asterisk/asterisk.git
2 years agoMerge "res_pjsip: New endpoint option "refer_blind_progress""
Jenkins2 [Thu, 1 Jun 2017 15:05:53 +0000 (10:05 -0500)]
Merge "res_pjsip: New endpoint option "refer_blind_progress""

2 years agoMerge "Sqlite3: make busy_timeout configurable."
Jenkins2 [Thu, 1 Jun 2017 14:31:24 +0000 (09:31 -0500)]
Merge "Sqlite3: make busy_timeout configurable."

2 years agoMerge "test_json: Fix test names with reserved words"
Jenkins2 [Wed, 31 May 2017 19:05:59 +0000 (14:05 -0500)]
Merge "test_json:  Fix test names with reserved words"

2 years agochannel / app_meetme: Fix parentheses.
Joshua Colp [Wed, 31 May 2017 09:25:02 +0000 (09:25 +0000)]
channel / app_meetme: Fix parentheses.

ASTERISK-27025

Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed

2 years agotest_json: Fix test names with reserved words
George Joseph [Tue, 30 May 2017 14:43:49 +0000 (08:43 -0600)]
test_json:  Fix test names with reserved words

Some of the test names were actually reserved words (true, false,
int, null, string, bool).  When the jenkins test results analyzer
does its thing it tries to create a map using the test names as
keys and fails because they're reserved words.

Added "type_" to those test names.

Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b

2 years agoMerge "format_mp3: Don't try to build format_mp3 if we don't have sources"
Jenkins2 [Tue, 30 May 2017 11:03:26 +0000 (06:03 -0500)]
Merge "format_mp3: Don't try to build format_mp3 if we don't have sources"

2 years agomanager: Clear the flag on the other channel.
Joshua Colp [Fri, 26 May 2017 16:41:59 +0000 (16:41 +0000)]
manager: Clear the flag on the other channel.

During the channel flag audit an incorrect change was
done. The flag should be cleared on the second channel.

ASTERISK-26469

Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8

2 years agoMerge "asterisk: Audit locking of channel when manipulating flags."
Jenkins2 [Fri, 26 May 2017 14:25:51 +0000 (09:25 -0500)]
Merge "asterisk: Audit locking of channel when manipulating flags."

2 years agoMerge "res_agi: Prevent crash when SET VARIABLE called without arguments"
George Joseph [Fri, 26 May 2017 12:12:16 +0000 (07:12 -0500)]
Merge "res_agi: Prevent crash when SET VARIABLE called without arguments"

2 years agoMerge "res_agi: Allow configuration of audio format of EAGI pipe"
George Joseph [Fri, 26 May 2017 00:01:57 +0000 (19:01 -0500)]
Merge "res_agi: Allow configuration of audio format of EAGI pipe"

2 years agoMerge "res_agi: Fix malformed AGI usage response"
Jenkins2 [Thu, 25 May 2017 20:23:18 +0000 (15:23 -0500)]
Merge "res_agi: Fix malformed AGI usage response"

2 years agoMerge "unittests: Add a unit test that causes a SEGV and..."
Jenkins2 [Thu, 25 May 2017 19:44:35 +0000 (14:44 -0500)]
Merge "unittests:  Add a unit test that causes a SEGV and..."

2 years agoformat_mp3: Don't try to build format_mp3 if we don't have sources
Sean Bright [Thu, 25 May 2017 16:10:00 +0000 (12:10 -0400)]
format_mp3: Don't try to build format_mp3 if we don't have sources

ASTERISK-23951 #close
Reported by: Tzafrir Cohen

Change-Id: Iebf181d44bb735787fde4b5be863c4d7e2478a30

2 years agoSqlite3: make busy_timeout configurable.
Martin Tomec [Tue, 23 May 2017 16:07:53 +0000 (18:07 +0200)]
Sqlite3: make busy_timeout configurable.

Enables runtime configuration of busy_timeout for sqlite databases.
Default timeout remains 1000ms.

ASTERISK-27014 #close

Change-Id: I8921a3aac3c335843be4cb17d2dd0a5c157a36da

2 years agoMerge "res_agi: Clarify 'RECORD FILE' documentation"
Jenkins2 [Wed, 24 May 2017 23:09:33 +0000 (18:09 -0500)]
Merge "res_agi: Clarify 'RECORD FILE' documentation"

2 years agounittests: Add a unit test that causes a SEGV and...
George Joseph [Wed, 24 May 2017 20:50:56 +0000 (14:50 -0600)]
unittests:  Add a unit test that causes a SEGV and...

...that can only be run by explicitly calling it with
'test execute category /DO_NOT_RUN/ name RAISE_SEGV'

This allows us to more easily test CI and debugging tools that
should do certain things when asterisk coredumps.

To allow this a new member was added to the ast_test_info
structure named 'explicit_only'.  If set by a test, the test
will be skipped during a 'test execute all' or
'test execute category ...'.

Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed

2 years agoMerge "app_queue: Add QUEUE_RAISE_PENALTY feature"
Joshua Colp [Wed, 24 May 2017 18:04:16 +0000 (13:04 -0500)]
Merge "app_queue: Add QUEUE_RAISE_PENALTY feature"

2 years agoMerge "chan_sip: Better ICE handling for RTCP-MUX"
Jenkins2 [Wed, 24 May 2017 16:41:02 +0000 (11:41 -0500)]
Merge "chan_sip: Better ICE handling for RTCP-MUX"

2 years agoMerge "res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm"
Jenkins2 [Wed, 24 May 2017 16:25:58 +0000 (11:25 -0500)]
Merge "res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm"

2 years agoMerge "res_format_attr_h26x: Trim blanks in fmtp attributes"
Jenkins2 [Wed, 24 May 2017 16:02:24 +0000 (11:02 -0500)]
Merge "res_format_attr_h26x: Trim blanks in fmtp attributes"

2 years agores_agi: Allow configuration of audio format of EAGI pipe
Sean Bright [Tue, 23 May 2017 20:42:04 +0000 (16:42 -0400)]
res_agi: Allow configuration of audio format of EAGI pipe

This change allows the format of the EAGI audio pipe to be changed by
setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of
the loaded formats.

ASTERISK-26124 #close

Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd

2 years agores_agi: Clarify 'RECORD FILE' documentation
Sean Bright [Tue, 23 May 2017 18:33:16 +0000 (14:33 -0400)]
res_agi: Clarify 'RECORD FILE' documentation

Documented the 'beep' option in both the parameters list and the command
description.

ASTERISK-23839 #close

Change-Id: I4970395c922dbdce3f7cf0f56d5b065ec9aa53ea

2 years agores_agi: Prevent crash when SET VARIABLE called without arguments
Sean Bright [Tue, 23 May 2017 18:06:22 +0000 (14:06 -0400)]
res_agi: Prevent crash when SET VARIABLE called without arguments

Explicitly check that the appropriate number of arguments were passed to
SET VARIABLE before attempting to reference them. Also initialize the
arguments array to zeroes before populating it.

ASTERISK-22432 #close

Change-Id: I5143607d80a2724f749c1674f3126b04ed32ea97

2 years agores_agi: Fix malformed AGI usage response
Sean Bright [Tue, 23 May 2017 17:35:25 +0000 (13:35 -0400)]
res_agi: Fix malformed AGI usage response

If the generated XML documentation for a command does not end with a \n,
the postamble of the usage message does not appear on its own line.

ASTERISK-25662 #close

Change-Id: If190f1e9e37fe215fed95897d78d4a6e142b0020

2 years agores_format_attr_h26x: Trim blanks in fmtp attributes
Sean Bright [Tue, 23 May 2017 15:06:02 +0000 (11:06 -0400)]
res_format_attr_h26x: Trim blanks in fmtp attributes

Some devices separate format attributes with a semicolon followed by a
space, so trim blanks before trying to match them.

ASTERISK-27008 #close

Change-Id: Ia44cb2e4fef5c73dc541a29da79cb0e19c22d9cc

2 years agoapp_queue: Fix members showing as being in call when not.
Joshua Colp [Mon, 15 May 2017 20:03:36 +0000 (20:03 +0000)]
app_queue: Fix members showing as being in call when not.

A change was done which added an 'in_call' flag to queue
members that was set to true while talking to an agent.
Unfortunately in practice this does not accurately reflect
whether they are talking to an agent or not. If a Local
channel is involved and a transfer is performed then the
app_queue application would incorrectly think the agent
was still in a call with the caller. This was done to
fix a race condition between an agent becoming available
by device state and the checking of the last call information
for the wrapup time. There was a small window where the
last call information would be the previous value instead
of the new one.

This change goes about fixing the original issue in a
different way by considering the call completed if device
state is received which would make the agent available
and if they are currently in a call. If this occurs the
last call information is updated before the agent becomes
available ensuring that old information is not present
when checking if the member should be called. This also
improves the transfer situation by actually updating
and enforcing the wrapup time.

ASTERISK-26399
ASTERISK-26400
ASTERISK-26715
ASTERISK-26975

Change-Id: Ife1cb686e3173b3a6d368601adef9aff69d4beea

2 years agoMerge "res_pjsip_session : fixed wrong From Header number On Re-invite"
Joshua Colp [Tue, 23 May 2017 14:17:13 +0000 (09:17 -0500)]
Merge "res_pjsip_session : fixed wrong From Header number On Re-invite"

2 years agores_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm
Kevin Harwell [Mon, 22 May 2017 18:51:40 +0000 (13:51 -0500)]
res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm

When using rtcp mux if an rtcp payload came in it would still use the srtp
unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp
data was being passed to the rtp unprotect method this would result in an
error.

This patch ensures that the correct unprotect method is chosen by making
sure the passed in rtcp flag is appropriately set when rtcp mux is enabled
and an rtcp payload is received.

ASTERISK-26979 #close

Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241

2 years agochan_sip: Better ICE handling for RTCP-MUX
Sean Bright [Fri, 19 May 2017 15:05:36 +0000 (15:05 +0000)]
chan_sip: Better ICE handling for RTCP-MUX

If we are offered or are offering RTCP-MUX, don't consider RTCP ICE
candidates. This confuses certain browsers (current Firefox for
example) and causes intial audio setup delays.

