asterisk/asterisk.git
22 months 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

22 months 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"

22 months 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"

22 months 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

22 months agoMerge "app_directory: Handle a NULL mailbox without crashing"
Joshua Colp [Tue, 5 Sep 2017 13:41:19 +0000 (08:41 -0500)]
Merge "app_directory: Handle a NULL mailbox without crashing"

22 months agores_calendar*, res_smdi: Move to "extended" support
George Joseph [Tue, 5 Sep 2017 12:50:36 +0000 (06:50 -0600)]
res_calendar*, res_smdi: Move to "extended" support

Change-Id: I31eee8be30c6b0fc3dadb31111dd47742da8892d

22 months agoMerge "chan_ooh323: Fix confusing indentation warning"
Joshua Colp [Tue, 5 Sep 2017 12:16:41 +0000 (07:16 -0500)]
Merge "chan_ooh323: Fix confusing indentation warning"

22 months agores_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel
George Joseph [Tue, 5 Sep 2017 10:23:04 +0000 (04:23 -0600)]
res_pjsip_t38:  Make t38_reinvite_response_cb tolerant of NULL channel

t38_reinvite_response_cb can get called by res_pjsip_session's
session_inv_on_tsx_state_changed in situations where session->channel
is NULL.  If it is, the ast_log warning segfaults because it tries
to get the channel name from a NULL channel.

* Check session->channel and print "unknown channel" when it's NULL.

ASTERISK-27236
Reported by: Ross Beer

Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7

22 months agortp_engine: Prevent possible double free with DTLS config
Sean Bright [Fri, 1 Sep 2017 21:17:38 +0000 (17:17 -0400)]
rtp_engine: Prevent possible double free with DTLS config

ASTERISK-27225 #close
Reported by: Richard Kenner

Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89

22 months agochan_ooh323: Fix confusing indentation warning
Sean Bright [Fri, 1 Sep 2017 18:15:40 +0000 (14:15 -0400)]
chan_ooh323: Fix confusing indentation warning

ASTERISK-27177 #close
Reported by: Tzafrir Cohen

Change-Id: I40311c404edb2302a7543ad5ca7a06b2a38f2d97

22 months agoapp_directory: Handle a NULL mailbox without crashing
Sean Bright [Fri, 1 Sep 2017 14:51:06 +0000 (10:51 -0400)]
app_directory: Handle a NULL mailbox without crashing

ASTERISK-27241 #close
Reported by: David Moore

Change-Id: Ibbbca85517b04c315406ebfe3b6f7e0763daedc6

22 months agoMerge "chan_pjsip: Add tag info in CHANNEL function"
Jenkins2 [Thu, 31 Aug 2017 22:33:05 +0000 (17:33 -0500)]
Merge "chan_pjsip: Add tag info in CHANNEL function"

22 months agoMerge "res_rtp_asterisk: Allow remote SSRC to change on an RTP instance."
Joshua Colp [Thu, 31 Aug 2017 21:50:50 +0000 (16:50 -0500)]
Merge "res_rtp_asterisk: Allow remote SSRC to change on an RTP instance."

22 months agoMerge "res_rtp_asterisk: Only learn a new source in learn state."
Joshua Colp [Thu, 31 Aug 2017 13:34:48 +0000 (08:34 -0500)]
Merge "res_rtp_asterisk: Only learn a new source in learn state."

22 months agoMerge "pjsip_message_ip_updater: Fix issue handling "tel" URIs"
Jenkins2 [Thu, 31 Aug 2017 13:30:17 +0000 (08:30 -0500)]
Merge "pjsip_message_ip_updater:  Fix issue handling "tel" URIs"

22 months agopjsip_message_ip_updater: Fix issue handling "tel" URIs
George Joseph [Mon, 24 Jul 2017 15:48:14 +0000 (09:48 -0600)]
pjsip_message_ip_updater:  Fix issue handling "tel" URIs

sanitize_tdata was assuming all URIs were SIP URIs so when a non
SIP uri was in the From, To or Contact headers, the unconditional
cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused
a segfault when trying to access uri->other_param.

* Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri)
  checks before attempting to cast or use the returned uri.

ASTERISK-27152
Reported-by: Ross Beer

Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f

22 months agoAST-2017-006: Fix app_minivm application MinivmNotify command injection
Corey Farrell [Sun, 2 Jul 2017 00:24:27 +0000 (20:24 -0400)]
AST-2017-006: Fix app_minivm application MinivmNotify command injection

An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received.  The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.

* Add ast_safe_execvp() function.  This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding.  This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.

* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.

* Document code injection potential from untrusted data sources for other
shell commands that are under user control.

