asterisk/asterisk.git
14 months agopjsip_options: handle modification of qualify options in realtime
Alexei Gradinari [Tue, 22 May 2018 21:21:10 +0000 (17:21 -0400)]
pjsip_options: handle modification of qualify options in realtime

Currentrly pjsip_options code does not handle the situation when the
qualify options were changed in realtime database.
Only 'module reload res_pjsip' helps.

This patch add a check on contact add/update observers if the contact
qualify options are different than local aor qualify options.
If the qualify options were modified then synchronize
the pjsip_options AOR local state.

ASTERISK-27872

Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62

14 months agoMerge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated."
George Joseph [Tue, 5 Jun 2018 18:01:31 +0000 (13:01 -0500)]
Merge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated."

14 months agoMerge "app_meetme: Fix manager event documentation for several events."
Joshua Colp [Tue, 5 Jun 2018 11:54:00 +0000 (06:54 -0500)]
Merge "app_meetme: Fix manager event documentation for several events."

14 months agoapp_sendtext: Allow content types other than text/plain
George Joseph [Mon, 4 Jun 2018 14:50:51 +0000 (08:50 -0600)]
app_sendtext:  Allow content types other than text/plain

There was no real reason to limit the conteny type to text/plain other
than that's what it was limited to before.  Now any text/* content
type will be allowed for channel drivers that don't support enhanced
messaging and any type will be allowed for channel drivers that do
support enhanced messaging.

Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9

14 months agoMerge "ast_coredumper: Fix output directory and variable precedence"
Joshua Colp [Thu, 31 May 2018 10:16:08 +0000 (05:16 -0500)]
Merge "ast_coredumper:  Fix output directory and variable precedence"

14 months agoapp_meetme: Fix manager event documentation for several events.
Richard Mudgett [Tue, 29 May 2018 17:28:48 +0000 (12:28 -0500)]
app_meetme: Fix manager event documentation for several events.

The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and
MeetmeTalkRequest AMI events were documented with sending out a Usernum
header when the User header was actually output.

* Change the online documentation to match reality.

ASTERISK-27873
ASTERISK-25261

Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39

14 months agoMerge "libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated."
Joshua Colp [Tue, 29 May 2018 17:07:51 +0000 (12:07 -0500)]
Merge "libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated."

14 months agotcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated.
Alexander Traud [Fri, 25 May 2018 12:22:14 +0000 (14:22 +0200)]
tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated.

ASTERISK-27874

Change-Id: Ica65113511c7a1c13f7988e7d9e7d9e7f3f620dd

14 months agoMerge "rtp: Add support for RTP extension negotiation and abs-send-time."
Joshua Colp [Thu, 24 May 2018 20:26:57 +0000 (15:26 -0500)]
Merge "rtp: Add support for RTP extension negotiation and abs-send-time."

14 months agoMerge "res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change"
Joshua Colp [Thu, 24 May 2018 20:09:58 +0000 (15:09 -0500)]
Merge "res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change"

14 months agoast_coredumper: Fix output directory and variable precedence
George Joseph [Tue, 15 May 2018 13:45:20 +0000 (07:45 -0600)]
ast_coredumper:  Fix output directory and variable precedence

The OUTPUTDIR variable in ast_debug_tools.conf.sample is now set
to "/tmp" instead of "/some/directory".

Variables set on the command line or that are already in the
environment now take predecence over variables set in the config files.

ASTERISK-27846
Reported by: Ted G

Change-Id: Ie8baec52d531886bf5849ec1d59bb59dc87ad387

14 months agoMerge "tcptls: Repair ./configure --with-ssl=PATH."
Joshua Colp [Thu, 24 May 2018 11:20:15 +0000 (06:20 -0500)]
Merge "tcptls: Repair ./configure --with-ssl=PATH."

14 months agoMerge "app_queue: Update year Copyright and fix missing tabs in documentation"
Joshua Colp [Thu, 24 May 2018 10:49:41 +0000 (05:49 -0500)]
Merge "app_queue: Update year Copyright and fix missing tabs in documentation"

14 months agoMerge "config.c: Fix successful DELETE treated as failure"
Joshua Colp [Thu, 24 May 2018 10:49:21 +0000 (05:49 -0500)]
Merge "config.c: Fix successful DELETE treated as failure"

14 months agoMerge "channel.c: Fix off nominal channel allocation failure path."
Joshua Colp [Thu, 24 May 2018 10:18:16 +0000 (05:18 -0500)]
Merge "channel.c: Fix off nominal channel allocation failure path."

