back to all shared lines instead of just the calling device
2014-11-11 08:41:16 -06:00
34cf3b9069
FS-6980 #resolve don't crash when using native recording on recordstop
2014-11-11 07:45:50 -06:00
0f8b993769
fix mod_say_es_ar Makefile.am
2014-11-10 11:12:56 -05:00
11e62dd40d
Revert "Revert "FS-6967 New mod_say_es_AR to support Argentina Spanish variant.""
...
This reverts commit d9d9510ce4
2014-11-10 11:12:15 -05:00
d9d9510ce4
Revert "FS-6967 New mod_say_es_AR to support Argentina Spanish variant."
...
This reverts commit e75d0675afebb3c8fbfa
2014-11-08 15:53:57 -06:00
a3a80401fd
fix regression caused by missing ! char in commit: 4eb5b388
2014-11-07 17:11:47 -06:00
5ce5199be9
FS-6969 #resolve #comment This patch should accomplish the same and handle other platforms, please test
2014-11-07 08:38:01 -06:00
b7741916eb
Merge pull request #49 in FS/freeswitch from ~VIPKILLA/freeswitch-mod_odbc_cdr:master to master
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* commit '544c5faf5e6ce6fe2b87523304d1f00e2d201d90':
Add module mod_odbc_cdr
2014-11-07 08:34:55 -06:00
bf7c161d36
Merge pull request #68 in FS/freeswitch from ~HRISTO/freeswitch:interrupt-conf-moh to master
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* commit '94278b5e545b58bad784a95da6181fc5f299457f':
allow enter and exit sounds to interrupt the MOH in a wait_mod conference
2014-11-07 08:09:05 -06:00
9939671b50
Merge pull request #102 in FS/freeswitch from ~AMARUS_CAMERON/freeswitch:mod_fifo-outbound_per_cycle_min to master
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* commit '1944f9a5ee63ec51bed1bfb900072d168a81d004':
FS-6968 Changes to mod_fifo.c to add outbound_per_cycle_min
2014-11-07 08:06:54 -06:00
9e9c4378de
Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch
2014-11-06 18:26:28 -06:00
415f82fe93
FS-6954 #resolve #comment please test
2014-11-06 18:26:16 -06:00
1944f9a5ee
FS-6968 Changes to mod_fifo.c to add outbound_per_cycle_min
2014-11-06 19:21:58 -05:00
60b51c46cc
Merge pull request #50 in FS/freeswitch from ~JMESQUITA/freeswitch:mod_say_es_AR to master
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* commit 'e75d0675afd8974687931143709814544299fadc':
FS-6967 New mod_say_es_AR to support Argentina Spanish variant.
2014-11-06 17:21:08 -06:00
f66f2cae8c
FS-6890 #comment please test
2014-11-06 17:13:02 -06:00
e75d0675af
FS-6967 New mod_say_es_AR to support Argentina Spanish variant.
2014-11-06 19:21:13 -03:00
cf1424cfe5
mod_rayo: update config to use spandsp_start_tone_detect and spandsp_stop_tone_detect
2014-11-06 14:54:55 -05:00
0e9e8a9bd6
Renaming mod_spandsp's cadence + tone detection APPs and APIs (start_tone_detect / stop_tone_detect) to
...
spandsp_start_tone_detect and spandsp_stop_tone_detect to resolve conflict with mod_dptools'
tone_detect/stop_tone_detect APPs.
2014-11-06 14:46:56 -05:00
94278b5e54
allow enter and exit sounds to interrupt the MOH in a wait_mod conference
...
This patch does the following:
* only starts MOH if no other file (sync or async) is currently playing
* adds a variable "conference_permanent_wait_mod_moh" that controls the
behavior of how the enter and exit sounds interact with the MOH when
wait_mod is set. When the variable is set, the MOH keeps playing and
the enter and exit sounds are mixed with the MOH. When the variable
is unset, then any playing MOH is first stopped, then the enter or
exit sound is played and the MOH is started again.
This functionality is useful in case the enter and exit sounds are
used to announce the name of the caller, who is joining or leaving a
conference.
