asterisk/asterisk.git
23 months agoMerge "pjsip_message_filter: Fix regression causing bad contact address"
Jenkins2 [Thu, 28 Sep 2017 18:36:28 +0000 (13:36 -0500)]
Merge "pjsip_message_filter: Fix regression causing bad contact address"

23 months agoMerge "res_stasis: Add 'video_sfu' as a requested bridge type."
Joshua Colp [Thu, 28 Sep 2017 18:13:31 +0000 (13:13 -0500)]
Merge "res_stasis: Add 'video_sfu' as a requested bridge type."

23 months agoMerge "res_pjsip_session: outgoing call did not offer all configured codecs"
Joshua Colp [Thu, 28 Sep 2017 17:24:11 +0000 (12:24 -0500)]
Merge "res_pjsip_session: outgoing call did not offer all configured codecs"

23 months agoMerge "pjproject: Patch to correct STUN FINGERPRINT usage"
Jenkins2 [Thu, 28 Sep 2017 13:30:54 +0000 (08:30 -0500)]
Merge "pjproject: Patch to correct STUN FINGERPRINT usage"

23 months agoMerge "res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential."
Joshua Colp [Thu, 28 Sep 2017 12:08:33 +0000 (07:08 -0500)]
Merge "res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential."

23 months agoMerge "res_rtp_asterisk: Trim trailing byte off of SDES packet"
Joshua Colp [Thu, 28 Sep 2017 11:45:28 +0000 (06:45 -0500)]
Merge "res_rtp_asterisk: Trim trailing byte off of SDES packet"

23 months agores_stasis: Add 'video_sfu' as a requested bridge type.
Joshua Colp [Thu, 28 Sep 2017 10:33:00 +0000 (10:33 +0000)]
res_stasis: Add 'video_sfu' as a requested bridge type.

This change adds 'video_sfu' as a requested bridge type when
creating a bridge. By specifying this a mixing type bridge is
created that exchanges video in an SFU fashion.

Change-Id: I2ada47cf5f3fc176518b647c0b4aa39d55339606

23 months agores_pjsip_outbound_publish.c: Fix misplaced parenthesis.
Richard Mudgett [Wed, 27 Sep 2017 16:16:16 +0000 (11:16 -0500)]
res_pjsip_outbound_publish.c: Fix misplaced parenthesis.

The pjsip_publishc_init() call was referenced with a misplaced
parentheses.  As a result, outbound publication messages went out with an
expiration of 1 second.

ASTERISK-27298

Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08

23 months agopjsip_message_filter: Fix regression causing bad contact address
George Joseph [Tue, 26 Sep 2017 16:01:48 +0000 (10:01 -0600)]
pjsip_message_filter: Fix regression causing bad contact address

The "res_pjsip:  Filter out non SIP(S) requests" commit moved the
filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER
in order to filter out incoming bad uri schemes as early as possible.
Since the change affected outgoing messages as well and the TRANSPORT
layer is the last to be run on outgoing messages, we were overwriting
the setting of external_signaling_address (which is set earlier by
res_pjsip_nat) with an internal address.

* pjsip_message_filter now registers itself as a pjproject module
twice.  Once in the TSX layer for the outgoing messages (as it was
originally), then a second time in the TRANSPORT layer for the
incoming messages to catch the invalid uri schemes.

ASTERISK-27295
Reported by: Sean Bright

Change-Id: I2c90190c43370f8a9d1c4693a19fd65840689c8c

23 months agores_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential.
Richard Mudgett [Thu, 14 Sep 2017 02:31:52 +0000 (21:31 -0500)]
res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential.

The bridge_p2p_rtp_write() has potential reentrancy problems.

* Accessing the bridged RTP members must be done with the instance1 lock
held.  The DTMF and asymmetric codec checks must be split to be done with
the correct RTP instance struct locked.  i.e., They must be done when
working on the appropriate side of the point to point bridge.

* Forcing the RTP mark bit was referencing the wrong side of the point to
point bridge.  The set mark bit is used everywhere else to set the mark
bit when sending not receiving.

The patches for ASTERISK_26745 and ASTERISK_27158 did not take into
account that not everything carried by RTP uses a codec.  The telephony
DTMF events are not exchanged with a codec.  As a result when
RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is
enabled, the DTMF digits would always get passed to the core even though
the local native RTP bridge is active, and the DTMF digits would go out
using the wrong SSRC id.

* Add protection for non-format payload types like DTMF when updating the
lastrxformat and lasttxformat.  Also protect against non-format payload
types when checking for asymmetric codecs.

ASTERISK-27292

Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186

23 months agores_rtp_asterisk: Trim trailing byte off of SDES packet
Sean Bright [Tue, 26 Sep 2017 15:55:29 +0000 (11:55 -0400)]
res_rtp_asterisk: Trim trailing byte off of SDES packet

This could have been fixed by subtracting 1 from the final value of
'len' but the way the packet was being constructed was confusing so I
took the opportunity to (I think) make it more clear.

