3 years agores_pjsip_session.c: Fix crash when declining an active stream.
Richard Mudgett [Fri, 18 Aug 2017 22:37:12 +0000 (17:37 -0500)]
res_pjsip_session.c: Fix crash when declining an active stream.

If a previously active stream is declined we could crash because the
channel's thread is still using the stream while we are updating the
topology in the serializer thread.

* Defer removing any declined stream's handler until we have blocked the
channel's thread with the channel lock.


Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420

3 years agobridge_channel.c: Fix FRACK when mapping frames to the bridge.
Richard Mudgett [Wed, 16 Aug 2017 22:50:18 +0000 (17:50 -0500)]
bridge_channel.c: Fix FRACK when mapping frames to the bridge.

* Add protection checks when mapping streams to the bridge.  The channel
and bridge may be in the process of updating the stream mapping when a
media frame comes in so we may not be able to map the frame at the time.

* We need to map the streams to the bridge's stream numbers right before
they are written into the bridge.  That way we don't have to keep
locking/unlocking the bridge and we won't have any synchronization
problems before the frames actually go into the bridge.

* Protect the deferred queue with the bridge_channel lock.


Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a

3 years agobridge: Fix softmix bridge deadlock.
Richard Mudgett [Wed, 16 Aug 2017 20:22:04 +0000 (15:22 -0500)]
bridge: Fix softmix bridge deadlock.

* Fix deadlock in
bridge_softmix.c:softmix_bridge_stream_topology_changed() between
bridge_channel and channel locks.

* The new bridge technology topology change callbacks must be called with
the bridge locked.  The callback references the bridge channel list, the
bridge technology could change, and the bridge stream mapping is updated.


Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be

3 years agochannel: Fix topology API locking.
Richard Mudgett [Fri, 11 Aug 2017 21:31:45 +0000 (16:31 -0500)]
channel: Fix topology API locking.

* ast_channel_request_stream_topology_change() must not be called with any
channel locks held.

* ast_channel_stream_topology_changed() must be called with only the
passed channel lock held.


Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691

3 years agoMerge "res_xmpp: fix inverted return code check in OAuth"
Jenkins2 [Tue, 22 Aug 2017 12:57:39 +0000 (07:57 -0500)]
Merge "res_xmpp: fix inverted return code check in OAuth"

3 years agoMerge "res_calendar_icalendar: Properly handle recurring events"
Joshua Colp [Tue, 22 Aug 2017 10:11:51 +0000 (05:11 -0500)]
Merge "res_calendar_icalendar: Properly handle recurring events"

3 years agores_xmpp: fix inverted return code check in OAuth
Michael Kuron [Sun, 20 Aug 2017 13:15:37 +0000 (15:15 +0200)]
res_xmpp: fix inverted return code check in OAuth

fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon
success and -1 if the function is not available.
This commit inverts the return code check so that an error is printed if the
module is not loaded and not if it is loaded.

ASTERISK-27207 #close

Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb

3 years agoMerge "Fix downloader not working with curl"
Jenkins2 [Fri, 18 Aug 2017 15:36:46 +0000 (10:36 -0500)]
Merge "Fix downloader not working with curl"

3 years agores_calendar_icalendar: Properly handle recurring events
Sean Bright [Thu, 17 Aug 2017 17:00:09 +0000 (13:00 -0400)]
res_calendar_icalendar: Properly handle recurring events

When looking for recurring events, use the correct end time based on the
configured 'timeframe.'

ASTERISK-27174 #close
Reported by: Mark Thompson

Change-Id: Id90c3cfc79d561a5521d79be176683e225f2edef

3 years agoFix downloader not working with curl
George Joseph [Wed, 16 Aug 2017 20:43:10 +0000 (14:43 -0600)]
Fix downloader not working with curl

The codec/dpma downloader wasn't handling curl correctly.  The logic
that transforms makeopts into a bash-sourceable file wasn't
handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was
looking for an 'or' command.

That logic has been eliminated.  Instead of trying to transform
and source makeopts, the downloader now calls a make scriptlet
to print the value of a specific variable.  This way, make handles
the ors (or any other make construct that happens to creep into
that file).

Reported by: Sean McCord

Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99

3 years agomanager: hook event is not being raised
Kevin Harwell [Tue, 15 Aug 2017 18:12:10 +0000 (13:12 -0500)]
manager: hook event is not being raised

When the iostream code went in it introduced a conditional that made it so the
hook event was not being raised even if a hook is present. This patch adds a
check to see if a hook is present in astman_append. If so then call into the
send_string function, which in turn raises the even for specified hook.

Also updated the ami hooks unit test, so the test could be automated.

