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
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* 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
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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
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* 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
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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
34bc98cafa
Merge pull request #47 in FS/freeswitch from ~FLAVIO/freeswitch-fs-5106:master to master
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* 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
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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
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
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
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
Jeff Lenk