ASTERISK-27103

Change-Id: I7552472247a84cde24e1358aaf64af160107aef1

22 months agores_rtp_asterisk: Only learn a new source in learn state.
Joshua Colp [Mon, 22 May 2017 15:36:38 +0000 (15:36 +0000)]
res_rtp_asterisk: Only learn a new source in learn state.

This change moves the logic which learns a new source address
for RTP so it only occurs in the learning state. The learning
state is entered on initial allocation of RTP or if we are
told that the remote address for the media has changed. While
in the learning state if we continue to receive media from
the original source we restart the learning process. It is
only once we receive a sufficient number of RTP packets from
the new source that we will switch to it. Once this is done
the closed state is entered where all packets that do not
originate from the expected source are dropped.

The learning process has also been improved to take into
account the time between received packets so a flood of them
while in the learning state does not cause media to be switched.

Finally RTCP now drops packets which are not for the learned
SSRC if strict RTP is enabled.

ASTERISK-27013

Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c

22 months agores_rtp_asterisk: Allow remote SSRC to change on an RTP instance.
Joshua Colp [Wed, 30 Aug 2017 12:28:58 +0000 (12:28 +0000)]
res_rtp_asterisk: Allow remote SSRC to change on an RTP instance.

When SDP renegotiation occurs it is possible for an RTP
instance to be reused for a new stream, resulting in the remote
SSRC changing if it is part of a bundle group. This change
allows this and updates its mapping in the current bundle
group.

ASTERISK-27231

Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490

22 months agoMerge "bridge_native_rtp.c: Fixup native_rtp_framehook()"
Jenkins2 [Wed, 30 Aug 2017 13:58:35 +0000 (08:58 -0500)]
Merge "bridge_native_rtp.c: Fixup native_rtp_framehook()"

22 months agochan_pjsip: Add tag info in CHANNEL function
Andre Nazario [Sat, 26 Aug 2017 02:06:10 +0000 (23:06 -0300)]
chan_pjsip: Add tag info in CHANNEL function

Create local_tag and remote_tag in CHANNEL info to get tag from From and
To headers of a SIP dialog.

ASTERISK-27220

Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524

22 months agobridge_native_rtp.c: Fixup native_rtp_framehook()
Richard Mudgett [Tue, 29 Aug 2017 19:22:15 +0000 (14:22 -0500)]
bridge_native_rtp.c: Fixup native_rtp_framehook()

* Fix framehook to test frame type for control frame.
* Made framehook exit early if frame type is not a control frame.
* Eliminated RAII_VAR in framehook.
* Use switch instead of else-if ladder for control frame handling.

Change-Id: Ia555fc3600bd85470e3c0141147dbe3ad07c1d18

22 months agoconfbridge: Handle user hangup during name recording
Sean Bright [Tue, 29 Aug 2017 14:26:17 +0000 (10:26 -0400)]
confbridge: Handle user hangup during name recording

This prevents orphaned CBAnn channels from getting stuck in the bridge.

ASTERISK-26994 #close
Reported by: James Terhune

Change-Id: I5e43e832a9507ec3f2c59752cd900b41dab80457

22 months agoMerge "core: Reduce video update queueing."
Jenkins2 [Tue, 29 Aug 2017 11:13:09 +0000 (06:13 -0500)]
Merge "core: Reduce video update queueing."

22 months agoMerge "app_record: Resolve some absolute vs. relative filename bugs"
Jenkins2 [Tue, 29 Aug 2017 10:57:07 +0000 (05:57 -0500)]
Merge "app_record: Resolve some absolute vs. relative filename bugs"

22 months agoMerge "voicemail: Fix various abuses of mkstemp"
Jenkins2 [Tue, 29 Aug 2017 10:17:21 +0000 (05:17 -0500)]
Merge "voicemail: Fix various abuses of mkstemp"

22 months agocore: Reduce video update queueing.
Joshua Colp [Thu, 24 Aug 2017 16:45:08 +0000 (13:45 -0300)]
core: Reduce video update queueing.

A video update frame is used to indicate that a channel
with video negotiated should provide a full frame so the
decoder decoding the stream is able to do so. In situations
where a queue is used to store frames it makes no sense
for the queue to contain multiple video update frames. One
is sufficient to have a full frame be sent.

ASTERISK-27222

Change-Id: Id3f40a6f51b740ae4704003a1800185c0c658ee7

22 months agoMerge "res/res_pjsip_session: allow SDP answer to be regenerated"
Joshua Colp [Mon, 28 Aug 2017 12:34:47 +0000 (07:34 -0500)]
Merge "res/res_pjsip_session: allow SDP answer to be regenerated"

22 months agoMerge "alembic: Add dtls_fingerprint column in ps_endpoints table"
Jenkins2 [Mon, 28 Aug 2017 11:47:40 +0000 (06:47 -0500)]
Merge "alembic: Add dtls_fingerprint column in ps_endpoints table"

22 months agovoicemail: Fix various abuses of mkstemp
Sean Bright [Fri, 25 Aug 2017 18:44:35 +0000 (14:44 -0400)]
voicemail: Fix various abuses of mkstemp

mkstemp() returns a unique filename, but appending an extension to that
filename does not guarantee uniqueness. Instead, use mkdtemp() and we
can put whatever extension we want on the files that we create inside
the directory.