ASTERISK-27200 #close

Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36

3 years agoMerge "configure: Check cache for valid pjproject tarball before downloading."
Jenkins2 [Wed, 16 Aug 2017 12:32:16 +0000 (07:32 -0500)]
Merge "configure: Check cache for valid pjproject tarball before downloading."

3 years agoconfigure: Check cache for valid pjproject tarball before downloading.
Richard Mudgett [Tue, 15 Aug 2017 20:15:58 +0000 (15:15 -0500)]
configure: Check cache for valid pjproject tarball before downloading.

On a fresh Asterisk source directory, the bundled pjproject tarball is
unconditionally downloaded even if the tarball is already in a specified
cache directory.

* Made check if the pjproject tarball is valid in the cache directory
before downloading the tarball on a fresh source directory.

Change-Id: Ic7ec842d3c97ecd8dafbad6f056b7fdbce41cae5

3 years agores_pjsip: Fix prune_on_boot to remove only contacts for the host.
Richard Mudgett [Tue, 15 Aug 2017 16:14:20 +0000 (11:14 -0500)]
res_pjsip: Fix prune_on_boot to remove only contacts for the host.

* Check that the contact's reg_server matches the host's name before
deleting any prune_on_boot contacts.  We don't want to delete reliable
transport contacts made with other servers if the ps_contacts database
table is shared with other servers.

Thanks to Ross Beer for pointing out that the original prune logic would
delete reliable transport contacts from other servers.


Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0

3 years agores_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif
Andrey Egorov [Fri, 4 Aug 2017 14:25:52 +0000 (17:25 +0300)]
res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif

Add ability to use tokens instead of passwords according to Google OAuth 2.0

Reported by: Andrey Egorov
Tested by: Andrey Egorov

Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db

3 years agoMerge "STUN/netsock2: Fix some valgrind uninitialized memory findings."
Jenkins2 [Mon, 14 Aug 2017 18:45:25 +0000 (13:45 -0500)]
Merge "STUN/netsock2: Fix some valgrind uninitialized memory findings."

3 years agoMerge "res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown."
George Joseph [Mon, 14 Aug 2017 17:20:21 +0000 (12:20 -0500)]
Merge "res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown."

3 years agoMerge "res_pjsip: Remove ephemeral registered contacts on transport shutdown."
George Joseph [Mon, 14 Aug 2017 17:20:14 +0000 (12:20 -0500)]
Merge "res_pjsip: Remove ephemeral registered contacts on transport shutdown."

3 years agoMerge "res_pjsip: PJSIP Transport state monitor refactor."
George Joseph [Mon, 14 Aug 2017 17:19:53 +0000 (12:19 -0500)]
Merge "res_pjsip: PJSIP Transport state monitor refactor."

3 years agoMerge "res_pjsip_transport_management.c: Rename some variables."
Jenkins2 [Mon, 14 Aug 2017 14:28:41 +0000 (09:28 -0500)]
Merge "res_pjsip_transport_management.c: Rename some variables."

3 years agoSTUN/netsock2: Fix some valgrind uninitialized memory findings.
Richard Mudgett [Thu, 10 Aug 2017 19:18:01 +0000 (14:18 -0500)]
STUN/netsock2: Fix some valgrind uninitialized memory findings.

* netsock2.c: Test the addr->len member first as it may be the only member
initialized in the struct.

* stun.c:ast_stun_handle_packet(): The combinded[] local array could get
used uninitialized by ast_stun_request().  The uninitialized string gets
copied to another location and could overflow the destination memory

These valgrind findings were found for ASTERISK_27150 but are not
necessarily a fix for the issue.

Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57

3 years agores_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown.
Richard Mudgett [Wed, 2 Aug 2017 23:44:12 +0000 (18:44 -0500)]
res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown.

The fix for the issue is broken up into three parts.

This is part three which handles the client side of REGISTER requests.
The registered contact may no longer be valid on the server when the
transport used is reliable and the connection is broken.

* Re-REGISTER our contact if the reliable transport is broken after
registration completes.  We attempt to re-REGISTER immediately to minimize
the time we are unreachable.  Time may have already passed between the
connection being broken and the loss being detected.

* Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's
are still correct if an allocation failure happens.


Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83

3 years agores_pjsip: Remove ephemeral registered contacts on transport shutdown.
Richard Mudgett [Mon, 31 Jul 2017 19:21:06 +0000 (14:21 -0500)]
res_pjsip: Remove ephemeral registered contacts on transport shutdown.

The fix for the issue is broken up into three parts.

This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled.  Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.

* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown.  If it is shutdown then the contact must be removed because it
is no longer valid.  Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there.  Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request.  The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.