14 months agores/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change
Torrey Searle [Wed, 9 May 2018 13:31:47 +0000 (15:31 +0200)]
res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change

Certain race conditions between changing bridge types and DTMF can
cause the current FLAG_NEED_MARKER_BIT to send the marker bit before
the actual first packet of native bridging.

This logic keeps track of the ssrc the bridge is currently sending
and will correctly ensure the marker bit is set if SSRC as changed
from the previous sent packet.

ASTERISK-27845

Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b

14 months agoMerge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"
Joshua Colp [Wed, 23 May 2018 17:10:13 +0000 (12:10 -0500)]
Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"

15 months agortp: Add support for RTP extension negotiation and abs-send-time.
Joshua Colp [Mon, 23 Apr 2018 14:04:01 +0000 (14:04 +0000)]
rtp: Add support for RTP extension negotiation and abs-send-time.

When RTP was originally created it had the ability to place a single
extension in an RTP packet. In practice people wanted to potentially
put multiple extensions in one and so RFC 5285 (obsoleted by RFC
8285) came into existence. This allows RTP extensions to be negotiated
with a unique identifier to be used in the RTP packet, allowing
multiple extensions to be present in the packet.

This change extends the RTP engine API to add support for this. A
user of it can enable extensions and the API provides the ability to
retrieve the information (to construct SDP for example) and to provide
negotiated information (from SDP). The end result is that the RTP
engine can then query to see if the extension has been negotiated and
what unique identifier is to be used. It is then up to the RTP engine
implementation to construct the packet appropriately.

The first extension to use this support is abs-send-time which is
defined in the REMB draft[1] and is a second timestamp placed in an
RTP packet which is for when the packet has left the sending system.
It is used to more accurately determine the available bandwidth.

ASTERISK-27831

[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03

Change-Id: I508deac557867b1e27fc7339be890c8018171588

15 months agochannel.c: Fix off nominal channel allocation failure path.
Richard Mudgett [Tue, 22 May 2018 22:17:31 +0000 (17:17 -0500)]
channel.c: Fix off nominal channel allocation failure path.

__ast_channel_alloc_ap() had a failure exit path that hadn't setup the fd
descriptors to -1 yet.  The destructor would then attempt to close these
fd's that had never been opened.

Change-Id: Icf21093f36c60781e8cf6ee9d586536302af33e3

15 months agoapp_queue: Update year Copyright and fix missing tabs in documentation
Rodrigo Ramírez Norambuena [Fri, 18 May 2018 17:46:03 +0000 (13:46 -0400)]
app_queue: Update year Copyright and fix missing tabs in documentation

Change-Id: Ieb8faf37dc765463ee5dbca1d1343242c756b1c7

15 months agoconfig.c: Fix successful DELETE treated as failure
Alexei Gradinari [Fri, 18 May 2018 21:45:22 +0000 (17:45 -0400)]
config.c: Fix successful DELETE treated as failure

The config engine destroy_func callback function returns the number of
rows deleted or -1 on error.  But the function
ast_destroy_realtime_fields treated non-zero return values as error.

ASTERISK-27863

Change-Id: Ied02b38e8196cb03043e609a0679feebd288d17b

15 months agonetsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API
Matthew Fredrickson [Mon, 14 May 2018 11:07:34 +0000 (06:07 -0500)]
netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API

This function originally was used in chan_sip to enable some simplifying
assumptions and eventually was copy and pasted into res_pjsip_logger and
res_hep.  Since it's replicated in three places, it's probably best to
move it into the public netsock2 API for these modules to use.

Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04

15 months agoMerge "app_voicemail: Fix data-type mismatch between app_voicemail and database"
Joshua Colp [Mon, 21 May 2018 14:05:19 +0000 (09:05 -0500)]
Merge "app_voicemail: Fix data-type mismatch between app_voicemail and database"

15 months agolibasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated.
Alexander Traud [Sun, 20 May 2018 11:36:51 +0000 (13:36 +0200)]
libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated.