In the case of app_minivm, we also now properly clean up any temporary
files that we create.

ASTERISK-20858 #close
Reported by: Walter Doekes

Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43

22 months agoapp_record: Resolve some absolute vs. relative filename bugs
Sean Bright [Fri, 25 Aug 2017 17:20:16 +0000 (13:20 -0400)]
app_record: Resolve some absolute vs. relative filename bugs

If the Record() application is called with a relative filename that
includes directories, we were not properly creating the intermediate
directories and Record() would fail.

Secondarily, updated the documentation for RECORDED_FILE to mention
that it does not include a filename extension.

Finally, rewrote the '%d' functionality to be a bit more straight
forward and less noisy.

ASTERISK-16777 #close
Reported by: klaus3000

Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2

22 months agoMerge "app_queue: Evaluate realtime queues when running dialplan functions"
Jenkins2 [Fri, 25 Aug 2017 14:32:28 +0000 (09:32 -0500)]
Merge "app_queue: Evaluate realtime queues when running dialplan functions"

22 months agoMerge "chan_pjsip.c: Fix topology refresh response code accuracy."
Joshua Colp [Fri, 25 Aug 2017 13:32:43 +0000 (08:32 -0500)]
Merge "chan_pjsip.c: Fix topology refresh response code accuracy."

22 months agoMerge "app_voicemail: Honor escape digits in "greeting only" mode"
Joshua Colp [Fri, 25 Aug 2017 13:28:11 +0000 (08:28 -0500)]
Merge "app_voicemail: Honor escape digits in "greeting only" mode"

22 months agoalembic: Add dtls_fingerprint column in ps_endpoints table
Florian Floimair [Wed, 23 Aug 2017 15:01:09 +0000 (17:01 +0200)]
alembic: Add dtls_fingerprint column in ps_endpoints table

The ps_endpoints table was missing the dtls_fingerprint column
introduced with commit adba2a8d7fd.

ASTERISK-27168 #close

Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd

22 months agores/res_pjsip_session: allow SDP answer to be regenerated
Torrey Searle [Mon, 21 Aug 2017 09:28:52 +0000 (11:28 +0200)]
res/res_pjsip_session: allow SDP answer to be regenerated

If an SDP answer hasn't been sent yet, it's legal to change it.
This is required for PJSIP_DTMF_MODE to work correctly, and can
also have use in the future for updating codecs too.

ASTERISK-27209 #close

Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1

22 months agoapp_queue: Evaluate realtime queues when running dialplan functions
Sean Bright [Thu, 24 Aug 2017 14:42:24 +0000 (10:42 -0400)]
app_queue: Evaluate realtime queues when running dialplan functions

ASTERISK-19103 #close
Reported by: Jim Van Meggelen

Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b

22 months agoapp_voicemail: Honor escape digits in "greeting only" mode
Sean Bright [Wed, 23 Aug 2017 14:19:35 +0000 (10:19 -0400)]
app_voicemail: Honor escape digits in "greeting only" mode

ASTERISK-21241 #close
Reported by: Eelco Brolman
Patches:
Patch uploaded by Eelco Brolman (License 6442)

Change-Id: Icbe39b5c82a49b46cf1d168dc17766f3d84f54fe

22 months agores_smdi: Clean up memory leak
Sean Bright [Thu, 24 Aug 2017 13:35:45 +0000 (09:35 -0400)]
res_smdi: Clean up memory leak

Change-Id: I1e33290929e1aa7c5b9cb513f8254f2884974de8

22 months agoMerge "res_pjsip_session.c: Fix crash when declining an active stream."
Joshua Colp [Wed, 23 Aug 2017 19:49:26 +0000 (14:49 -0500)]
Merge "res_pjsip_session.c: Fix crash when declining an active stream."

22 months agoMerge changes from topic 'ASTERISK-27212'
Jenkins2 [Wed, 23 Aug 2017 19:45:52 +0000 (14:45 -0500)]
Merge changes from topic 'ASTERISK-27212'

* changes:
  bridge_channel.c: Fix FRACK when mapping frames to the bridge.
  bridge: Fix softmix bridge deadlock.

22 months agoMerge "channel: Fix topology API locking."
Jenkins2 [Wed, 23 Aug 2017 19:17:11 +0000 (14:17 -0500)]
Merge "channel: Fix topology API locking."