We were sending 1 extra byte at the end of the SDES RTCP packet which
caused Chrome to complain (in its debug log):

    Too little data (1 byte) remaining in buffer to parse
    RTCP header (4 bytes).

We now send the correct number of bytes.

Change-Id: I9dcf087cdaf97da0374ae0acb7d379746a71e81b

23 months agoMerge "webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file"
Joshua Colp [Tue, 26 Sep 2017 12:37:22 +0000 (07:37 -0500)]
Merge "webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file"

23 months agoMerge "channel.c: Fix invalid reference in conditionaled out code."
Joshua Colp [Tue, 26 Sep 2017 12:29:02 +0000 (07:29 -0500)]
Merge "channel.c: Fix invalid reference in conditionaled out code."

23 months agoMerge "app_queue: Only do announcement logic between ringing cycles"
Joshua Colp [Tue, 26 Sep 2017 11:36:47 +0000 (06:36 -0500)]
Merge "app_queue: Only do announcement logic between ringing cycles"

23 months agoMerge "res_pjsip_session: Reduce (and improve) SDP renegotiation."
Joshua Colp [Mon, 25 Sep 2017 20:35:11 +0000 (15:35 -0500)]
Merge "res_pjsip_session: Reduce (and improve) SDP renegotiation."

23 months agowebrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file
Sean Bright [Mon, 25 Sep 2017 18:00:53 +0000 (14:00 -0400)]
webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file

If using a legitimate certificate from a trusted certificate authority,
you don't need to provide CA file.

Change-Id: I8623973b4209b44889243716d7880274caed8a6d

23 months agopjproject: Patch to correct STUN FINGERPRINT usage
Sean Bright [Mon, 25 Sep 2017 18:09:33 +0000 (14:09 -0400)]
pjproject: Patch to correct STUN FINGERPRINT usage

Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6

23 months agoMerge "build: A few gcc 7 error fixes"
Joshua Colp [Mon, 25 Sep 2017 18:04:22 +0000 (13:04 -0500)]
Merge "build:  A few gcc 7 error fixes"

23 months agores_pjsip_session: outgoing call did not offer all configured codecs
Kevin Harwell [Mon, 25 Sep 2017 17:30:56 +0000 (12:30 -0500)]
res_pjsip_session: outgoing call did not offer all configured codecs

For some scenarios when an outgoing call was made only a subset of the
configured codecs were offered. If the codecs being offered happened to
not have a codec supported by the phone then the call would fail.

For instance Alice and Bob both are configured in Asterisk for g722 and ulaw(
allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's
only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with
Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains
g722 and not both g722 and ulaw, so the call ends.

This patch makes it so all the audio codecs configured on the endpoint always
get sent, and not just a subset. However priority is given to those codecs that
are compatible with the "other side".

ASTERISK-27259 #close

Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696

23 months agoMerge "res_pjsip_session: Don't end session when receiving a 500 on a reinvite"
Joshua Colp [Mon, 25 Sep 2017 17:23:53 +0000 (12:23 -0500)]
Merge "res_pjsip_session: Don't end session when receiving a 500 on a reinvite"

23 months agochannel.c: Fix invalid reference in conditionaled out code.
Richard Mudgett [Mon, 25 Sep 2017 15:59:17 +0000 (10:59 -0500)]
channel.c: Fix invalid reference in conditionaled out code.

ASTERISK-27289

Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c

23 months agoMerge "app_stream_echo: Don't echo declined streams"
Joshua Colp [Mon, 25 Sep 2017 13:00:58 +0000 (08:00 -0500)]
Merge "app_stream_echo: Don't echo declined streams"

23 months agobuild: A few gcc 7 error fixes
George Joseph [Mon, 25 Sep 2017 12:25:06 +0000 (06:25 -0600)]
build:  A few gcc 7 error fixes

Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec

23 months agoMerge "res_pjsip: Use ast_sip_is_content_type() where appropriate"
Joshua Colp [Mon, 25 Sep 2017 12:28:41 +0000 (07:28 -0500)]
Merge "res_pjsip: Use ast_sip_is_content_type() where appropriate"

23 months agoapp_queue: Only do announcement logic between ringing cycles
StefanEng86 [Fri, 15 Sep 2017 07:59:59 +0000 (09:59 +0200)]
app_queue: Only do announcement logic between ringing cycles

This patch reverts the change by patch 2263 from old reviewboard.
Note that reverting that 2263-patch still preserves the behaviour that
the commit log of the 2263-patch claimed to add. The reason for this is:

The function wait_for_answer is only called from try_calling which
in turn is only called from the main for loop in queue_exec, and
earlier in that loop we already check the things that's removed by
this patch. There's no need to check those things twice each loop
iteration, and I think the proper place to check it is before each
ringing cycle. By checking it in wait_for_answer, you allow the issue
explained in the jira - that the head caller hears announcements while
the agents' sip phones are actively ringing.

Reported-by: Stefan Engström
Tested-by: Stefan Engström
ASTERISK-27216 #close

Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0

23 months agoapp_stream_echo: Don't echo declined streams
Sean Bright [Sat, 23 Sep 2017 17:32:26 +0000 (13:32 -0400)]
app_stream_echo: Don't echo declined streams

Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic
Edition after accepting the audio request but declining the video one.

Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c

23 months agores_pjsip_session: Reduce (and improve) SDP renegotiation.
Joshua Colp [Fri, 22 Sep 2017 22:49:21 +0000 (22:49 +0000)]
res_pjsip_session: Reduce (and improve) SDP renegotiation.

When pruning a request to change the topology of a channel be
more intelligent about the resulting topology that is actually
used for SDP renegotiation.

In a case where a stream has not already been negotiated we
don't need to renegotiate and offer a declined stream. This can
occur if something in Asterisk (such as ConfBridge) requests
to add video to a PJSIP channel that has no video codecs configured.
In this case since the stream did not already exist we can safely
remove the stream from the requested topology, resulting in no
renegotiation occurring.

In a case where a renegotiation is requested with a codec that is
not supported we can reuse the formats of the existing stream if
it exists to ensure that the stream continues to flow, instead of
removing it.

Change-Id: I636540798d55922377318fe619c510fb6ed125fb

23 months agores_pjsip_session: Don't end session when receiving a 500 on a reinvite
Kevin Harwell [Fri, 22 Sep 2017 20:29:24 +0000 (15:29 -0500)]
res_pjsip_session: Don't end session when receiving a 500 on a reinvite

During a reinvite, if a remote endpoint error occurs and it returns a 500 the
session would end. This patch makes it so the session is not terminated, but
continues as it was.

The reason for this is because some endpoints may send non session terminating
"server errors" like a failed codec negotiation. So in this case instead of
ending the call it can hopefully continue. In the case of a real server error
the session is already "doomed", will be known soon enough and appropriately
ended by Asterisk later.

Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5

23 months agoMerge "res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint"
Joshua Colp [Fri, 22 Sep 2017 20:35:29 +0000 (15:35 -0500)]
Merge "res_pjsip_session/BUNDLE:  Handle no audio codecs on endpoint"

23 months agoMerge "res_pjsip_session: Change some asserts to warning/debug messages"
Joshua Colp [Fri, 22 Sep 2017 16:10:11 +0000 (11:10 -0500)]
Merge "res_pjsip_session:  Change some asserts to warning/debug messages"

23 months agores_pjsip: Use ast_sip_is_content_type() where appropriate
Sean Bright [Fri, 22 Sep 2017 15:02:11 +0000 (11:02 -0400)]
res_pjsip: Use ast_sip_is_content_type() where appropriate

Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d

23 months agores_pjsip_session/BUNDLE: Handle no audio codecs on endpoint
George Joseph [Thu, 21 Sep 2017 14:47:11 +0000 (08:47 -0600)]
res_pjsip_session/BUNDLE:  Handle no audio codecs on endpoint

When an INVITE came in with both audio and video streams but there
were no audio codecs defined for the endpoint, we weren't declining
the audio stream.  Since it's usually the first/transport stream,
when the video stream was processed and tried to use the transport,
it was empty and caused a crash.  We now decline the the stream if
there are no matching codecs so when the video stream is processed,
it's now the first/transport stream and processes normally.

Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692

23 months agoMerge "res_rtp_asterisk.c: Fix bundled SSRC handling."
Joshua Colp [Fri, 22 Sep 2017 11:42:30 +0000 (06:42 -0500)]
Merge "res_rtp_asterisk.c: Fix bundled SSRC handling."

23 months agoMerge "res_config_pgsql: Fix removed support to previous for versions PostgreSQL...
Joshua Colp [Fri, 22 Sep 2017 10:42:43 +0000 (05:42 -0500)]
Merge "res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1"

23 months agoMerge "bridge: Change participant SFU streams when source streams change."
Joshua Colp [Thu, 21 Sep 2017 20:56:47 +0000 (15:56 -0500)]
Merge "bridge: Change participant SFU streams when source streams change."

23 months agores_rtp_asterisk.c: Fix bundled SSRC handling.
Richard Mudgett [Tue, 19 Sep 2017 19:28:37 +0000 (14:28 -0500)]
res_rtp_asterisk.c: Fix bundled SSRC handling.

Assertions in the v15+ AST-2017-008 patches found that we were not
handling the case if the incoming SDP did not specify the required SSRC
attributes for bundled to work.

* Be strict on matching SSRC for bundled instances including the parent
instance.  If the SSRC doesn't match then discard the packet.  Bundled has
to tell us in the SDP signaling what SSRC to expect.  Otherwise, we will
not know how to find the bundled instance structure.

Change-Id: I152830bbff71c662408909042068fada39e617f9

23 months agoMerge "AST-2017-008: Improve RTP and RTCP packet processing."
Joshua Colp [Thu, 21 Sep 2017 19:48:17 +0000 (14:48 -0500)]
Merge "AST-2017-008: Improve RTP and RTCP packet processing."