Use CRYPTO_set_id_callback(.) only with OpenSSL 0.9.8 and older.

ASTERISK-27867

Change-Id: Iadd58d5bf6f538eb224203970a4e88e26f259655

15 months agotcptls: Repair ./configure --with-ssl=PATH.
Alexander Traud [Sat, 19 May 2018 13:23:30 +0000 (15:23 +0200)]
tcptls: Repair ./configure --with-ssl=PATH.

SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 got discovered without honoring a PATH.

ASTERISK-27865

Change-Id: I8cd358eed7411726d08fa7b01691bef122fbeb71

15 months agoMerge "chan_mobile: support handling of caller-id names ("cnam")."
Kevin Harwell [Fri, 18 May 2018 22:44:23 +0000 (17:44 -0500)]
Merge "chan_mobile: support handling of caller-id names ("cnam")."

15 months agoMerge "app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail"
Kevin Harwell [Fri, 18 May 2018 22:18:59 +0000 (17:18 -0500)]
Merge "app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail"

15 months agoMerge "rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again."
Kevin Harwell [Fri, 18 May 2018 21:42:29 +0000 (16:42 -0500)]
Merge "rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again."

15 months agoapp_voicemail: Fix data-type mismatch between app_voicemail and database
Nic Colledge [Tue, 27 Mar 2018 23:53:07 +0000 (00:53 +0100)]
app_voicemail: Fix data-type mismatch between app_voicemail and database

Fix data-type mismatch between app_voicemail and database columns
exposed by new version of MariaDB

ASTERISK-27760

Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b

15 months agoapp_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail
Nic Colledge [Sat, 12 May 2018 11:53:13 +0000 (12:53 +0100)]
app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail

Correct the log warning message shown when ODBC voicemail
retrieve_file is called and there is a null value in the category
column.
A more meaningfull message is now written at debug level.

ASTERISK-27853

Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4

15 months agochan_mobile: support handling of caller-id names ("cnam").
Brian P. Martin [Wed, 18 Apr 2018 02:15:08 +0000 (19:15 -0700)]
chan_mobile: support handling of caller-id names ("cnam").

Add support to handle caller-ID names ("cnam") in addition to caller-ID
numbers.  The prior code ignored the caller-ID name altogether, and
used the local name for the cell phone (e.g. "my-iphone") in its place.

Note: as of this writing, at least some Android phones don't pass cnam to
us. This can be seen by issuing "core set debug 2" in the CLI and watching
the "CLIP" record when a call comes in.  If cnam isn't in the CLIP record,
there's nothing we can do to provide one.  We'll provide a null cnam field,
so later Asterisk processes know to try other sources (e.g. cidname database,
OpenCNAM, etc.).

Reported by: Brian Martin
Tested by: Brian Martin
ASTERISK-27726

Change-Id: I89490d85fa406c36261879c50ae5e65595538ba5

15 months agores_pjsip_endpoint_identifier_ip: Unregister the module for headers.
Alexander Traud [Thu, 17 May 2018 06:58:43 +0000 (08:58 +0200)]
res_pjsip_endpoint_identifier_ip: Unregister the module for headers.

Asterisk uses Reference Counting to track whether a module can be unloaded.
Every consumer who requires a module, increases the reference count. When the
consumer goes, is unloaded itself, it has to decrease the reference count on
all its used/required modules. That way
 core stop gracefully
works on the command-line interface (CLI): One module after the other is
unloaded. A recent change broke this for the module res_pjsip.

ASTERISK-27861

Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3

15 months agoMerge "cli: Display correct unit for HTTP timeout in "manager show settings"."
Joshua Colp [Wed, 16 May 2018 18:56:48 +0000 (13:56 -0500)]
Merge "cli: Display correct unit for HTTP timeout in "manager show settings"."

15 months agoMerge "Fix GCC 8 build issues."
Joshua Colp [Wed, 16 May 2018 18:56:34 +0000 (13:56 -0500)]
Merge "Fix GCC 8 build issues."