FS-5159 #resolve
2014-11-06 19:32:17 +01:00
e86d359443
Merge pull request #57 in FS/freeswitch from ~LEKENSTEYN/freeswitch:fixes to master
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* commit '8e4423f126b9476123c6fa8c41c5f8ebfe1d0cb5':
Document Dbh.test_reactive, return saner values
2014-11-06 11:48:12 -06:00
7b7685484d
Merge pull request #101 in FS/freeswitch from ~RAVENOX/freeswitch:fixes_for_pull_request_#93 to master
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* commit 'c97c0e8a7805f15fd52e7400b3a984d981e2524a':
Fix for case-sensitive filesystems
Fixed code formatting. Return ListDelegate
2014-11-06 11:42:35 -06:00
1190e59adf
FS-6965 #resolve
2014-11-06 09:47:54 -06:00
bc767bb35a
Adding rfc6598.auto and adding rfc6598 space to nat.auto acl, This is the NAT444 carrier grade nat space
2014-11-06 08:55:03 -06:00
c97c0e8a78
Fix for case-sensitive filesystems
2014-11-06 12:36:02 +03:00
14bf6c1604
Fixed code formatting.
...
Return ListDelegate
2014-11-06 12:29:10 +03:00
fbe81ff886
Merge pull request #69 in FS/freeswitch from ~HRISTO/freeswitch:conference-flags to master
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* commit '3695bdd9e4e0d925ab7296d2f4ce120bf656f623':
set conference flags from a dial plan variable or via +flags{ }
2014-11-05 16:25:40 -06:00
78cab12dd2
Merge pull request #48 in FS/freeswitch from ~ANTONIO/freeswitch-fs-6809:master to master
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* commit '69d5cda6d67074d6e5c1b7038b4dd7cab0adf60f':
resolve FS-6809
2014-11-05 16:05:00 -06:00
9c1e6037c9
FS-6954 #comment we fixed another bug and this is the side effect which is completely valid, too bad you can never fix broken t38 endpoints. Can you please test this patch
2014-11-05 11:51:30 -06:00
a4971693d3
FS-6890 #comment please test
2014-11-05 11:35:16 -06:00
4eb5b38848
fix bug where re-invites needlessly re-init the codec and jb
2014-11-05 11:35:16 -06:00
01395c508b
FS-6916 fix typos in code
2014-11-05 10:58:35 -06:00
8478874ab9
FS-6831 while you can already unset by calling set with var=, same with set in dialplan this is a convience function similar to our unset in dialplan
2014-11-05 09:44:24 -06:00
2b4082c236
Merge pull request #93 in FS/freeswitch from ~RAVENOX/freeswitch:mod_managed_improvements to master
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* commit '889b678e58bf38eb86df7885b8f054d3d9d92d74':
mod_managed: Added GetPtr to Util class for internal pointers extraction (very useful when using native api)
mod_managed: Added pure CreateStateHandlerDelegate in ManagedSession for native api usage
mod_managed: added console log level
mod_managed: support per-module references directory
mod_managed: not crash when cannot remove shadow directory (this sometimes happens when restarting from FS console)
mod_managed: managedlist command must return value to api stream instead of log
2014-11-04 16:35:11 -06:00
32a9ff3d39
Merge pull request #60 in FS/freeswitch from ~SJTHOMASON/freeswitch:FS-6823 to master
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* commit 'afb00b2ecc8a9b049801f3f475c80e1111070fa8':
Force rport on ADTRAN TA Devices
2014-11-04 07:36:36 -06:00
4b87056625
remove debugging printf
2014-11-03 14:17:24 -06:00
91bc09525f
Fix copy and paste error CID 1250786
2014-11-03 14:15:04 -06:00
b5d8fa9f46
FS-6956 modify commit 16365501 to use one single line which is only tolerable way to have bracket-less if stmts.
2014-11-03 14:03:12 -06:00
8f3c157f12
FS-6957 #resolve
2014-11-03 11:32:40 -06:00
f87c335e8a
Only waiting for agent channel to hangup.
...
If the member do an attended transfer this loop stops and
the agent is set to Available state again, when in fact he still
busy with other channel.
This was happening, for example, when a member calls support queue, then
the support operator do an attended transfer to sales queue, while the sales
operator is talking the member that was transferred, the sales queue will
send calls to the operator, which is not expected.
2014-11-03 12:02:03 -03:00
889b678e58
mod_managed: Added GetPtr to Util class for internal pointers extraction (very useful when using native api)
2014-11-03 00:43:59 +03:00
33cb950500
mod_managed: Added pure CreateStateHandlerDelegate in ManagedSession for native api usage
2014-11-03 00:42:00 +03:00
10ebebaae0
mod_managed: added console log level
2014-11-03 00:34:04 +03:00
4037e782a5
mod_managed: support per-module references directory
2014-11-03 00:32:30 +03:00
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
7c0cf506d8
mod_managed: managedlist command must return value to api stream instead of log
2014-11-03 00:17:57 +03:00
7ca4ac566c
FS-5949 FS-6945 #comment this change should be relevant to both of these issues, please test. This patch improves the hold parsing and ignores connection address of 0 implying hold when ice is present and disables the auto interpretation by the lower level stack of 0.0.0.0 to automatically imply sendonly to allow FS to decide on its own
2014-10-31 13:49:39 -05:00
39be877760
One place we said Failed Registration, the other we said Registration Failed, lets try to be consistent.