* Prune any rewrite_contact's registered reliable transport contacts on
boot.  The reliable transport no longer exists so the contact is invalid.

* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.

* Made the websocket transport set a unique name since that is what we use
as the ao2 container key.  Otherwise, we would not know which transport we
find when one of them shuts down.  The names are also used for PJPROJECT
debug logging.

* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event.  Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket

* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.


Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4

3 years agores_pjsip: PJSIP Transport state monitor refactor.
Richard Mudgett [Fri, 28 Jul 2017 23:26:17 +0000 (18:26 -0500)]
res_pjsip: PJSIP Transport state monitor refactor.

The fix for the issue is broken up into three parts.

This is part one which refactors the transport state monitor code to allow
more modules to be able to monitor transports.

* Pull the management of PJPROJECT's transport state callback code from
res_pjsip_transport_management.c into res_pjsip.  Now other modules can
dynamically add and remove themselves from transport monitoring without
worrying about breaking PJPROJECT's callback chain.

* Add the ability for other modules to get a callback whenever a specific
transport is shutdown.


Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912

3 years agores_pjsip_transport_management.c: Rename some variables.
Richard Mudgett [Thu, 27 Jul 2017 20:36:20 +0000 (15:36 -0500)]
res_pjsip_transport_management.c: Rename some variables.

* Use monitored instead of the misleading keepalive name.

Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6

3 years agoUPGRADE notes: Prepare for the eventual 16 branch.
Richard Mudgett [Wed, 9 Aug 2017 20:24:58 +0000 (15:24 -0500)]
UPGRADE notes: Prepare for the eventual 16 branch.

Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c

3 years agores_pjsip_messaging: IPv6 receive address needs brackets
Scott Griepentrog [Thu, 10 Aug 2017 14:09:29 +0000 (09:09 -0500)]
res_pjsip_messaging: IPv6 receive address needs brackets

When handling an incoming SIP MESSAGE, PJSIP
attaches the IP address that the message was
received from to the message in the variable
PJSIP_RECVADDR.  When the IP address is IPv6
the :PORT appended results in an unparseable
mess. By using an additional bit flag on the
pj_sockaddr_print call, the conventional use
of brackets around the address is achieved.

ASTERISK-27193 #close

Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9

3 years agoMerge "Make --with-pjproject-bundled the default for Asterisk 15"
Jenkins2 [Thu, 10 Aug 2017 12:25:26 +0000 (07:25 -0500)]
Merge "Make --with-pjproject-bundled the default for Asterisk 15"

3 years agoMerge "res_rtp_asterisk: Make P2P bridge Asymmetric codec aware"
Jenkins2 [Wed, 9 Aug 2017 20:39:34 +0000 (15:39 -0500)]
Merge "res_rtp_asterisk:  Make P2P bridge Asymmetric codec aware"

3 years agores_rtp_asterisk: enable rtcp & QOS stats on native bridge
Torrey Searle [Wed, 26 Jul 2017 14:17:02 +0000 (16:17 +0200)]
res_rtp_asterisk: enable rtcp & QOS stats on native bridge

Asterisk wasn't generating or forwarding RTCP packets when native
bridge was activated.  Also the stats weren't available via
CHANNEL(qos). Now the RTCP stats are always calculated.

ASTERISK-27158 #close

Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b

3 years agores_rtp_asterisk: Make P2P bridge Asymmetric codec aware
Torrey Searle [Fri, 28 Jul 2017 12:53:44 +0000 (14:53 +0200)]
res_rtp_asterisk:  Make P2P bridge Asymmetric codec aware

Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not.  If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed

ASTERISK-26745 #close

Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f

3 years agoMerge "res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect"
Jenkins2 [Wed, 9 Aug 2017 13:15:24 +0000 (08:15 -0500)]
Merge "res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect"

3 years agoMake --with-pjproject-bundled the default for Asterisk 15
George Joseph [Tue, 8 Aug 2017 18:33:50 +0000 (12:33 -0600)]
Make --with-pjproject-bundled the default for Asterisk 15

'--with-pjproject-bundled' is now the default when running
./configure. It can be disabled with '--without-pjproject-bundled'.

To make building without an internet connection easier, a new
./configure option '--with-download-cache' was added that sets
the cache for externals (like pjproject, the codecs and the DPMA),
AND the sounds files.  It can also be specified as an environment
variable named "AST_DOWNLOAD_CACHE".  The existing
'--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and
'--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable
remain and if specified, will override '--with-downloads-cache'.


Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce

3 years agores_pjsip_session: Release media resources on session end quicker.
Joshua Colp [Sat, 5 Aug 2017 11:36:49 +0000 (11:36 +0000)]
res_pjsip_session: Release media resources on session end quicker.

A change was made long ago where the session was kept around
until the underlying INVITE session had been destroyed. This
had the side effect of also keeping the underlying media resources
around for this time as well.

This change ensures that when we are told to terminate the
session we immediately release any media sessions associated
with it.


Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82

3 years agoMerge "bridge: Fix stream topology/participant locking and video misrouting."
Jenkins2 [Mon, 7 Aug 2017 23:49:24 +0000 (18:49 -0500)]
Merge "bridge: Fix stream topology/participant locking and video misrouting."

3 years agoMerge "chan_sip: Access incoming REFER headers in dialplan"
Joshua Colp [Mon, 7 Aug 2017 14:51:57 +0000 (09:51 -0500)]
Merge "chan_sip: Access incoming REFER headers in dialplan"

3 years agoMerge "channel: Fix leak on successful call to chan->tech->requester."
Jenkins2 [Mon, 7 Aug 2017 14:30:21 +0000 (09:30 -0500)]
Merge "channel: Fix leak on successful call to chan->tech->requester."

3 years agoMerge "res_pjsip_nat.c: Remove unnecessary CMP_STOP."
Joshua Colp [Mon, 7 Aug 2017 13:31:50 +0000 (08:31 -0500)]
Merge "res_pjsip_nat.c: Remove unnecessary CMP_STOP."

3 years agoMerge "Support GMIME 3.0"
Jenkins2 [Mon, 7 Aug 2017 12:33:03 +0000 (07:33 -0500)]
Merge "Support GMIME 3.0"

3 years agoMerge "app_privacy: remove unused header asterisk/image.h"
Jenkins2 [Mon, 7 Aug 2017 12:04:13 +0000 (07:04 -0500)]
Merge "app_privacy: remove unused header asterisk/image.h"

3 years agochan_sip: Access incoming REFER headers in dialplan
kkm [Sun, 30 Jul 2017 01:03:02 +0000 (18:03 -0700)]
chan_sip: Access incoming REFER headers in dialplan

This adds a way to access information passed along with SIP headers in
a REFER message that initiates a transfer. Headers matching a dialplan
variable GET_TRANSFERRER_DATA in the transferrer channel are added to
a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH.

The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for
headers that should be put into the hash. If not set, no headers are
included. If set to a string (perhaps 'X-' in a typical case), all headers
starting this string are added. Empty string matches all headers.

If there are multiple of the same header, only the latest occurrence in
the REFER message is available in the hash.

Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the
referrer channel, and should be set with the '_' or '__' prefix.

I avoided a specific reference to SIP or REFER, as in my mind the mechanism
can be generalized to other channel techs.


Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e

3 years agobridge: Fix stream topology/participant locking and video misrouting.
Joshua Colp [Sun, 6 Aug 2017 16:15:34 +0000 (16:15 +0000)]
bridge: Fix stream topology/participant locking and video misrouting.

This change fixes a few locking issues and some video misrouting.

1. When accessing the stream topology of a channel the channel lock
must be held to guarantee the topology remains valid.

2. When a channel was joined to a bridge the bridge specific
implementation for stream mapping was not invoked, causing video
to be misrouted for a brief period of time.


Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03

3 years agochannel: Fix leak on successful call to chan->tech->requester.
Corey Farrell [Sat, 5 Aug 2017 19:43:39 +0000 (15:43 -0400)]
channel: Fix leak on successful call to chan->tech->requester.

joint_cap needs to be released unconditionally as chan->tech->requester
does not steal the reference even on success.

ASTERISK-27180 #close

Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6

3 years agores_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect
Kevin Harwell [Fri, 4 Aug 2017 21:47:30 +0000 (16:47 -0500)]
res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect

Currently, the handling of the msid attribute is not quite right. According to
the spec the msid's between the offer/answer are not dependent upon one another.
Meaning the same msid's given in an offer do not have to be returned in the
answer for a given stream. And they probably shouldn't be (copied/reused) since
this can potentially cause some browser side confusion.

This patch generates new msids when both an offer and answer are sent from
Asterisk. However, Asterisk does reuse the original msid it sent out for a
reinvite. Also audio+video streams are paired together by sharing the same
stream id, but a different track id.