23 months agoMerge "res_config_pgsql: Add missing \n in debug log and update copyright year"
Jenkins2 [Thu, 21 Sep 2017 18:06:18 +0000 (13:06 -0500)]
Merge "res_config_pgsql: Add missing \n in debug log and update copyright year"

23 months agoMerge "res_pjsip_session: Check for removed stream state."
Joshua Colp [Thu, 21 Sep 2017 17:30:18 +0000 (12:30 -0500)]
Merge "res_pjsip_session: Check for removed stream state."

23 months agobridge: Change participant SFU streams when source streams change.
Joshua Colp [Sat, 16 Sep 2017 14:19:59 +0000 (11:19 -0300)]
bridge: Change participant SFU streams when source streams change.

Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.

This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.

ASTERISK-27277

Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07

23 months agoMerge "res_srtp: lower log level of auth failures"
Jenkins2 [Thu, 21 Sep 2017 17:07:57 +0000 (12:07 -0500)]
Merge "res_srtp: lower log level of auth failures"

23 months agoMerge "chan_sip: Expose read-only access to the full SIP INVITE Request-URI"
Jenkins2 [Thu, 21 Sep 2017 16:11:15 +0000 (11:11 -0500)]
Merge "chan_sip: Expose read-only access to the full SIP INVITE Request-URI"

23 months agoMerge "bridge : Fix one-way direct-media when early bridging with native_rtp"
Joshua Colp [Thu, 21 Sep 2017 15:57:56 +0000 (10:57 -0500)]
Merge "bridge : Fix one-way direct-media when early bridging with native_rtp"

23 months agores_pjsip_session: Change some asserts to warning/debug messages
George Joseph [Wed, 20 Sep 2017 15:45:16 +0000 (09:45 -0600)]
res_pjsip_session:  Change some asserts to warning/debug messages

There was an issue reported where an SDP received on a 183 Session
Progress message caused a crash because the pending streams had
already been processed when the OK was received.  In that case the
pending topology was legitimately NULL.  There was an assert for an
incorrect number of streams in the topology but not one for
topology being NULL.  In any case, if you're not in dev-mode the
asserts don't do anything and since the scenario is legit, the
asserts weren't appropriate anyway.

* Changed several asserts to warning or debug messages and return
codes as appropriate.

ASTERISK-27264
Reported by: Daniel Heckl

Change-Id: I58daaa9d2938fa980857ab3ec41925ab5ff9c848

23 months agores_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1
Rodrigo Ramírez Norambuena [Tue, 19 Sep 2017 10:22:50 +0000 (07:22 -0300)]
res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1

In PostgreSQL 9.1 the backslash are string literals and not the escape
of characters.

In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the
support for old version of Postgresql than 9.1 was dropped. The sentence
before make was "ESCAPE '\'" but in version before than 9.1  need it to be
as follow "ESCAPE '\\'".

ASTERISK-27283

Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949

23 months agores_pjsip_session: Check for removed stream state.
Ben Ford [Fri, 15 Sep 2017 14:43:21 +0000 (09:43 -0500)]
res_pjsip_session: Check for removed stream state.

When a sip session is refreshed, the stream topology is looped
through, checking each stream for compatible formats. This would
cause a crash if the stream state was AST_STREAM_STATE_REMOVED,
since the formats would never be set for this stream, causing
a NULL value to be returned from ast_stream_get_formats. This
commit adds a check for streams with removed states.

Also removed a stray semicolon.

Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42

23 months agochan_pjsip: Ignore AST_CONTROL_STREAM_TOPOLOGY_CHANGED for now
George Joseph [Tue, 19 Sep 2017 10:44:28 +0000 (04:44 -0600)]
chan_pjsip: Ignore AST_CONTROL_STREAM_TOPOLOGY_CHANGED for now

chan_pjsip_indicate was missing a case for the recently added
AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an
error and causing the call to be hung up instead of just ignoring
it.

ASTERISK-27260
Reported by: Daniel Heckl

Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80

23 months agobridge : Fix one-way direct-media when early bridging with native_rtp
Jean Aunis [Thu, 7 Sep 2017 09:41:09 +0000 (11:41 +0200)]
bridge : Fix one-way direct-media when early bridging with native_rtp

When two channels were early bridged in a native_rtp bridge, the RTP description
on one side was not updated when the other side answered.
This patch forbids non-answered channels to enter a native_rtp bridge, and
triggers a bridge reconfiguration when an ANSWER frame is received.

ASTERISK-27257

Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df

23 months agoMerge "res_calendar: Plug memory leak and micro-optimization"
Joshua Colp [Wed, 20 Sep 2017 14:44:53 +0000 (09:44 -0500)]
Merge "res_calendar: Plug memory leak and micro-optimization"

23 months agoMerge "res_pjsip_pubsub: Check for Content-Type header in rx_notify_request"
Jenkins2 [Wed, 20 Sep 2017 14:14:12 +0000 (09:14 -0500)]
Merge "res_pjsip_pubsub:  Check for Content-Type header in rx_notify_request"

23 months agores_srtp: lower log level of auth failures
Alexander Traud [Mon, 18 Sep 2017 14:51:15 +0000 (16:51 +0200)]
res_srtp: lower log level of auth failures

Previously, sRTP authentication failures were reported on log level WARNING.
When such failures happen, each RT(C)P packet is affected, spamming the log.
Now, those failures are reported at log level VERBOSE 2. Furthermore, the
amount is further reduced (previously all two seconds, now all three seconds).
Additionally, the new log entry informs whether media (RTP) or statistics (RTCP)
are affected.