2014-10-30 10:40:52 -05:00
52ae551d1a
FS-6954 #resolve #comment technically the new way is more correct but there is no hope for making fax endpoints follow a real spec. This should take care of it.
2014-10-30 10:15:10 -05:00
429912e34f
git push
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Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch
2014-10-29 16:01:30 -05:00
3b9f0c32e6
FS-6927 #comment allow sub millisecond resolution for option ping times
2014-10-29 16:01:28 -05:00
43e6146ece
translate dtmf directly to b leg in bypass call in verto
2014-10-29 13:14:55 -05:00
cba216ca27
Merge pull request #56 in FS/freeswitch from ~OLEGSTOLYAR/freeswitch:master to master
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* commit '49a3672e4206a2a730220ec4bc2426274441ef64':
Add ability to log commands executed in mod_xml_rpc
2014-10-29 12:59:43 -05:00
30e793a7ef
fix missing paren in logic for hard mute enter muted mode
2014-10-29 12:16:32 -05:00
49a3672e42
Add ability to log commands executed in mod_xml_rpc
2014-10-29 17:08:21 +00:00
5488757917
FS-6950 #resolve
2014-10-28 15:01:08 -05:00
f85c82446c
FS-6949 #resolve
2014-10-28 14:19:06 -05:00
443ab8a8db
FS-5949 #resolve
2014-10-28 13:38:06 -05:00
9a2ac280ba
Merge pull request #90 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/FS-6756-metadata-in-mp3-recordings-are-not to master
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* commit '3f0d6b3f2d548de7d07d8503a1f73f381e3d1a72':
FS-6756 lame_init_params must be called after setting all id3tag stuff, otherwise id3 tags will not be written. So, instead of calling it early, revert FS-3646 and add a check on free_context to really do lame stuff only if lame has been set ready, avoid seg faults in some corner cases.
2014-10-28 09:46:53 -05:00
75815877d6
FS-6688 #resolve
2014-10-27 14:14:21 -05:00
d1e529aefd
Add new hard_mute control to allow apps to request low level mute e.g. from the rtp stack level. Its used in mod_conference to avoid reading audio while muted and possibly reduce some transcoding load
2014-10-27 15:13:42 -04:00
0386db7dfd
add some asserts to catch buffer overflow
2014-10-27 15:13:42 -04:00
f772b400bf
FS-6939 please do pull request next time. ;)
2014-10-27 14:12:55 -05:00
bea7d8ec71
FS-5853 #resolve #comment mod_rayo now reports record completion cause
2014-10-27 13:41:52 -04:00
26af9c3d67
FS-6939 #resolve
2014-10-27 12:23:44 -05:00
b25ae6ab6c
FS-5816 #resolve #comment Record-Completion-Cause added to session recording RECORD_STOP event and record_completion_cause channel variable added.
2014-10-27 13:17:38 -04:00
a43e3496c2
FS-6921 #resolve #comment rayo APP now accepts optional comma separated list of JIDs or user names to steer incoming calls to specific rayo clients
2014-10-27 12:18:17 -04:00
57e8231cba
FS-6929 #resolve #comment fix deadlock in mod_rayo
2014-10-27 10:05:12 -04:00
12b6940644
update jb command parser
2014-10-24 15:26:44 -05:00
8d720d5bcc
FS-6940 #resolve #comment %FEATURE use the variable digits_dialed_filter to set regular expressions with () captures and anything matched will be replaced with X's in the CDR
2014-10-23 12:47:27 -05:00
3f0d6b3f2d
FS-6756 lame_init_params must be called after setting all id3tag stuff, otherwise id3 tags will not be written.
...
So, instead of calling it early, revert FS-3646 and add a check on free_context to really do lame stuff only
if lame has been set ready, avoid seg faults in some corner cases.