15 months agoMerge "rtp_engine: Remove the double assigned RTP payload ID of H.263+."
Joshua Colp [Tue, 15 May 2018 09:06:39 +0000 (04:06 -0500)]
Merge "rtp_engine: Remove the double assigned RTP payload ID of H.263+."

15 months agoMerge "git: Ignore *.orig."
Joshua Colp [Mon, 14 May 2018 11:56:08 +0000 (06:56 -0500)]
Merge "git: Ignore *.orig."

15 months agoMerge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code."
Joshua Colp [Mon, 14 May 2018 11:30:44 +0000 (06:30 -0500)]
Merge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code."

15 months agoMerge "pjsip: Rewrite OPTIONS support with new eyes."
Joshua Colp [Mon, 14 May 2018 09:06:53 +0000 (04:06 -0500)]
Merge "pjsip: Rewrite OPTIONS support with new eyes."

15 months agortp_engine: Remove the double assigned RTP payload ID of H.263+.
Alexander Traud [Fri, 11 May 2018 17:49:12 +0000 (19:49 +0200)]
rtp_engine: Remove the double assigned RTP payload ID of H.263+.

Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format
H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8
assigned another payload ID 98 for this format in Asterisk 1.6.

Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667

15 months agocli: Display correct unit for HTTP timeout in "manager show settings".
Corey Farrell [Fri, 11 May 2018 17:26:39 +0000 (13:26 -0400)]
cli: Display correct unit for HTTP timeout in "manager show settings".

HTTP timeout is in seconds, not minutes.

ASTERISK-27852 #close

Change-Id: Ie6640835cb07307555741f9b559c2eb876d9343e

15 months agortp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code.
Alexander Traud [Fri, 11 May 2018 15:37:57 +0000 (17:37 +0200)]
rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code.

Change-Id: Ica089d4507a27ddfc4ce3a88d697ffbef378de48

15 months agoFix GCC 8 build issues.
Corey Farrell [Mon, 7 May 2018 02:17:34 +0000 (22:17 -0400)]
Fix GCC 8 build issues.

This fixes build warnings found by GCC 8.  In some cases format
truncation is intentional so the warning is just suppressed.

ASTERISK-27824 #close

Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84

15 months agortp_engine: Allow Media Formats with add_static_payload(-1) on egress again.
Alexander Traud [Fri, 11 May 2018 12:10:51 +0000 (14:10 +0200)]
rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again.

This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf
which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples
of media formats which were affected.

ASTERISK-27850
Reported by: Dinis Brazão, Selene Feigl

Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496

15 months agoMerge "makeopts.in: Remove unused/undefined AST_MARCH_NATIVE."
Joshua Colp [Thu, 10 May 2018 08:44:39 +0000 (03:44 -0500)]
Merge "makeopts.in: Remove unused/undefined AST_MARCH_NATIVE."

15 months agoMerge "sip_to_pjsip: Enable python3 compatibility."
Joshua Colp [Thu, 10 May 2018 00:25:55 +0000 (19:25 -0500)]
Merge "sip_to_pjsip: Enable python3 compatibility."

15 months agoMerge "res_hep: Adds hostname resolution support for capture_address"
Joshua Colp [Thu, 10 May 2018 00:00:41 +0000 (19:00 -0500)]
Merge "res_hep: Adds hostname resolution support for capture_address"

15 months agoMerge "app_macro: Prevent infinite loop in find_matching_priority."
Jenkins2 [Wed, 9 May 2018 16:32:30 +0000 (11:32 -0500)]
Merge "app_macro: Prevent infinite loop in find_matching_priority."

15 months agogit: Ignore *.orig.
Corey Farrell [Wed, 9 May 2018 14:30:41 +0000 (10:30 -0400)]
git: Ignore *.orig.

This prevents accidental commit of files created by patch.

Change-Id: I68380db61f0f9d620046f719ccd978811d0e9964

15 months agosip_to_pjsip: Enable python3 compatibility.
Alexander Traud [Wed, 18 Apr 2018 07:27:51 +0000 (09:27 +0200)]
sip_to_pjsip: Enable python3 compatibility.

The script remains compatible with Python 2.7 but now also works with
Python 3.3 and newer; to ease the migration from chan_sip to chan_pjsip.