22 months agoMerge "app_confbridge: Document sfu video_mode value."
Joshua Colp [Wed, 23 Aug 2017 18:05:35 +0000 (13:05 -0500)]
Merge "app_confbridge: Document sfu video_mode value."

22 months agoMerge "bridge_softmix.c: Restored softmix_bridge_leave() shortcut exit."
Jenkins2 [Wed, 23 Aug 2017 17:21:47 +0000 (12:21 -0500)]
Merge "bridge_softmix.c: Restored softmix_bridge_leave() shortcut exit."

22 months agoMerge "confbridge.h: Fix doxygen comments."
Jenkins2 [Wed, 23 Aug 2017 17:05:29 +0000 (12:05 -0500)]
Merge "confbridge.h: Fix doxygen comments."

22 months agoMerge "bridge_softmix.c: Remove always true test."
Jenkins2 [Wed, 23 Aug 2017 16:55:01 +0000 (11:55 -0500)]
Merge "bridge_softmix.c: Remove always true test."

22 months agoMerge "app_queue: Fix initial hold time queue statistic"
Jenkins2 [Wed, 23 Aug 2017 16:11:38 +0000 (11:11 -0500)]
Merge "app_queue: Fix initial hold time queue statistic"

22 months 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.

ASTERISK-27212

Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420

22 months 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.

ASTERISK-27212

Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a

22 months 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.

ASTERISK-27212

Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691

22 months 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.

ASTERISK-27212

Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be

22 months agochan_pjsip.c: Fix topology refresh response code accuracy.
Richard Mudgett [Mon, 14 Aug 2017 17:20:25 +0000 (12:20 -0500)]
chan_pjsip.c: Fix topology refresh response code accuracy.

There are other 1xx and 2xx codes than 100 and 200 respectively.

Change-Id: I680db0997343256add1478714f5bf5b5569aee17

22 months agobridge_softmix.c: Restored softmix_bridge_leave() shortcut exit.
Richard Mudgett [Fri, 11 Aug 2017 22:06:01 +0000 (17:06 -0500)]
bridge_softmix.c: Restored softmix_bridge_leave() shortcut exit.

Change-Id: I13026cd90954e0265eab94a0faf635a3e11f0e35

22 months agoapp_confbridge: Document sfu video_mode value.
Richard Mudgett [Thu, 17 Aug 2017 22:07:18 +0000 (17:07 -0500)]
app_confbridge: Document sfu video_mode value.

Change-Id: I26e17df2c93f3933b23f78070603adbcc84ba204

22 months agoconfbridge.h: Fix doxygen comments.
Richard Mudgett [Thu, 17 Aug 2017 22:06:21 +0000 (17:06 -0500)]
confbridge.h: Fix doxygen comments.

Change-Id: I16133166a85fdb557c66ffcbfe8128d0b4725b0e

22 months agobridge_softmix.c: Remove always true test.
Richard Mudgett [Fri, 11 Aug 2017 16:40:46 +0000 (11:40 -0500)]
bridge_softmix.c: Remove always true test.

Change-Id: I26238df2ff0d0f6dfe95c3aa35da588f1ee71727

22 months 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"

22 months agoapp_queue: Fix initial hold time queue statistic
Sungtae Kim [Thu, 17 Aug 2017 21:46:49 +0000 (23:46 +0200)]
app_queue: Fix initial hold time queue statistic

Fixed to use correct initial value and fixed to use the
correct queue info to check the first value.

ASTERISK-27204

Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73

22 months 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"

22 months 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

23 months 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"

23 months 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

23 months 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).

ASTERISK-27202
Reported by: Sean McCord

Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99

23 months 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

23 months 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."

23 months 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

23 months 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.

ASTERISK-27147

Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0

23 months 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
protocol.

ASTERISK-27169
Reported by: Andrey Egorov
Tested by: Andrey Egorov

Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db

23 months 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."

23 months 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."

23 months 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."

23 months 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."

23 months 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."

23 months 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
buffer.

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

Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57

23 months 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.

ASTERISK-27147

Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83

23 months 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
transports.

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

ASTERISK-27147

Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4

23 months 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.

ASTERISK-27147

Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912

23 months 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

23 months 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

23 months 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

23 months 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"

23 months 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"

23 months 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

23 months 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

23 months 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"

23 months 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'.

ASTERISK-27189

Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce

23 months 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.

ASTERISK-27110

Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82

23 months 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."

23 months 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"

23 months 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."

23 months 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."

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

23 months 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"

23 months 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.

ASTERISK-27162

Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e

23 months 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.

ASTERISK-27182

Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03

23 months 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

23 months 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

23 months 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"

23 months 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"

23 months 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."