ASTERISK-16898 #close

Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0

23 months agores_pjsip_pubsub: Check for Content-Type header in rx_notify_request
George Joseph [Tue, 19 Sep 2017 15:38:30 +0000 (09:38 -0600)]
res_pjsip_pubsub:  Check for Content-Type header in rx_notify_request

pubsub_on_rx_notify_request wasn't checking for a null
Content-Type header before checking that it was
application/simple-message-summary.

ASTERISK-27279
Reported by: Ross Beer

Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52

23 months agochan_sip: Expose read-only access to the full SIP INVITE Request-URI
David J. Pryke [Tue, 19 Sep 2017 14:34:01 +0000 (10:34 -0400)]
chan_sip: Expose read-only access to the full SIP INVITE Request-URI

Provide a way to get the contents of the the Request URI from the initial SIP
INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}")

ASTERISK-27278
Reported by: David J. Pryke
Tested by: David J. Pryke

Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e

23 months agoapp_confbridge: Only create a channel that records audio.
Joshua Colp [Tue, 19 Sep 2017 12:53:41 +0000 (12:53 +0000)]
app_confbridge: Only create a channel that records audio.

This change makes it so that the conference recorder channel
that is created only contains audio formats and an audio stream.
This is because the underlying application used by ConfBridge to
record, MixMonitor, only allows recording audio.

Having additional streams (and in particular a video stream) can
result in clients needlessly renegotiating to add a video stream
that will never receive video.

Change-Id: I89d38aedc9205eca7741d5435e73e73bb9de97a0

23 months agores_config_pgsql: Add missing \n in debug log and update copyright year
Rodrigo Ramírez Norambuena [Tue, 19 Sep 2017 11:34:28 +0000 (08:34 -0300)]
res_config_pgsql: Add missing \n in debug log and update copyright year

Change-Id: I4ba338ecbdecc6a814a902eddc4121c8ef3cda58

23 months agores_calendar: Plug memory leak and micro-optimization
Sean Bright [Wed, 13 Sep 2017 19:14:25 +0000 (15:14 -0400)]
res_calendar: Plug memory leak and micro-optimization

ast_variables_destroy is NULL safe, so there is no need to check its
argument before passing it.

ASTERISK-25524 #close
Reported by: Jesper

Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b

23 months agocdr_mysql.c: Apply cdrzone to start and answer
alex [Wed, 13 Sep 2017 08:46:27 +0000 (11:46 +0300)]
cdr_mysql.c: Apply cdrzone to start and answer

Change-Id: I7de0a5adc89824a5f2b696fc22c80fc22dff36b0

23 months agoAST-2017-008: Improve RTP and RTCP packet processing.
Richard Mudgett [Fri, 25 Aug 2017 22:01:57 +0000 (17:01 -0500)]
AST-2017-008: Improve RTP and RTCP packet processing.

Validate RTCP packets before processing them.

* Validate that the received packet is of a minimum length and apply the
RFC3550 RTCP packet validation checks.

* Fixed potentially reading garbage beyond the received RTCP record data.

* Fixed rtp->themssrc only being set once when the remote could change
the SSRC.  We would effectively stop handling the RTCP statistic records.

* Fixed rtp->themssrc to not treat a zero value as special by adding
rtp->themssrc_valid to indicate if rtp->themssrc is available.

ASTERISK-27274

Make strict RTP learning more flexible.

Direct media can cause strict RTP to attempt to learn a remote address
again before it has had a chance to learn the remote address the first
time.  Because of the rapid relearn requests, strict RTP could latch onto
the first remote address and fail to latch onto the direct media remote
address.  As a result, you have one way audio until the call is placed on
and off hold.

The new algorithm learns remote addresses for a set time (1.5 seconds)
before locking the remote address.  In addition, we must see a configured
number of remote packets from the same address in a row before switching.

* Fixed strict RTP learning from always accepting the first new address
packet as the new stream.

* Fixed strict RTP to initialize the expected sequence number with the
last received sequence number instead of the last transmitted sequence
number.

* Fixed the predicted next sequence number calculation in
rtp_learning_rtp_seq_update() to handle overflow.