ASTERISK-27811

Change-Id: I59cc6b52a1a89777eebcf25b3023bdf93babf835

15 months agomakeopts.in: Remove unused/undefined AST_MARCH_NATIVE.
Corey Farrell [Tue, 8 May 2018 19:28:10 +0000 (15:28 -0400)]
makeopts.in: Remove unused/undefined AST_MARCH_NATIVE.

Change-Id: I617a96ebb83ec99f5d3176bbbee2d2a272ccb203

15 months agomanager: fix digest auth for ami/http mechanism.
Jaco Kroon [Tue, 8 May 2018 09:59:02 +0000 (11:59 +0200)]
manager: fix digest auth for ami/http mechanism.

Due to a fixed size buffer the digest authentication could be
incorrectly calculated if a large URI was provided, causing
authentication failure. The buffer is now dynamically allocated to allow
any size URI within the normal limits of the HTTP request size.

ASTERISK-27841

Change-Id: I660609db13b8f9e5f9567f339dd804f4985d41b3

15 months agoMerge "stream: Make the topology a reference counted object."
Jenkins2 [Tue, 8 May 2018 10:42:53 +0000 (05:42 -0500)]
Merge "stream: Make the topology a reference counted object."

15 months agoapp_macro: Prevent infinite loop in find_matching_priority.
Corey Farrell [Fri, 4 May 2018 18:47:25 +0000 (14:47 -0400)]
app_macro: Prevent infinite loop in find_matching_priority.

Use AST_PBX_MAX_STACK to escape if we recurse 128 times.  This will
prevent crash if dialplan contains an include loop.  Log an error when
this occurs, at most one message per call to Macro() so we avoid logger
spam.

ASTERISK-26570 #close

Change-Id: I6c71b76998c31434391b150de055ae9a531e31da

15 months agores_hep: Adds hostname resolution support for capture_address
Matthew Fredrickson [Fri, 4 May 2018 21:07:10 +0000 (16:07 -0500)]
res_hep: Adds hostname resolution support for capture_address

Previously, only an IP address would be accepted for the capture_address config
setting in hep.conf.  This change allows capture_address to be a resolvable
hostname or an IP address.

ASTERISK-27796 #close
Reported-By: Sebastian Gutierrez

Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f

15 months agoMerge "res_ari: Remove requirement that body exists when debug is on."
Jenkins2 [Fri, 4 May 2018 11:32:38 +0000 (06:32 -0500)]
Merge "res_ari: Remove requirement that body exists when debug is on."

15 months agoMerge "iostreams: Add some documentation for the ast_iostream_* functions"
Jenkins2 [Fri, 4 May 2018 11:14:56 +0000 (06:14 -0500)]
Merge "iostreams: Add some documentation for the ast_iostream_* functions"

15 months agoMerge "chan_dahdi: Configurable dialed digit timeouts"
Joshua Colp [Thu, 3 May 2018 17:07:14 +0000 (12:07 -0500)]
Merge "chan_dahdi: Configurable dialed digit timeouts"

15 months agoMerge "pbx_lua: Support displaying lua error message if no debug table exists"
Jenkins2 [Thu, 3 May 2018 16:41:35 +0000 (11:41 -0500)]
Merge "pbx_lua:  Support displaying lua error message if no debug table exists"

15 months agoMerge "res_pjsip/pjsip_distributor.c: Add missing off-nominal request response."
Jenkins2 [Thu, 3 May 2018 16:32:08 +0000 (11:32 -0500)]
Merge "res_pjsip/pjsip_distributor.c: Add missing off-nominal request response."

15 months agostream: Make the topology a reference counted object.
Joshua Colp [Fri, 20 Apr 2018 23:12:53 +0000 (23:12 +0000)]
stream: Make the topology a reference counted object.

The stream topology has no lock of its own resulting in
another lock protecting it in some way (for example the
channel lock). If multiple channels are being juggled at
the same time this can be problematic. This change makes
the topology a reference counted object instead which
guarantees it will remain valid even without the channel
lock being held.

Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03

15 months agoMerge "Add the ability to read the media file type from HTTP header for playback"
Jenkins2 [Thu, 3 May 2018 15:50:51 +0000 (10:50 -0500)]
Merge "Add the ability to read the media file type from HTTP header for playback"

15 months agochan_dahdi: Configurable dialed digit timeouts
Tzafrir Cohen [Wed, 21 Mar 2018 12:30:18 +0000 (14:30 +0200)]
chan_dahdi: Configurable dialed digit timeouts

Analog phones dial overlap dialing and it is chan_dahdi's job to read the
numbers.  It has three timeout constants that this commit converts to
channel-level configuration options:

* firstdigit_timeout: Default time (ms) to detect first digit

* interdigit_timeout: Default time (ms) to detect following digits

* matchdigit_timeout: Default time (ms) to wait in case of ambiguous
match.  This happens when the dialed digits match a number in the current
context but are also the prefix of another number.

Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213

15 months agoMerge "res_rtp_asterisk: Always update SRTP on local SSRC change."
Jenkins2 [Thu, 3 May 2018 15:29:38 +0000 (10:29 -0500)]
Merge "res_rtp_asterisk: Always update SRTP on local SSRC change."

15 months agoMerge "pjsip: Increase maximum number of usable ciphers & other cleanups"
Jenkins2 [Thu, 3 May 2018 12:41:33 +0000 (07:41 -0500)]
Merge "pjsip: Increase maximum number of usable ciphers & other cleanups"

15 months agores_ari: Remove requirement that body exists when debug is on.
Joshua Colp [Thu, 3 May 2018 11:34:32 +0000 (11:34 +0000)]
res_ari: Remove requirement that body exists when debug is on.

The "ari set debug" code for incoming requests incorrectly assumed
that all requests would contain a body. If one did not exist the
request would be incorrectly rejected. The response that was sent
was also incomplete as an incorrect function was used to construct
the response.

The code has now been changed to no longer require a request to have
a body and the response updated to use the correct function.

ASTERISK-27801

Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8

15 months agoiostreams: Add some documentation for the ast_iostream_* functions
Sean Bright [Mon, 30 Apr 2018 20:15:49 +0000 (16:15 -0400)]
iostreams: Add some documentation for the ast_iostream_* functions

Change-Id: Id71b87637f0a484eb5a1cd26c3d1c7c15c7dcf26

15 months agopjsip: Increase maximum number of usable ciphers & other cleanups
Sean Bright [Wed, 2 May 2018 12:43:35 +0000 (08:43 -0400)]
pjsip: Increase maximum number of usable ciphers & other cleanups

* Increase maximum number of ciphers from 100 to 256 (or whatever
  PJ_SSL_SOCK_MAX_CIPHERS is #define'd to)

* Simplify logic in cipher_name_to_id()

* Make signed/unsigned comparison consistent

Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412

Reported by: Ondřej Holas

Change-Id: Iea620f03915a1b873e79743154255c3148a514e7

15 months agores_pjsip/pjsip_distributor.c: Add missing off-nominal request response.
Richard Mudgett [Mon, 30 Apr 2018 22:24:33 +0000 (17:24 -0500)]
res_pjsip/pjsip_distributor.c: Add missing off-nominal request response.

Change-Id: I389579b39c523d1d1e8ce020ef549a8bb5781c9b

15 months agores_pjsip/pjsip_distributor.c: Pull some assignments out of if tests.
Richard Mudgett [Mon, 30 Apr 2018 22:20:13 +0000 (17:20 -0500)]
res_pjsip/pjsip_distributor.c: Pull some assignments out of if tests.

Change-Id: I3d30d638b53a4bbe9bf9aad853c649d583894112

15 months agores_rtp_asterisk: Always update SRTP on local SSRC change.
Joshua Colp [Mon, 30 Apr 2018 14:38:36 +0000 (14:38 +0000)]
res_rtp_asterisk: Always update SRTP on local SSRC change.

When the local SSRC changes we need to update the SRTP information
so that the proper key is used. This is commonly done as a result
of bridging two channels together. Previously we only updated
the SRTP information if media had already flowed, but in practice
the channel driver may have already performed SRTP negotiation and
set up the previous SSRC. We now always do it on a local SSRC
change.