ASTERISK-26982 #close

Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91

2 years agoapp_queue: Add QUEUE_RAISE_PENALTY feature
Steve Davies [Fri, 12 May 2017 15:38:27 +0000 (16:38 +0100)]
app_queue: Add QUEUE_RAISE_PENALTY feature

Additional variable to work alongside QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY,
including an extra parameter in queuerules.conf. This value causes lower
Agent penalty values to "raise up" so that they can join higher penalty agents
and be treated equally after a period of time.

ASTERISK-26995 #close

Change-Id: If1c6421a983667a5ac4c359f6dac25b212b4c459

2 years agoMerge "app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON"
Joshua Colp [Mon, 22 May 2017 10:37:32 +0000 (05:37 -0500)]
Merge "app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON"

2 years agoMerge "app_stream_echo: Added a multi-stream echo application"
Joshua Colp [Mon, 22 May 2017 10:03:05 +0000 (05:03 -0500)]
Merge "app_stream_echo: Added a multi-stream echo application"

2 years agoMerge "core/conversions: Added string to unsigned integer and long conversions"
Jenkins2 [Mon, 22 May 2017 09:59:49 +0000 (04:59 -0500)]
Merge "core/conversions: Added string to unsigned integer and long conversions"

2 years agoMerge "res_hep_rtcp: Add support level to module info"
Jenkins2 [Fri, 19 May 2017 23:19:01 +0000 (18:19 -0500)]
Merge "res_hep_rtcp: Add support level to module info"

2 years agoMerge "AST-2017-004: chan_skinny: Add EOF check in skinny_session"
Jenkins2 [Fri, 19 May 2017 20:08:42 +0000 (15:08 -0500)]
Merge "AST-2017-004: chan_skinny:  Add EOF check in skinny_session"

2 years agoMerge "AST-2017-003: Handle zero-length body parts correctly."
Jenkins2 [Fri, 19 May 2017 19:41:50 +0000 (14:41 -0500)]
Merge "AST-2017-003: Handle zero-length body parts correctly."

2 years agoAST-2017-003: Handle zero-length body parts correctly.
Mark Michelson [Thu, 13 Apr 2017 22:17:36 +0000 (17:17 -0500)]
AST-2017-003: Handle zero-length body parts correctly.

ASTERISK-26939 #close

Change-Id: I7ea235ab39833a187db4e078f0788bd0af0a24fd

2 years agoAST-2017-004: chan_skinny: Add EOF check in skinny_session
George Joseph [Thu, 13 Apr 2017 16:14:48 +0000 (10:14 -0600)]
AST-2017-004: chan_skinny:  Add EOF check in skinny_session

The while(1) loop in skinny_session wasn't checking for EOF so
a packet that was longer than a header but still truncated
would spin the while loop infinitely.  Not only does this
permanently tie up a thread and drive a core to 100% utilization,
the call of ast_log() in such a tight loop eats all available
process memory.

Added poll with timeout to top of read loop

ASTERISK-26940 #close
Reported-by: Sandro Gauci

Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898

2 years agoAST-2017-002: Ensure transaction key buffer is large enough.
Mark Michelson [Thu, 13 Apr 2017 22:16:42 +0000 (17:16 -0500)]
AST-2017-002: Ensure transaction key buffer is large enough.

ASTERISK-26938 #close

Change-Id: I266490792fd8896a23be7cb92f316b7e69356413

2 years agores_hep_rtcp: Add support level to module info
Sean Bright [Thu, 18 May 2017 21:35:21 +0000 (17:35 -0400)]
res_hep_rtcp: Add support level to module info

Change-Id: I5661478f9cf12d431f730e42be79323b62831e92

2 years agoapp_stream_echo: Added a multi-stream echo application
Kevin Harwell [Mon, 15 May 2017 18:26:50 +0000 (13:26 -0500)]
app_stream_echo: Added a multi-stream echo application

If the channel does not have multi-stream support then this application acts
just like app_echo. If it does have multi-stream support then each stream is
echoed back to itself (one-to-one).

If a "num" is specified, then a new topology is made that contains clones (from
the channel's topology) of all media types that are not equal to the given
"type". If the media type differs then the first stream matching the "type" is
cloned into the new topology and then up to "num" - 1 of the same stream are
also cloned into it. Any additional streams from the original topology matching
the "type" are subsequently ignored (i.e. not added to the new topology).

For this same case when a frame is read from a stream that frame is still
echoed back like before, but now that frame is also echoed out to the
additional streams that matched on the specified "type".

ASTERISK-26997 #close

Change-Id: I254144486734178e196c7f590a26ffc13543ff2c

2 years agocore/conversions: Added string to unsigned integer and long conversions
Kevin Harwell [Mon, 15 May 2017 18:25:43 +0000 (13:25 -0500)]
core/conversions: Added string to unsigned integer and long conversions

Added functions that convert a string to an unsigned integer or unsigned long.
A couple of unit test were also created to test the routines. The reasons for
adding these conversion utilities (and hopefully eventually more) are as
follows:

  * Conversion routines are functionally contained with consistent and
    better error checking
  * The function names offer a better description of what is happening
  * It encourages code reuse for easier bug fixing at a single source
  * It's simpler to use
  * It's unit testable

For instance, currently in a lot of places when converting to an integer or
similar the "sscanf" function is used. When using "sscanf" it may not be
immediately clear what's happening as it lacks semantic naming. Limited error
checking is usually done as well. For example, most of the time a check is done
to make sure the value converted, but does not check for overflows or negative
valued conversions when converting unsigned numbers.

Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the
built in error handling that "strtoul" has. For instance "strtoul" contains
overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly
more complex in its use. And maybe a bit controversial, but it may be ("big if")
potentially slower than "strtoul" in some cases.

Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb

2 years agoMerge "res_pjsip_session.c: Process initial INVITE sooner. (key exists)"
Jenkins2 [Wed, 17 May 2017 16:40:28 +0000 (11:40 -0500)]
Merge "res_pjsip_session.c: Process initial INVITE sooner. (key exists)"

2 years agoMerge "Fix spelling queues.conf.sample file"
Joshua Colp [Wed, 17 May 2017 15:40:11 +0000 (10:40 -0500)]
Merge "Fix spelling queues.conf.sample file"

2 years agoasterisk: Audit locking of channel when manipulating flags.
Joshua Colp [Sat, 13 May 2017 16:40:00 +0000 (16:40 +0000)]
asterisk: Audit locking of channel when manipulating flags.

When manipulating flags on a channel the channel has to be
locked to guarantee that nothing else is also manipulating
the flags. This change introduces locking where necessary to
guarantee this. It also adds helper functions that manipulate
channel flags and lock to reduce repeated code.

ASTERISK-26789

Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10

2 years agores_pjsip_session.c: Process initial INVITE sooner. (key exists)
Richard Mudgett [Sat, 13 May 2017 02:04:59 +0000 (21:04 -0500)]
res_pjsip_session.c: Process initial INVITE sooner. (key exists)

Retransmissions of an initial INVITE could be queued in the serializer
before we have processed the first INVITE message.  If the first INVITE
message doesn't get completely processed before the retransmissions are
seen then we could try to setup the same call from the retransmissions.  A
symptom of this is seeing a (key exists) message associated with an
INVITE.  An earlier change attempted to address this kind of problem by
calculating a distributor serializer to use for unassociated messages.
Part of that change also made incoming calls keep using that distributor
serializer.  (ASTERISK-26088) However, some leftover code was still
deferring the INVITE processing to the session's serializer even though we
were already in that serializer.  This not only is unnecessary but would
cause the same call resetup problem.

* Removed the code to defer processing the initial INVITE to the session's
serializer because we are already running in that serializer.

ASTERISK-26998 #close

Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6

2 years agoMerge "chan_sip: Change sip_get_codec() to return correct codec list"
Joshua Colp [Mon, 15 May 2017 14:28:11 +0000 (09:28 -0500)]
Merge "chan_sip: Change sip_get_codec() to return correct codec list"

2 years agoFix spelling queues.conf.sample file
Rodrigo Ramírez Norambuena [Sun, 14 May 2017 05:37:09 +0000 (01:37 -0400)]
Fix spelling queues.conf.sample file

Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee

2 years agoMerge changes from topic 'sdp_api_adjustments'
George Joseph [Fri, 12 May 2017 17:29:39 +0000 (12:29 -0500)]
Merge changes from topic 'sdp_api_adjustments'

* changes:
  SDP: Make process possible multiple fmtp attributes per rtpmap.
  SDP: Explicitly stop a RTP instance before destoying it.
  SDP: Rework merge_capabilities().
  SDP: Update ast_get_topology_from_sdp() to keep RTP map.

2 years agoMerge "SDP: Remove sdp_state.remote_capabilities"
George Joseph [Fri, 12 May 2017 17:29:15 +0000 (12:29 -0500)]
Merge "SDP: Remove sdp_state.remote_capabilities"

2 years agoMerge "SDP: Add interface_address to specify our address to use."
Jenkins2 [Fri, 12 May 2017 16:49:58 +0000 (11:49 -0500)]
Merge "SDP: Add interface_address to specify our address to use."

2 years agochan_sip: Change sip_get_codec() to return correct codec list
Vitezslav Novy [Mon, 8 May 2017 18:40:47 +0000 (20:40 +0200)]
chan_sip: Change sip_get_codec() to return correct codec list

Return cahnnel nativeformats to fix bridge technology selection process.
Same approach as in pjsip module.

ASTERISK-26143
Reported-by: Henning Holtschneider

Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48

2 years agoMerge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages."
Jenkins2 [Thu, 11 May 2017 21:39:54 +0000 (16:39 -0500)]
Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages."

2 years agoMerge "logger: Added logger_queue_limit to the configuration options."
Jenkins2 [Thu, 11 May 2017 17:03:07 +0000 (12:03 -0500)]
Merge "logger:  Added logger_queue_limit to the configuration options."

2 years agores_pjsip: New endpoint option "refer_blind_progress"
Alexei Gradinari [Mon, 8 May 2017 20:56:32 +0000 (16:56 -0400)]
res_pjsip: New endpoint option "refer_blind_progress"

This option was added to turn off notifying the progress details
on Blind Transfer. If this option is not set then the chan_pjsip
will send NOTIFY "200 OK" immediately after "202 Accepted".

Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".

ASTERISK-26333 #close

Change-Id: Id606fbff2e02e967c02138457badc399144720f2

2 years agoMerge "tcptls: Improve error messages for TLS connections."
Jenkins2 [Thu, 11 May 2017 15:46:15 +0000 (10:46 -0500)]
Merge "tcptls: Improve error messages for TLS connections."

2 years agoMerge "Prevent Undefined Capath Crash"
Jenkins2 [Thu, 11 May 2017 15:38:38 +0000 (10:38 -0500)]
Merge "Prevent Undefined Capath Crash"

2 years agoapp_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON
Ivan Poddubny [Thu, 11 May 2017 05:25:44 +0000 (07:25 +0200)]
app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON

There are 2 places in app_queue.c that log EXITEMPTY event: one in
wait_our_turn, and another one in queue_exec in the loop trying to
call an agent after wait_our_turn.

In most cases it leads to logging EXITEMPTY twice.

ABANDON is also logged on two places, and in the rare case when an agent
and caller hang up simultaneously it's also possible to get duplicates
in queue_log.

This commit changes wait_our_turn to return -1 ("the caller should exit
the queue") instead of 0 ("the caller's turn has arrived") in case of
leaving when empty, so queue_exec skips the agent calling loop.

Also, leave_queue is now executed only once in this case, because 2nd
time is just a noop when the queue entry has already been removed.

Also, it sets qe->handled to -1 to indicate that the call was not
answered by an agent, but the necessary handling has already been done
in order to avoid logging an extra ABANDON entry.

ASTERISK-25665 #close
Reported by: Ove Aursand

Change-Id: I4578dd383bf2ac41589cf167865e8aaebcd4c11e

2 years agoSDP: Make process possible multiple fmtp attributes per rtpmap.
Richard Mudgett [Fri, 28 Apr 2017 00:37:53 +0000 (19:37 -0500)]
SDP: Make process possible multiple fmtp attributes per rtpmap.

Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349

2 years agoSDP: Remove sdp_state.remote_capabilities
Richard Mudgett [Fri, 28 Apr 2017 16:53:37 +0000 (11:53 -0500)]
SDP: Remove sdp_state.remote_capabilities

The sdp_state.remote_capabilities was only used inside merge_sdps() and
subsequent calls to merge_sdps() by re-INVITE's would leak them.

Change-Id: I0ceb7838ea044cc913e8ad4a255c39c9740ae0ce

2 years agoSDP: Add interface_address to specify our address to use.
Richard Mudgett [Fri, 5 May 2017 19:30:40 +0000 (14:30 -0500)]
SDP: Add interface_address to specify our address to use.

When we optionally set the interface_address we are forcing the media to
go out a specific interface address.  This allows us to optionally have
the media go out the interface that SIP signalling came in on or if we are
configured to have the media always go out a specific address.

Change-Id: I160d9fac322a075bd2557b430632544178196189

2 years agoSDP: Explicitly stop a RTP instance before destoying it.
Richard Mudgett [Fri, 5 May 2017 19:49:30 +0000 (14:49 -0500)]
SDP: Explicitly stop a RTP instance before destoying it.

* Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream()
handle generating disabled/declined streams.

* Added /main/sdp/sdp_merge_asymmetric unit test.  It currently does not
check the offerer side negotiated SDP because that isn't the purpose of
this patch and there is much to be done to handle declined/dummy streams.

* Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and
/main/sdp/sdp_merge_crisscross unit tests.

Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31

2 years agoSDP: Rework merge_capabilities().
Richard Mudgett [Sat, 29 Apr 2017 00:48:29 +0000 (19:48 -0500)]
SDP: Rework merge_capabilities().

* Tried to give better variable names.
* Made our SDP answer use the offer's RTP payload types as the SDP RFC
says we SHOULD.
* Updating the local topology now takes the stream format caps.  We are
likely preparing to send an offer.

Change-Id: I34d3be8e3036402a8575ffcae3eebc5ce348d7c0

2 years agoSDP: Update ast_get_topology_from_sdp() to keep RTP map.
Richard Mudgett [Fri, 28 Apr 2017 17:30:34 +0000 (12:30 -0500)]
SDP: Update ast_get_topology_from_sdp() to keep RTP map.

* Add failure exits to ast_get_topology_from_sdp().

Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049

2 years agotcptls: Improve error messages for TLS connections.
Joshua Colp [Tue, 9 May 2017 15:34:49 +0000 (15:34 +0000)]
tcptls: Improve error messages for TLS connections.

This change uses the functions provided by OpenSSL to query
and better construct error messages for situations where
the connection encounters a problem.

ASTERISK-26606

Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b

2 years agoPrevent Undefined Capath Crash
Joshua Elson [Thu, 4 May 2017 22:28:55 +0000 (18:28 -0400)]
Prevent Undefined Capath Crash

It is possible to initialize a valid config without a capath
or cafile definition. This will cause a crash on a reload.

This fix ensures capath is always allocated.