ASTERISK-27179 #close

Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643

3 years agoMerge "alembic/res_pjsip: Add "webrtc" configuration option"
Jenkins2 [Fri, 4 Aug 2017 18:11:34 +0000 (13:11 -0500)]
Merge "alembic/res_pjsip: Add "webrtc" configuration option"

3 years agoMerge "chan_sip: Add dialplan function SIP_HEADERS"
Joshua Colp [Fri, 4 Aug 2017 17:57:58 +0000 (12:57 -0500)]
Merge "chan_sip: Add dialplan function SIP_HEADERS"

3 years agoMerge "Fix compile error for old versions of GCC."
Jenkins2 [Fri, 4 Aug 2017 17:03:23 +0000 (12:03 -0500)]
Merge "Fix compile error for old versions of GCC."

3 years agoMerge "Correct some leaks in unit tests."
Jenkins2 [Fri, 4 Aug 2017 16:50:45 +0000 (11:50 -0500)]
Merge "Correct some leaks in unit tests."

3 years agoMerge "res_pjsip_transport_websocket.c: Fix serializer ref leak."
Jenkins2 [Fri, 4 Aug 2017 15:51:42 +0000 (10:51 -0500)]
Merge "res_pjsip_transport_websocket.c: Fix serializer ref leak."

3 years agoMerge "res_pjsip_outbound_registration.c: Misc fixes."
Jenkins2 [Fri, 4 Aug 2017 15:00:56 +0000 (10:00 -0500)]
Merge "res_pjsip_outbound_registration.c: Misc fixes."

3 years agoCorrect some leaks in unit tests.
Corey Farrell [Fri, 4 Aug 2017 01:58:25 +0000 (21:58 -0400)]
Correct some leaks in unit tests.

* chan_sip: channel in test_sip_rtpqos_1.
* test_config: config hook, config info and global config holder.
* test_core_format: format in format_attribute_set_without_interface.
* test_stream: unneeded frame duplication.
* test_taskprocessor: task_data.

Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31

3 years agores_pjsip_transport_websocket.c: Fix serializer ref leak.
Richard Mudgett [Wed, 26 Jul 2017 22:49:57 +0000 (17:49 -0500)]
res_pjsip_transport_websocket.c: Fix serializer ref leak.

Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57

3 years agores_pjsip_outbound_registration.c: Misc fixes.
Richard Mudgett [Wed, 2 Aug 2017 23:41:49 +0000 (18:41 -0500)]
res_pjsip_outbound_registration.c: Misc fixes.

* Remove unnecessary CMP_STOP.

* In handle_client_registration() use DEBUG_ATLEAST() to only do work
needed for the debug log message when the debug log message is needed.

* In sip_outbound_registration_state_destroy() check state->registration
for NULL.

Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80

3 years agores_pjsip_nat.c: Remove unnecessary CMP_STOP.
Richard Mudgett [Tue, 1 Aug 2017 01:20:13 +0000 (20:20 -0500)]
res_pjsip_nat.c: Remove unnecessary CMP_STOP.

Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da

3 years agores_pjsip_registrar.c: Remove unnecessary CMP_STOP.
Richard Mudgett [Mon, 31 Jul 2017 19:20:02 +0000 (14:20 -0500)]
res_pjsip_registrar.c: Remove unnecessary CMP_STOP.

Most uses of CMP_STOP are superfluous and are only respected when
OBJ_MULTIPLE is used to search the container.

Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8

3 years agoSupport GMIME 3.0
Tzafrir Cohen [Thu, 3 Aug 2017 18:13:01 +0000 (14:13 -0400)]
Support GMIME 3.0

Support building the Asterisk httpd with version 3.0 of gmime as
well as earlier versions of that library.


Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f

3 years agoalembic/res_pjsip: Add "webrtc" configuration option
Kevin Harwell [Wed, 2 Aug 2017 14:43:56 +0000 (09:43 -0500)]
alembic/res_pjsip: Add "webrtc" configuration option

When the "webrtc" option was added in res_pjsip it was not added to the alembic
scripts. This patch adds the option for alembic.

Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of
an OPT_BOOL_T so if this field is ever written to a database it will write out
the correct value.

ASTERISK-27119 #close

Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b

3 years agochan_sip: Add dialplan function SIP_HEADERS
kkm [Sun, 30 Jul 2017 06:17:00 +0000 (23:17 -0700)]
chan_sip: Add dialplan function SIP_HEADERS

Syntax: SIP_HEADERS([prefix])

If the argument is specified, only the headers matching the given prefix
are returned.

The function returns a comma-separated list of SIP header names from an
incoming INVITE message. Multiple headers with the same name are included
in the list only once. The returned list can be iterated over using the
functions POP() and SIP_HEADER().

For example, '${SIP_HEADERS(Co)}' might return the string

Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional
extended headers sent by a peer.


Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267

3 years agoFix compile error for old versions of GCC.
Corey Farrell [Wed, 2 Aug 2017 19:16:43 +0000 (15:16 -0400)]
Fix compile error for old versions of GCC.

Use -Wno-format-truncation only if supported by compiler.

ASTERISK-27171 #close

Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6

3 years agoapp_privacy: remove unused header asterisk/image.h
Corey Farrell [Wed, 2 Aug 2017 21:08:38 +0000 (17:08 -0400)]
app_privacy: remove unused header asterisk/image.h

Change-Id: I56ed530633a642633b18383821069e806c92ae82

3 years agores_pjsip_pidf_eyebeam_body_supplement: Correct status presentation
Sean Bright [Wed, 26 Jul 2017 13:48:29 +0000 (09:48 -0400)]
res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation

This change fixes PIDF content generation when the underlying device
state is considered in use. Previously it was incorrectly marked
as closed meaning they were offline/unavailable. The code now
correctly marks them as open.


  * Generate an XML element for our activity instead of a using a text

  * Consider every extension state other than "unavailable" to be 'open'

  * Update the XML namespaces and structure to reflect those
    documented in RFC 4480

  * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the
    "in use" activity. This change results in eyeBeam using the
    appropriate icon for the watched user.

This was tested on eyeBeam build 59030 on Windows.

ASTERISK-26659 #close
Reported by: Abraham Liebsch
  ASTERISK-26659.diff submitted by snuffy (license 5024)

Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810

3 years agores_pjsip: Add support for dnsmgr to external_media_address.
Joshua Colp [Sun, 23 Jul 2017 23:34:32 +0000 (23:34 +0000)]
res_pjsip: Add support for dnsmgr to external_media_address.

The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to

Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2

3 years agoFix compiler warnings on Fedora 26 / GCC 7.
Corey Farrell [Fri, 28 Jul 2017 01:58:22 +0000 (21:58 -0400)]
Fix compiler warnings on Fedora 26 / GCC 7.

GCC 7 has added capability to produce warnings, this fixes most of those
warnings.  The specific warnings are disabled in a few places:

* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().

ASTERISK-27156 #close

Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88

3 years agoapp_queue: Add announce-position-only-up option
Sean Bright [Wed, 26 Jul 2017 14:27:00 +0000 (10:27 -0400)]
app_queue: Add announce-position-only-up option

Setting this option will cause the Queue application to only announce
the caller's position if it has improved since the last time that we
announced it.

Change-Id: I173a124121422209485b043e2bf784f54242fce6

3 years agobundled_pjproject: Improve SSL/TLS error handling
George Joseph [Thu, 27 Jul 2017 11:35:51 +0000 (05:35 -0600)]
bundled_pjproject:  Improve SSL/TLS error handling

OpenSSL has 2 levels or error processing.  It's possible for the
top layer to return SSL_ERROR_SYSCALL but the lower layer return
no error, in which case processing should continue.  Only the top
layer was being examined though so connections were being torn
down when they didn't need to be.  This patch adds the examination
of the lower level codes, and if they return no errors, allows
processing to continue.

Reported-by: Ian Gilmour
pjproject-2.6.patch submitted by Ian Gilmour (license 6889)

Updated-by: George Joseph and Sauw Ming (Teluu)

Merged to upstream pjproject on 7/27/2017 (commit 5631)

Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2

3 years agochan_pjsip: add a new function PJSIP_DTMF_MODE
Torrey Searle [Mon, 26 Jun 2017 12:52:52 +0000 (14:52 +0200)]
chan_pjsip: add a new function PJSIP_DTMF_MODE

This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis

ASTERISK-27085 #close

Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612

3 years agores_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours
Sean Bright [Tue, 25 Jul 2017 20:17:45 +0000 (16:17 -0400)]
res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours

Change-Id: Ia578ede1a55b21014581793992a429441903278b

3 years agoMerge "Core: Add support for systemd socket activation."
Jenkins2 [Wed, 26 Jul 2017 14:17:40 +0000 (09:17 -0500)]
Merge "Core: Add support for systemd socket activation."

3 years agoMerge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."
Joshua Colp [Wed, 26 Jul 2017 13:31:13 +0000 (08:31 -0500)]
Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."

3 years agoMerge "res_stasis_device_state: Unsubscribe should remove old subscriptions"
Joshua Colp [Wed, 26 Jul 2017 13:27:31 +0000 (08:27 -0500)]
Merge "res_stasis_device_state: Unsubscribe should remove old subscriptions"

3 years agoMerge "SDP: Create declined m= SDP lines using remote SDP if applicable."
Joshua Colp [Wed, 26 Jul 2017 13:20:38 +0000 (08:20 -0500)]
Merge "SDP: Create declined m= SDP lines using remote SDP if applicable."