2014-10-22 12:31:21 +02:00
7faf9f4c25
FS-6767 #comment Add initial support for mod_verto on Windows
2014-10-18 09:53:57 +02:00
1f9025d446
FS-6926 #resolve #comment please test and reopen if necessary
2014-10-16 17:57:46 -05:00
f1ee4ba4d7
Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch
2014-10-16 17:03:57 -05:00
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
2e10407336
actual fix for commit cff5209ca3 which was in the wrong place
2014-10-16 16:04:15 -05:00
3bdbdd4abf
revert cff5209ca3
2014-10-16 14:39:59 -05:00
8b1fad813b
Merge pull request #89 in FS/freeswitch from ~TMK/freeswitch:master to master
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* 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
6b42e5a231
FS-6849 #comment change to 'import freeswitch' to only load it
2014-10-16 11:36:59 -05:00
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
f5f6d15709
add command 'action' with types 'reboot', 'reset', 'dialeddel', 'misseddel', 'receiveddel'
2014-10-14 23:26:48 +02:00
412f214809
Fix url encoding for snom remote commands (required to make # key work)
2014-10-14 22:31:40 +02:00
e3e267f462
vs2010 trivial compiler warnings
2014-10-14 13:20:52 -05:00
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
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
beb1d17921
FS-6400 Improve sip ping generation by distributing them across an interval
2014-10-14 14:24:21 +02:00
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
e245e90761
fix some jitterbuffer constants
2014-10-13 13:05:57 -05:00
9bd3bd30d3
FS-6911 #resolve
2014-10-13 10:36:51 -05:00
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
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
66dafbde8c
FS-6902 #comment add patch to make this problem obvious and fail on record and playback
2014-10-09 16:53:38 -05:00
43c2c6dd24
FS-6815 FS-6903 #resolve
2014-10-09 15:47:36 -05:00
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
b578aa7c9e
Merge pull request #86 in FS/freeswitch from ~HRISTO/freeswitch:proper-ptime-detection-with-packet-loss to master
...
* commit 'd48057e23f97af11cbdc482b20a06eaba776ea82':
account for lost frames during ptime detection
2014-10-09 11:53:25 -05:00
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
d48057e23f
account for lost frames during ptime detection
...
This allows the "broken ptime" detection to work correctly when packet
loss is present on the wire. In addition to the timestamps this patch
adds frame sequence tracking and corrects the timestamp difference
only as needed and according to the number of lost packets.
FS-6898 #resolve
2014-10-09 11:37:52 +02:00
2eb117bbe9
minor jb improvement
2014-10-08 13:10:15 -05:00
544c5faf5e
Add module mod_odbc_cdr
2014-10-08 08:31:35 -04:00
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
2051a86df2
FS-6889 #resolve
2014-10-07 13:47:44 -05:00
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
a4f840b947
more jb improvements
2014-10-07 12:48:58 -05:00
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
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
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
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
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
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
...
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.
...
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
...
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
83acda0413
file_string write failover FS-4930
2014-10-02 09:16:01 +02:00
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
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
...
* 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
4590220b53
Merge pull request #74 in FS/freeswitch from ~DDRAGIC/freeswitch:gsmopen_completion to master
...
* commit 'a94fbe807905be714c774f7479936387b31602b2':
mod_gsmopen: add tab completion for api commands
2014-09-30 09:41:28 -05:00
3695bdd9e4
set conference flags from a dial plan variable or via +flags{ }
...
This patch allows conference flags to be set dynamically from the
dial plan by either passing them to the conference application in
the +flags{ } string or by setting the "conference_flags" dial plan
variable.
The +flags{ } string is currently used to set *user* flags only.
This patch changes this by allowing the +flags{ } string to contain
conference related flags as well (for example wait_mod). It shouldn't
be a problem to pass both types of flags via +flags{ } as long as
the user and conference flag names are kept unique.
FS-5099 #resolve
2014-09-30 11:31:03 +02:00
fbe857e6fa
fix ptime from known broken endpoints on re-invite
...
Freeswitch tries to fix timing issues (wrong ptime) on re-invite the same way
it does for the initial invite. This results in small audio glitches, while it
sends a couple of packets with different ptime, before the timing detection
logic figures out the remote (broken) endpoint true ptime.
In order to avoid unnecessary timing changes, this patch overwrites the
advertised ptime from known broken endpoints with the ptime, which was detected
by freeswitch. It does this by checking if the sip_h_X-Broken-PTIME (1.2.x) or
rtp_h_X-Broken-PTIME (master) variables are set.
FS-6644 #resolve
2014-09-30 11:19:35 +02:00
da51603a2c
FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 #resolve #comment 5 bugs one typo. From commit 1b612fecb6
2014-09-29 19:26:32 +05:00
e94af49e1e
revert
2014-09-29 19:26:01 +05:00
d619017621
FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 5 bugs one typo. From commit 1b612fecb6
2014-09-29 19:21:01 +05:00
a94fbe8079
mod_gsmopen: add tab completion for api commands
2014-09-29 13:25:30 +02:00
dac4afbfdb
this was alraedy in there, whoops
2014-09-28 10:40:57 -04:00
c1e9b0d414
expose apr socket put
2014-09-27 15:02:41 -07:00
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
...