ASTERISK-26983 #close

Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12

2 years agocel_odbc: Fix timestamp processing for microseconds
George Joseph [Fri, 5 May 2017 16:33:34 +0000 (10:33 -0600)]
cel_odbc:  Fix timestamp processing for microseconds

When a column is of type timestamp, the fraction part of the event
field's seconds was frequently parsed incorrectly especially if
there were leading zeros.  For instance "2017-05-23 23:55:03.023"
would be parsed into an int as "23" then when the timestamp was
formatted again to be inserted into the database column it'd be
"2017-05-23 23:55:03.23" which is now 230 milliseconds instead of
23 milliseconds.  "03.000001" would be transformed to "03.1", etc.

* If the event field is 'eventtime' and the db column is timestamp,
  then existing processing has already correctly formatted the
  timestamp so now we simply use it rather than parsing it and
  re-printing it. This is the most common use case anyway.

* If the event field is other than 'eventtime' and the db column
  is timestamp, we now parse the seconds, including the fractional
  part into a double rather than 2 ints.  This preserves the
  magnitude and precision of the fractional part.  When we print
  it, we now print it as a "%09.6lf" which correctly represents the
  input.

To be honest, why we parse the string timestamp into components,
test the components, then print the components back into a string
timestamp is beyond me.  We should use parse it, test it, then if
it passes, use the original string representation in the database
call.  Maybe someone thought that some implementations wouldn't
take a partial timestamp string like "2017-05-06" and decided to
always produce a full timestamp string even if an abbreviated one
was supplied.  Anyway, I'm leaving it as it is.

ASTERISK-25032 #close
Reported-by: Etienne Lessard

Change-Id: Id407e6221f79a5c1120e1a70bc7e893bbcaf1938

2 years agores_hep_rtcp: Provide chan_sip Call-ID for RTCP messages.
Joshua Colp [Tue, 9 May 2017 10:25:29 +0000 (10:25 +0000)]
res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages.

This change adds the required logic to allow the SIP
Call-ID to be placed into the HEP RTCP traffic if the
chan_sip module is used. In cases where the option is
enabled but the channel is not either SIP or PJSIP then
the code will fallback to the channel name as done
previously.

Based on the change on Nir's branch at:
team/nirs/hep-chan-sip-support

ASTERISK-26427

Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d

2 years agoMerge "func_cdr: Allow empty value for CDR dialplan function."
Joshua Colp [Mon, 8 May 2017 23:35:13 +0000 (18:35 -0500)]
Merge "func_cdr: Allow empty value for CDR dialplan function."

2 years agoMerge "stream: ast_stream_clone() cannot copy the opaque user data."
Joshua Colp [Mon, 8 May 2017 22:25:22 +0000 (17:25 -0500)]
Merge "stream: ast_stream_clone() cannot copy the opaque user data."

2 years agologger: Added logger_queue_limit to the configuration options.
George Joseph [Mon, 8 May 2017 21:11:19 +0000 (15:11 -0600)]
logger:  Added logger_queue_limit to the configuration options.

All log messages go to a queue serviced by a single thread
which does all the IO.  This setting controls how big that
queue can get (and therefore how much memory is allocated)
before new messages are discarded. The default is 1000.
Should something go bezerk and log tons of messages in a tight
loop, this will prevent memory escalation.

When the limit is reached, a WARNING is logged to that effect
and messages are discarded until the queue is empty again.  At
that time another WARNING will be logged with the count of
discarded messages.  There's no "low water mark" for this queue
because the logger thread empties the entire queue and processes it
in 1 batch before going back and waiting on the queue again.
Implementing a low water mark would mean additional locking as
the thread processes each message and it's not worth it.

A "test" was added to test_logger.c but since the outcome is
non-deterministic, it's really just a cli command, not a unit
test.

Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1

2 years agoMerge "netsock2.c: Made get/set addr port avoid potential uninitialized memory."
Joshua Colp [Mon, 8 May 2017 13:44:22 +0000 (08:44 -0500)]
Merge "netsock2.c: Made get/set addr port avoid potential uninitialized memory."

2 years agoMerge "bridge: Fix returning to dialplan when executing Bridge() from AMI."
Joshua Colp [Mon, 8 May 2017 12:33:07 +0000 (07:33 -0500)]
Merge "bridge: Fix returning to dialplan when executing Bridge() from AMI."

2 years agostream: ast_stream_clone() cannot copy the opaque user data.
Richard Mudgett [Tue, 2 May 2017 23:05:01 +0000 (18:05 -0500)]
stream: ast_stream_clone() cannot copy the opaque user data.

ast_stream_clone() cannot copy the opaque user data stored on a stream.
We don't know how to clone the data so it isn't copied into the clone.

Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367

2 years agonetsock2.c: Made get/set addr port avoid potential uninitialized memory.
Richard Mudgett [Thu, 4 May 2017 22:32:03 +0000 (17:32 -0500)]
netsock2.c: Made get/set addr port avoid potential uninitialized memory.

Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647

2 years agofunc_cdr: Allow empty value for CDR dialplan function.
Joshua Colp [Fri, 5 May 2017 13:48:34 +0000 (13:48 +0000)]
func_cdr: Allow empty value for CDR dialplan function.