3 years agoMerge "SDP: Rework SDP offer/answer model and update capabilities merges."
Joshua Colp [Wed, 26 Jul 2017 13:20:35 +0000 (08:20 -0500)]
Merge "SDP: Rework SDP offer/answer model and update capabilities merges."

3 years agoMerge "app_voicemail.c: Allow mailbox entry on authentication retry prompt."
Jenkins2 [Wed, 26 Jul 2017 11:49:41 +0000 (06:49 -0500)]
Merge "app_voicemail.c: Allow mailbox entry on authentication retry prompt."

3 years agoMerge "core: Add VP9 passthrough support."
Jenkins2 [Tue, 25 Jul 2017 15:37:45 +0000 (10:37 -0500)]
Merge "core: Add VP9 passthrough support."

3 years agores_stasis_device_state: Unsubscribe should remove old subscriptions
Sergej Kasumovic [Thu, 20 Jul 2017 13:08:05 +0000 (15:08 +0200)]
res_stasis_device_state: Unsubscribe should remove old subscriptions

Case scenario with Applications ARI:

* Once you subscribe to deviceState with Applications REST API, it will be
added into subscription pool.

* When you unsubscribe it will remove from the device_state_subscription
hash table but not from the subscription pool.

* When you subscribe again, it will add it to pool again.

* Now you will have two subscriptions and you will receive same event

This fix should now remove deviceState subscription from pool and it
should fix unsubscribe on deviceState.

ASTERISK-27130 #close

Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4

3 years agoMerge "say.c: Fix file locations for second, seconds, minute, minutes files"
George Joseph [Tue, 25 Jul 2017 12:45:00 +0000 (07:45 -0500)]
Merge "say.c: Fix file locations for second, seconds, minute, minutes files"

3 years agocore: Add VP9 passthrough support.
Joshua Colp [Mon, 24 Jul 2017 18:30:59 +0000 (18:30 +0000)]
core: Add VP9 passthrough support.

This change adds VP9 as a known codec and creates a cached
"vp9" media format for use.

Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc

3 years agoMerge "format.h: Fix a few minor errors in comments."
Jenkins2 [Mon, 24 Jul 2017 15:54:30 +0000 (10:54 -0500)]
Merge "format.h: Fix a few minor errors in comments."

3 years agoMerge "Update make_ari_stubs in master to make the version 16"
Joshua Colp [Mon, 24 Jul 2017 12:41:43 +0000 (07:41 -0500)]
Merge "Update make_ari_stubs in master to make the version 16"

3 years agoMerge "Restore the incorrectly deleted spandspflow2pcap.log"
Jenkins2 [Mon, 24 Jul 2017 12:05:59 +0000 (07:05 -0500)]
Merge "Restore the incorrectly deleted spandspflow2pcap.log"

3 years agoapp_voicemail.c: Allow mailbox entry on authentication retry prompt.
Richard Mudgett [Wed, 19 Jul 2017 23:11:19 +0000 (18:11 -0500)]
app_voicemail.c: Allow mailbox entry on authentication retry prompt.

The following testsuite voicemail tests were failing to re-enter the
mailbox after the first login attempt.


The tests were noting the start of the vm-incorrect-mailbox prompt and
immediately sending the mailbox for the next login attempt.  Since the
invalid message playback had to complete before the digits were
recognized, the test passed for the wrong reason and added approximately
20 seconds to the test times.

* Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox
digits like the initial vm-login prompt so the tests are able to enter the
intended mailbox.

Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8

3 years agoformat.h: Fix a few minor errors in comments.
Matthew Fredrickson [Fri, 21 Jul 2017 20:57:46 +0000 (15:57 -0500)]
format.h: Fix a few minor errors in comments.

A few minor problems were found in comments in format.h.  This patch fixes them.

Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94

3 years agosay.c: Fix file locations for second, seconds, minute, minutes files
Rusty Newton [Fri, 14 Jul 2017 18:47:50 +0000 (13:47 -0500)]
say.c: Fix file locations for second, seconds, minute, minutes files

The seconds and minutes files have always existed in the base language
directory of the Core package. So say.c has always been calling the wrong
location (under digits/) for those two files and in the case of second and
minute they didn't exist in the Core packages at all.

The 1.6 sounds release moves the second and minute files into Core from
Extra for the languages that already had them. A future release will include
the second and minute files for languages that didn't already have them.

This patch just changes all the target locations for second, seconds,
minute, and minutes that were under the digits subdir to be under the root of
sounds instead. Which is where the sounds will be for some languages after 1.6
sounds and for all languages after a future release.