* 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
...
* 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
c018c28738
FS-6851 #resolve
2014-09-24 20:40:27 +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
e8d6866899
use the more reliable offset_pos counter in file position parsing for seek in scripts
2014-09-23 21:01:25 +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
372455c30a
FS-6829 #resolve
2014-09-19 02:28:47 +05:00
8f85b5204c
vs2010 trival compiler warnings
2014-09-17 18:11:20 -05:00
1f5bb3470d
mod_skinny: avoid truncation of non-null-terminated strings in protocol
2014-09-17 11:13:15 -05:00
d2f8fca18a
FS-6825 #resolve #comment caused by regression in commit 0732c0b0 pertaining to FS-6825
2014-09-17 20:32:18 +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
16d947dd7a
can't have asserts here after all
2014-09-17 02:14:54 +05:00
b2917e06db
improve ssl errors
2014-09-17 02:14:43 +05:00
47ae1837d5
add some asserts
2014-09-16 20:44:10 +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
dca7bdde77
Merge pull request #55 in FS/freeswitch from ~DDRAGIC/freeswitch:bugfix/FS-6821-mod_gsmopen-wrong-interface-name-in-log to master
...
* commit 'f262dbce948e6043d48d7859da00fa7db5b47585':
FS-6821 mod_gsmopen: fix interface name in log
2014-09-15 14:31:49 -05:00
f924684eff
FS-6623 #resolve fix init and logging for rtcp
2014-09-15 20:08:09 +05:00
b738775876
[FS-6623] implement RTCP report generation
2014-09-15 20:08:09 +05:00
8e4423f126
Document Dbh.test_reactive, return saner values
...
In the FreeSWITCH core, the return value of switch_case_db_test_reactive
is ignored, but it is usable in LUA modules (and other bindings via
SWIG). The LUA API example[1] shows how to check the return value, but
that example miserably fails if the database did not exist before.
Changes:
- Document the expected behavior of the test_reactive function.
- Assert that test_sql and sql_reactive are both given. If either
query is not given, the caller is using the wrong API.
- When SCF_AUTO_SCHEMAS is cleared, use the return value of the
test_sql execution. Does anybody use this? Why not remove it?
- Do not unconditionally return SWITCH_FALSE when test_sql fails,
instead allow it to become SWITCH_TRUE when reactive_sql passes.
- Remove the unnecessary test_sql check for SCDB_TYPE_CORE_DB
(this is now enforced through an assert check). (+reindent)
- Clarify the error message of drop_sql, prepending "Ignoring" to
the "SQL ERR" message.
- LUA: Do not print "DBH NOT Connected" if the query fails. This was
the initial source of confusion.
[1]: https://confluence.freeswitch.org/display/FREESWITCH/Lua+API+Reference
2014-09-15 15:39:08 +02:00
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
3e8e2ce151
Revert commits pushed too early
...
Revert "depend on fs before install"
This reverts commit 6c522179206df5288f5a0e89bbd033764faad67150c62cdfd7ea973b0b4c0b7863a9b7
2014-09-12 17:07:50 +00:00
6df5288f5a
removing commented work in progress on SDES and logging tunning on
...
rtcp_init
2014-09-12 11:58:54 -05:00
0e89bbd033
more formatting and logging tuning
2014-09-12 11:58:53 -05:00
764faad671
logging adjustment
2014-09-12 11:58:53 -05:00
50c62cdfd7
missing host to network conversion highest_sequence_number_received
2014-09-12 11:58:53 -05:00
ea973b0b4c
logging correction
2014-09-12 11:58:53 -05:00
0b7863a9b7
[FS-6623] implement RTCP report generation
2014-09-12 11:58:53 -05:00
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
80542e20f0
FS-5142 don't multipart/mixed if body has content-type present
2014-09-11 14:18:42 -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
Dave Kompel
François
Italo Rossi
Chris Rienzo
Brian West
Anthony Minessale
Hristo Trendev
Seven Du
Mike Jerris
Flavio Grossi
Travis Cross
Julien Chavanton
Nathan Neulinger
Ken Rice
Aaron Paolozzi
Joao Mesquita
Artur Kraev
olegstolyar
Matteo Brancaleoni
Peter Olsson
Thomas Kleffel
Jeff Lenk
Marc Olivier Chouinard
Anthony Minessale II
E. Schmidbauer
Markus von Arx
Spencer Thomason
Tamas Cseke
Dušan Dragić
Darren Schreiber
Giovanni Maruzzelli
Peter Wu
root