ASTERISK-27252

Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c

23 months agoMerge "res_pjsip: Filter out non SIP(S) requests"
Jenkins2 [Fri, 15 Sep 2017 20:37:04 +0000 (15:37 -0500)]
Merge "res_pjsip:  Filter out non SIP(S) requests"

23 months agoMerge "res_calendar: Various fixes"
Joshua Colp [Fri, 15 Sep 2017 13:20:45 +0000 (08:20 -0500)]
Merge "res_calendar: Various fixes"

2 years agores_pjsip: Filter out non SIP(S) requests
George Joseph [Wed, 13 Sep 2017 21:23:54 +0000 (15:23 -0600)]
res_pjsip:  Filter out non SIP(S) requests

Incoming requests with non sip(s) URIs in the Request, To, From
or Contact URIs are now rejected with
PJSIP_SC_UNSUPPORTED_URI_SCHEME (416).  This is performed in
pjsip_message_filter (formerly pjsip_message_ip_updater) and is
done at pjproject's "TRANSPORT" layer before a request can even
reach the distributor.

URIs read by res_pjsip_outbound_publish from pjsip.conf are now
also checked for both length and sip(s) scheme.  Those URIs read
by outbound registration and aor were already being checked for
scheme but their error messages needed to be updated to include
scheme failure as well as length failure.

Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460

2 years agoMerge "chan_rtp: Use μ-law by default instead of signed linear"
Jenkins2 [Thu, 14 Sep 2017 17:37:43 +0000 (12:37 -0500)]
Merge "chan_rtp: Use μ-law by default instead of signed linear"

2 years agoMerge "tcptls: Change error message to debug."
Joshua Colp [Thu, 14 Sep 2017 17:11:38 +0000 (12:11 -0500)]
Merge "tcptls: Change error message to debug."

2 years agotcptls: Change error message to debug.
Joshua Colp [Thu, 14 Sep 2017 12:54:40 +0000 (12:54 +0000)]
tcptls: Change error message to debug.

The Websocket implementation will steal the underlying stream of
TCP/TLS sessions. This results in an error message being output
about a stream not being present when in reality this is actually
fine.

This change moves it to a debug message instead.

Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094

2 years agores_calendar: Various fixes
Sean Bright [Wed, 13 Sep 2017 19:08:39 +0000 (15:08 -0400)]
res_calendar: Various fixes

* The way that we were looking at XML elements for CalDAV was extremely
  fragile, so use SAX2 for increased robustness.

* Don't complain about a 'channel' not be specified if autoreminder is
  not set. Assume that if 'channel' is not set, we don't want to be
  notified.

* Fix some truncated CLI output in 'calendar show calendar' and make the
  'Autoreminder' description a bit more clear

ASTERISK-24588 #close
Reported by: Stefan Gofferje

ASTERISK-25523 #close
Reported by: Jesper

Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c

2 years agochan_rtp: Use μ-law by default instead of signed linear
Sean Bright [Wed, 13 Sep 2017 14:38:11 +0000 (10:38 -0400)]
chan_rtp: Use μ-law by default instead of signed linear

Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.

ASTERISK-21399 #close
Reported by: Tzafrir Cohen

Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3

2 years agores_pjsip: Add handling for incoming unsolicited MWI NOTIFY
George Joseph [Mon, 11 Sep 2017 10:46:35 +0000 (04:46 -0600)]
res_pjsip:  Add handling for incoming unsolicited MWI NOTIFY

A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.

res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.

Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c

2 years agoMerge "alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode"
Jenkins2 [Tue, 12 Sep 2017 19:30:41 +0000 (14:30 -0500)]
Merge "alembic:  Fix typo in add_auto_info_to_endpoint_dtmf_mode"

2 years agores_rtp_asterisk.c: Add doxygen to RTCP payload types.
Richard Mudgett [Sat, 9 Sep 2017 02:41:35 +0000 (21:41 -0500)]
res_rtp_asterisk.c: Add doxygen to RTCP payload types.

Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998

2 years agoMerge "cdr_pgsql: Refactor magic number by definition for version"
Joshua Colp [Mon, 11 Sep 2017 12:22:59 +0000 (07:22 -0500)]
Merge "cdr_pgsql: Refactor magic number by definition for version"

2 years agoMerge "alembic: Add support for MS-SQL"
Jenkins2 [Mon, 11 Sep 2017 11:55:12 +0000 (06:55 -0500)]
Merge "alembic: Add support for MS-SQL"

2 years agoalembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode
George Joseph [Mon, 11 Sep 2017 10:52:51 +0000 (04:52 -0600)]
alembic:  Fix typo in add_auto_info_to_endpoint_dtmf_mode

The downgrade function was missing "_v2" at the end of the
alter column type.

Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e

2 years agores/res_pjsip: Fix localnet checks in pjsip, part 2.
Walter Doekes [Sun, 10 Sep 2017 11:17:27 +0000 (13:17 +0200)]
res/res_pjsip: Fix localnet checks in pjsip, part 2.

In 45744fc53, I mistakenly broke SDP media address rewriting by
misinterpreting which address was checked in the localnet comparison.

Instead of checking the remote peer address to decide whether we need
media address rewriting, we check our local media address: if it's
local, then we rewrite. This feels awkward, but works and even made
directmedia work properly if you set local_net. (For the record: for
local peers, the SDP media rewrite code is not called, so the
comparison does no harm there.)