ASTERISK-25810 #close

Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702
Reported-by: Nicolas Riendeau

3 years agoSounds: Update Makefile for Extra sounds 1.5.1 release
Rusty Newton [Fri, 21 Jul 2017 19:20:10 +0000 (14:20 -0500)]
Sounds: Update Makefile for Extra sounds 1.5.1 release

Incrementing version for the Extra sounds release. 1.5.1 Extra sounds
removes two prompts that were moved into the Core packages in the 1.6 Core
sounds release.

ASTERISK-27142 #close

Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7

3 years agoUpdate make_ari_stubs in master to make the version 16
George Joseph [Fri, 21 Jul 2017 16:17:38 +0000 (10:17 -0600)]
Update make_ari_stubs in master to make the version 16

Ready for next major version

Change-Id: If9dc99b3b78768529e69a297d8f87e23582ca6d0

3 years agoRestore the incorrectly deleted spandspflow2pcap.log
George Joseph [Fri, 21 Jul 2017 16:24:24 +0000 (10:24 -0600)]
Restore the incorrectly deleted spandspflow2pcap.log

Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5

3 years agoMerge "corosync: Fix corosync library name in"
George Joseph [Fri, 21 Jul 2017 11:54:00 +0000 (06:54 -0500)]
Merge "corosync: Fix corosync library name in"

3 years agoMerge "Update AMI and ARI versions for master/15 and update UPDATE.txt"
Jenkins2 [Thu, 20 Jul 2017 17:17:48 +0000 (12:17 -0500)]
Merge "Update AMI and ARI versions for master/15 and update UPDATE.txt"

3 years agoMerge "pjsip: Increase maximum packet size."
George Joseph [Thu, 20 Jul 2017 16:08:44 +0000 (11:08 -0500)]
Merge "pjsip: Increase maximum packet size."

3 years agoUpdate AMI and ARI versions for master/15 and update UPDATE.txt
George Joseph [Thu, 20 Jul 2017 15:52:38 +0000 (09:52 -0600)]
Update AMI and ARI versions for master/15 and update UPDATE.txt

AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0

Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago

Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7

3 years agocorosync: Fix corosync library name in
Sean Bright [Thu, 20 Jul 2017 14:57:08 +0000 (10:57 -0400)]
corosync: Fix corosync library name in

Also add new corosync packages to install_prereq.

Reported by Travis Ryan in #asterisk-dev

Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db

3 years agoMerge "core: Add digit filtering to ast_waitfordigit_full"
Joshua Colp [Wed, 19 Jul 2017 18:09:56 +0000 (13:09 -0500)]
Merge "core: Add digit filtering to ast_waitfordigit_full"

3 years agoMerge "app_playback.c: Use the timezonename parameter"
George Joseph [Wed, 19 Jul 2017 17:11:09 +0000 (12:11 -0500)]
Merge "app_playback.c: Use the timezonename parameter"

3 years agoMerge "bridge_softmix: Use removed stream spots when renegotiating."
Jenkins2 [Wed, 19 Jul 2017 15:42:51 +0000 (10:42 -0500)]
Merge "bridge_softmix: Use removed stream spots when renegotiating."

3 years agoMerge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."
Jenkins2 [Wed, 19 Jul 2017 14:25:59 +0000 (09:25 -0500)]
Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."

3 years agobridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
Joshua Colp [Mon, 17 Jul 2017 16:01:24 +0000 (16:01 +0000)]
bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.

This change does a few things to improve packet loss and renegotiation:

1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.

2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.

3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.

4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.

5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.


Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45

3 years agopjsip: Increase maximum packet size.
Benjamin Keith Ford [Tue, 18 Jul 2017 20:04:44 +0000 (15:04 -0500)]
pjsip: Increase maximum packet size.

The maximum packet size for PJSIP has been increased to handle the
multiple streams being added for WebRTC.

Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3

3 years agoMerge "app_queue: Add change priority of call"
George Joseph [Tue, 18 Jul 2017 14:37:36 +0000 (09:37 -0500)]
Merge "app_queue: Add change priority of call"

3 years agoMerge "bridge_softmix: Don't reorder streams on participant leaving."
Jenkins2 [Tue, 18 Jul 2017 13:13:15 +0000 (08:13 -0500)]
Merge "bridge_softmix: Don't reorder streams on participant leaving."

3 years agoMerge "bridge/core_unreal: Fix SFU bugs with forwarding frames."
Jenkins2 [Mon, 17 Jul 2017 22:59:32 +0000 (17:59 -0500)]
Merge "bridge/core_unreal: Fix SFU bugs with forwarding frames."