A regression was introduced in 12 where passing an empty value
to the CDR dialplan function was not longer allowed. This
change returns to the behavior of 11 where it is permitted.

ASTERISK-26173

Change-Id: I3f148203b54ec088007e29e30005a5de122e51c5

2 years agoapp_confbridge: Fix reference to cfg in menu_template_handler
George Joseph [Thu, 4 May 2017 21:04:46 +0000 (15:04 -0600)]
app_confbridge:  Fix reference to cfg in menu_template_handler

menu_template_handler wasn't properly accounting for the fact that
it might be called both during a load/reload (which isn't really
valid but not prevented) and by a dialplan function.  In both cases
it was attempting to use the "pending" config which wasn't valid in
the latter case.  aco_process_config is also partly to blame because
it wasn't properly cleaning "pending" up when a reload was done and
no changes were made.  Both of these contributed to a crash if
CONFBRIDGE(menu,template) was called in a dialplan after a reload.

* aco_process_config now sets info->internal->pending to NULL
  after it unrefs it although this isn't strictly necessary in the
  context of this fix.
* menu_template_handler now uses the "current" config and silently
  ignores any attempt to be called as a result of someone uses the
  "template" parameter in the conf file.

Luckily there's no other place in the codebase where
aco_pending_config is used outside of aco_process_config.

ASTERISK-25506 #close
Reported-by: Frederic LE FOLL

Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7

2 years agoMerge "SDP: Replace SDP telephone_event option with dtmf option"
Jenkins2 [Fri, 5 May 2017 00:17:06 +0000 (19:17 -0500)]
Merge "SDP: Replace SDP telephone_event option with dtmf option"

2 years agoMerge "res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures"
Jenkins2 [Thu, 4 May 2017 22:55:54 +0000 (17:55 -0500)]
Merge "res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures"

2 years agobridge: Fix returning to dialplan when executing Bridge() from AMI.
Joshua Colp [Sun, 30 Apr 2017 21:40:16 +0000 (21:40 +0000)]
bridge: Fix returning to dialplan when executing Bridge() from AMI.

When using the Bridge AMI action on the same channel multiple times
it was possible for the channel to return to the wrong location in
the dialplan if the other party hung up. This happened because the
priority of the channel was not preserved across each action
invocation and it would fail to move on to the next priority in
other cases.

This change makes it so that the priority of a channel is preserved
when taking control of it from another thread and it is incremented
as appropriate such that the priority reflects where the channel
should next be executed in the dialplan, not where it may or may not
currently be.

The Bridge AMI action was also changed to ensure that it too
starts the channels at the next location in the dialplan.

ASTERISK-24529

Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a

2 years agobridge_simple: Added support for streams
Kevin Harwell [Tue, 25 Apr 2017 16:49:16 +0000 (11:49 -0500)]
bridge_simple: Added support for streams

This patch is the first cut at adding stream support to the bridging framework.
Changes were made to the framework that allows mapping of stream topologies to
a bridge's supported media types.

The first channel to enter a bridge initially defines the media types for a
bridge (i.e. a one to one mapping is created between the bridge and the first
channel). Subsequently added channels merge their media types into the bridge's
adding to it when necessary. This allows channels with different sized
topologies to map correctly to each other according to media type. The bridge
drops any frame that does not have a matching index into a given write stream.

For now though, bridge_simple will align its two channels according to size or
first to join. Once both channels join the bridge the one with the most streams
will indicate to the other channel to update its streams to be the same as that
of the other. If both channels have the same number of streams then the first
channel to join is chosen as the stream base.

A topology change source was also added to a channel when a stream toplogy
change request is made. This allows subsystems to know whether or not they
initiated a change request. Thus avoiding potential recursive situations.

ASTERISK-26966 #close

Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163

2 years agores_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures
Kevin Harwell [Mon, 1 May 2017 18:04:16 +0000 (13:04 -0500)]
res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures

When a call gets put on hold RTP is temporarily stopped and Asterisk was
setting the remote RTCP address to NULL. Then when RTCP data was received
from the remote endpoint, Asterisk would be missing this information when
publishing the rtcp_message stasis event. Consequently, message subscribers
(in this case res_hep_rtcp) trying to parse the "from" field output the
following error:

"ast_sockaddr_split_hostport: Port missing in (null)"

This patch makes it so the remote RTCP address is no longer set to NULL when
stopping RTP. There was only one place that appeared to check if the remote
RTCP address was NULL as a way to tell if RTCP was running. This patch added
an additional check on the RTCP schedid for that case to make sure RTCP was
truly not running.

ASTERISK-26860 #close

Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b

2 years agoMerge "channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections"
Joshua Colp [Wed, 3 May 2017 16:05:35 +0000 (11:05 -0500)]
Merge "channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections"

2 years agocleanup: Change severity of fread short-read warning
Sean Bright [Tue, 2 May 2017 16:34:24 +0000 (12:34 -0400)]
cleanup: Change severity of fread short-read warning

Many sound files don't have a full frame's worth of data at EOF, so the
warning messages were a bit too noisy. So we demote them to debug
messages.