ASTERISK-27248 #close

Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f

2 years agocdr_pgsql: Refactor magic number by definition for version
Rodrigo Ramírez Norambuena [Sat, 9 Sep 2017 02:19:28 +0000 (23:19 -0300)]
cdr_pgsql: Refactor magic number by definition for version

Change-Id: I43f25976aa3069793ddbe0086833965a6fb0a518

2 years agoalembic: Add support for MS-SQL
Florian Floimair [Tue, 5 Sep 2017 16:13:19 +0000 (18:13 +0200)]
alembic: Add support for MS-SQL

MS-SQL has no native Enum-type support and therefore
needs to work with constraints.
Since these constraints need unique names the suggested approach
referenced in the following alembic documentation has been applied:
http://bit.ly/2x9r8pb

ASTERISK-27255 #close

Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95

2 years agoMerge "chan_sip: when getting sip pvt return failure if not found"
Jenkins2 [Fri, 8 Sep 2017 15:24:08 +0000 (10:24 -0500)]
Merge "chan_sip: when getting sip pvt return failure if not found"

2 years agoMerge "app_waitforsilence: Cleanup & don't treat missing frames as 'noise'"
Jenkins2 [Fri, 8 Sep 2017 15:20:10 +0000 (10:20 -0500)]
Merge "app_waitforsilence: Cleanup & don't treat missing frames as 'noise'"

2 years agoMerge "res_srtp: Add support for libsrtp2.1."
Joshua Colp [Fri, 8 Sep 2017 10:40:04 +0000 (05:40 -0500)]
Merge "res_srtp: Add support for libsrtp2.1."

2 years agoMerge "chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE"
Jenkins2 [Thu, 7 Sep 2017 18:04:35 +0000 (13:04 -0500)]
Merge "chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE"

2 years agoMerge "res_pjsip_session: Preserve stream name during renegotiation."
Jenkins2 [Thu, 7 Sep 2017 17:51:40 +0000 (12:51 -0500)]
Merge "res_pjsip_session: Preserve stream name during renegotiation."

2 years agoMerge "func_cdr: honour 'u' flag on dummy channel"
Jenkins2 [Thu, 7 Sep 2017 16:00:08 +0000 (11:00 -0500)]
Merge "func_cdr: honour 'u' flag on dummy channel"

2 years agoMerge "stasis/control.c: Fix set_interval_hook() ref leak."
Jenkins2 [Thu, 7 Sep 2017 15:46:43 +0000 (10:46 -0500)]
Merge "stasis/control.c: Fix set_interval_hook() ref leak."

2 years agofunc_cdr: honour 'u' flag on dummy channel
Jacek Konieczny [Tue, 5 Sep 2017 12:31:50 +0000 (14:31 +0200)]
func_cdr: honour 'u' flag on dummy channel

Fixes ${CDR(...,u)} when used in cdr_custom.conf

ASTERISK-27165 #close

Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83

2 years agoapp_waitforsilence: Cleanup & don't treat missing frames as 'noise'
Sean Bright [Wed, 6 Sep 2017 15:50:53 +0000 (11:50 -0400)]
app_waitforsilence: Cleanup & don't treat missing frames as 'noise'

* WaitForSilence completes successfully if it receives no media in the
  specified timeout, but when acting as WaitForNoise that logic needs
  to be reversed.

* Use standard argument parsing macros and add some error checking for
  invalid values.

* The documentation indicated that the first argument to both
  WaitForSilence and WaitForNoise was required when it was not. Update
  the documentation to reflect that.

* Wrap up some behavior in structs to avoid boolean checks all over the
  place.

ASTERISK-24066 #close
Reported by: M vd S

Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9

2 years agochan_sip: when getting sip pvt return failure if not found
Scott Griepentrog [Wed, 6 Sep 2017 21:05:32 +0000 (17:05 -0400)]
chan_sip: when getting sip pvt return failure if not found

In handle_request_invite, when processing a pickup, a call
is made to get_sip_pvt_from_replaces to locate the pvt for
the subscription. The pvt is assumed to be valid when zero
is returned indicating no error, and is dereferenced which
can cause a crash if it was not found.

This change checks the not found case and returns -1 which
allows the calling code to fail appropriately.

ASTERISK-27217 #close
Reported-by: Bryan Walters

Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612

2 years agostasis/control.c: Fix set_interval_hook() ref leak.
Richard Mudgett [Wed, 6 Sep 2017 18:38:17 +0000 (13:38 -0500)]
stasis/control.c: Fix set_interval_hook() ref leak.

Change-Id: Ia0edb7dc0dbbb879c079ff7000f1b722d86ce7dc

2 years agostasis/control: Fix possible deadlock with swap channel
George Joseph [Fri, 1 Sep 2017 10:17:02 +0000 (04:17 -0600)]
stasis/control:  Fix possible deadlock with swap channel

If an error occurs during a bridge impart it's possible that
the "bridge_after" callback might try to run before
control_swap_channel_in_bridge has been signalled to continue.
Since control_swap_channel_in_bridge is holding the control lock
and the callback needs it, a deadlock will occur.

* control_swap_channel_in_bridge now only holds the control
  lock while it's actually modifying the control structure and
  releases it while the bridge impart is running.
* bridge_after_cb is now tolerant of impart failures.

Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3

2 years agoMerge "alembic: Fix enum creation for dtls_fingerprint"
George Joseph [Wed, 6 Sep 2017 16:52:26 +0000 (11:52 -0500)]
Merge "alembic: Fix enum creation for dtls_fingerprint"

2 years agoMerge "alembic: fix erroneous commit for add_prune_on_boot"
Jenkins2 [Wed, 6 Sep 2017 15:55:35 +0000 (10:55 -0500)]
Merge "alembic: fix erroneous commit for add_prune_on_boot"

2 years agoMerge "res/res_pjsip: Standardize/fix localnet checks across pjsip."
Jenkins2 [Wed, 6 Sep 2017 15:17:06 +0000 (10:17 -0500)]
Merge "res/res_pjsip: Standardize/fix localnet checks across pjsip."

2 years agochan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE
Vitezslav Novy [Wed, 6 Sep 2017 10:23:25 +0000 (12:23 +0200)]
chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE

If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE
to both parties to set up media path directly between the endpoints.
In this reINVITE msg SDP origin line (o=) contains IP address of endpoint
instead of IP of asterisk. This behavior violates RFC3264, sec 8:
"When issuing an offer that modifies the session,
the "o=" line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP."
This patch assures IP address of Asterisk is always sent in
SDP origin line.

ASTERISK-17540
Reported by:  saghul

Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e

2 years agoMerge "formats: Restore previous fread() behavior"
Joshua Colp [Wed, 6 Sep 2017 14:25:40 +0000 (09:25 -0500)]
Merge "formats: Restore previous fread() behavior"

2 years agoalembic: Fix enum creation for dtls_fingerprint
George Joseph [Wed, 6 Sep 2017 12:54:00 +0000 (06:54 -0600)]
alembic: Fix enum creation for dtls_fingerprint

Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db

2 years agoMerge "res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel"
Jenkins2 [Wed, 6 Sep 2017 11:48:45 +0000 (06:48 -0500)]
Merge "res_pjsip_t38:  Make t38_reinvite_response_cb tolerant of NULL channel"

2 years agoMerge "res_calendar*, res_smdi: Move to "extended" support"
Jenkins2 [Wed, 6 Sep 2017 11:44:30 +0000 (06:44 -0500)]
Merge "res_calendar*, res_smdi: Move to "extended" support"

2 years agoalembic: fix erroneous commit for add_prune_on_boot
Florian Floimair [Tue, 5 Sep 2017 16:08:02 +0000 (18:08 +0200)]
alembic: fix erroneous commit for add_prune_on_boot

Added include for postgresql ENUM type and
redefined values in the same way as in the
other migration scripts.

ASTERISK-27254 #close

Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa

2 years agores_srtp: Add support for libsrtp2.1.
Alexander Traud [Wed, 6 Sep 2017 08:02:19 +0000 (10:02 +0200)]
res_srtp: Add support for libsrtp2.1.

Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro
SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with
previous versions of libSRTP, this change allows libSRTP 2.1.x as well.

ASTERISK-27253 #close

Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315

2 years agochan_pjsip: Suppress frame warnings.
Ben Ford [Tue, 5 Sep 2017 14:35:12 +0000 (09:35 -0500)]
chan_pjsip: Suppress frame warnings.

When rtp_keepalive is on for a PJSIP endpoint dialing to another
Asterisk instance also using PJSIP, Asterisk will continue to print
warning messages about not being able to send frames of a certain
type. This suppresses that warning message.

Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67

2 years agoformats: Restore previous fread() behavior
Sean Bright [Tue, 5 Sep 2017 15:05:48 +0000 (11:05 -0400)]
formats: Restore previous fread() behavior

Some formats are able to handle short reads while others are not, so
restore the previous behavior for the format modules so that we don't
have spurious errors when playing back files.

ASTERISK-27232 #close
Reported by: Jens T.

Change-Id: Iab7f52b25a394f277566c8a2a4b15a692280a300

2 years agores/res_pjsip: Standardize/fix localnet checks across pjsip.
Walter Doekes [Tue, 5 Sep 2017 14:16:01 +0000 (16:16 +0200)]
res/res_pjsip: Standardize/fix localnet checks across pjsip.

In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was
confusion about whether the transport_state->localnet ACL has ALLOW or
DENY semantics.

For the record: the localnet has DENY semantics, meaning that "not in
the list" means ALLOW, and the local nets are in the list.

Therefore, checks like this look wrong, but are right:

    /* See if where we are sending this request is local or not, and if
       not that we can get a Contact URI to modify */
    if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
        ast_debug(5, "Request is being sent to local address, "
                     "skipping NAT manipulation\n");

(In the list == localnet == DENY == skip NAT manipulation.)

And conversely, other checks that looked right, were wrong.

This change adds two macro's to reduce the confusion and uses those
instead:

    ast_sip_transport_is_nonlocal(transport_state, addr)
    ast_sip_transport_is_local(transport_state, addr)

ASTERISK-27248 #close

Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934