Change-Id: I6b617467d687658adca39170a81797a11cc766f6

2 years agoSDP: Replace SDP telephone_event option with dtmf option
Richard Mudgett [Wed, 26 Apr 2017 21:22:38 +0000 (16:22 -0500)]
SDP: Replace SDP telephone_event option with dtmf option

The telephone_event option was used as a flag and a bit mapped value in
different places when it is a boolean.  It is also inadequate to configure
the DTMF operation of the RTP instance created for the stream.

Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b

2 years agoMerge "res_pjsip_t38.c: Fix deadlock in T.38 framehook."
Jenkins2 [Tue, 2 May 2017 14:22:24 +0000 (09:22 -0500)]
Merge "res_pjsip_t38.c: Fix deadlock in T.38 framehook."

2 years agoMerge "res_sdp_translator_pjmedia.c: Add TODO notes."
Joshua Colp [Tue, 2 May 2017 10:20:03 +0000 (05:20 -0500)]
Merge "res_sdp_translator_pjmedia.c: Add TODO notes."

2 years agoMerge "SDP: Make SDP translation to/from internal representation more const."
Joshua Colp [Tue, 2 May 2017 10:19:59 +0000 (05:19 -0500)]
Merge "SDP: Make SDP translation to/from internal representation more const."

2 years agoMerge "stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap."
Joshua Colp [Tue, 2 May 2017 10:19:12 +0000 (05:19 -0500)]
Merge "stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap."

2 years agoMerge "SDP: Make ast_sdp_state_set_remote_sdp() return error."
Jenkins2 [Mon, 1 May 2017 22:01:20 +0000 (17:01 -0500)]
Merge "SDP: Make ast_sdp_state_set_remote_sdp() return error."

2 years agoMerge "res_pjsip_outbound_authenticator_digest: Add context to log messages"
Jenkins2 [Mon, 1 May 2017 20:08:21 +0000 (15:08 -0500)]
Merge "res_pjsip_outbound_authenticator_digest: Add context to log messages"

2 years agoMerge "SDP: Misc cleanups (Mostly memory leaks)"
Jenkins2 [Mon, 1 May 2017 19:19:34 +0000 (14:19 -0500)]
Merge "SDP: Misc cleanups (Mostly memory leaks)"

2 years agoMerge "SDP API: Add SSRC-level attributes"
Jenkins2 [Mon, 1 May 2017 19:16:55 +0000 (14:16 -0500)]
Merge "SDP API: Add SSRC-level attributes"

2 years agores_pjsip_t38.c: Fix deadlock in T.38 framehook.
Richard Mudgett [Sat, 29 Apr 2017 21:11:21 +0000 (16:11 -0500)]
res_pjsip_t38.c: Fix deadlock in T.38 framehook.

A deadlock can happen between a channel lock and a pjsip session media
container lock.  One thread is processing a reINVITE's SDP and walking
through the session's media container when it waits for the channel lock
to put the determined format capabilities onto the channel.  The other
thread is writing a frame to the channel and processing the T.38 frame
hook.  The T.38 frame hook then waits for the pjsip session's media
container lock.  The two threads are now deadlocked.

* Made the T.38 frame hook release the channel lock before searching the
session's media container.  This fix has been done to several other
frame hooks to fix deadlocks.

ASTERISK-26974 #close

Change-Id: Ie984a76ce00bef6ec9aa239010e51e8dd74c8186

2 years agores_pjsip_outbound_authenticator_digest: Add context to log messages
George Joseph [Fri, 28 Apr 2017 15:56:20 +0000 (09:56 -0600)]
res_pjsip_outbound_authenticator_digest: Add context to log messages

There was no context info in this module's log messages so it was
impossible to toubleshoot.

Added endpoint or host to all messages and added the realms in the
challenge for the "No auth credentials for any realm" message.

Change-Id: Ifeed2786f35fbea7d141237ae15625e472acff9b

2 years agoMerge "chan_vpb.cc: Fix compile error."
Jenkins2 [Fri, 28 Apr 2017 15:38:22 +0000 (10:38 -0500)]
Merge "chan_vpb.cc: Fix compile error."

2 years agoMerge "chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK"
George Joseph [Fri, 28 Apr 2017 00:17:09 +0000 (19:17 -0500)]
Merge "chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK"

2 years agores_sdp_translator_pjmedia.c: Add TODO notes.
Richard Mudgett [Thu, 27 Apr 2017 21:46:40 +0000 (16:46 -0500)]
res_sdp_translator_pjmedia.c: Add TODO notes.

Change-Id: If27ca61f79accc882c3376d2e876d2b44aa1347b

2 years agoSDP: Make SDP translation to/from internal representation more const.
Richard Mudgett [Mon, 24 Apr 2017 23:13:04 +0000 (18:13 -0500)]
SDP: Make SDP translation to/from internal representation more const.

Change-Id: I473a174b869728604b37c60853896b0c458bc504

2 years agostream: Make ast_stream_topology_create_from_format_cap() allow NULL cap.
Richard Mudgett [Fri, 21 Apr 2017 00:25:10 +0000 (19:25 -0500)]
stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap.

Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08