ASTERISK-27795
ASTERISK-27800

Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10

15 months agoAdd the ability to read the media file type from HTTP header for playback
Gaurav Khurana [Tue, 13 Feb 2018 18:55:47 +0000 (10:55 -0800)]
Add the ability to read the media file type from HTTP header for playback

How it works today:
media_cache tries to parse out the extension of the media file to be played
from the URI provided to Asterisk while caching the file.

What's expected:
Better will be to have Asterisk get extension from other ways too. One of the
common ways is to get the type of content from the CONTENT-TYPE header in the
HTTP response for fetching the media file using the URI provided.

Steps to Reproduce:
Provide a URL of the form: http://host/media/1234 to Asterisk for media
playback. It fails to play and logs show the following error line:

[Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c:
File http://host/media/1234 does not exist in any format

Scenario this issue is blocking:
In the case where the media files are stored in some cloud object store,
following can block the media being played via Asterisk:

Cloud storage generally needs authenticated access to the storage. The way
to do that is by using signed URIs. With the signed URIs there's no way to
preserve the name of the file.
In most cases Cloud storage returns a key to access the object and preserving
file name is also not a thing there

ASTERISK-27286

 Reporter: Gaurav Khurana

Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89

15 months agoMerge "bridge_softmix: Fix sporadic incorrect video stream mapping."
Jenkins2 [Mon, 30 Apr 2018 17:35:10 +0000 (12:35 -0500)]
Merge "bridge_softmix: Fix sporadic incorrect video stream mapping."

15 months agoMerge "BuildSystem: Add DragonFly BSD."
George Joseph [Mon, 30 Apr 2018 14:07:30 +0000 (09:07 -0500)]
Merge "BuildSystem: Add DragonFly BSD."

15 months agoMerge "translate: generic plc not filled in after translation"
Jenkins2 [Mon, 30 Apr 2018 13:33:09 +0000 (08:33 -0500)]
Merge "translate: generic plc not filled in after translation"

15 months agoMerge "app_sendtext: Enhance SendText to support Enhanced Messaging"
George Joseph [Mon, 30 Apr 2018 12:34:32 +0000 (07:34 -0500)]
Merge "app_sendtext:  Enhance SendText to support Enhanced Messaging"

15 months agopbx_lua: Support displaying lua error message if no debug table exists
Christof Lauber [Wed, 25 Apr 2018 06:57:39 +0000 (08:57 +0200)]
pbx_lua:  Support displaying lua error message if no debug table exists

The lua_error_function assumed that lua's debug table and traceback function
are always accessible, which is not the case. This fixes the error message
'Error in the lua error handler' triggred by switch exec() function.
If this happens lua's error message is shown without traceback.

Change-Id: I34ba0a098f1ae06a3af7b4d1b098bd43f42f96c8

15 months agopjsip: Rewrite OPTIONS support with new eyes.
Joshua Colp [Mon, 11 Dec 2017 18:34:53 +0000 (18:34 +0000)]
pjsip: Rewrite OPTIONS support with new eyes.

The OPTIONS support in PJSIP has organically grown, like many things in
Asterisk.  It has been tweaked, changed, and adapted based on situations
run into.  Unfortunately this has taken its toll.  Configuration file
based objects have poor performance and even dynamic ones aren't that
great.

This change scraps the existing code and starts fresh with new eyes.  It
leverages all of the APIs made available such as sorcery observers and
serializers to provide a better implementation.

1.  The state of contacts, AORs, and endpoints relevant to the qualify
process is maintained.  This state can be updated by external forces (such
as a device registering/unregistering) and also the reload process.  This
state also includes the association between endpoints and AORs.

2.  AORs are scheduled and not contacts.  This reduces the amount of work
spent juggling scheduled items.

3.  Manipulation of which AORs are being qualified and the endpoint states
all occur within a serializer to reduce the conflict that can occur with
multiple threads attempting to modify things.

4.  Operations regarding an AOR use a serializer specific to that AOR.

5.  AORs and endpoint state act as state compositors.  They take input
from lower level objects (contacts feed AORs, AORs feed endpoint state)
and determine if a sufficient enough change has occurred to be fed further
up the chain.

6.  Realtime is supported by using observers to know when a contact has
been registered.  If state does not exist for the associated AOR then it
is retrieved and becomes active as appropriate.

The end result of all of this is best shown with a configuration file of
3000 endpoints each with an AOR that has a static contact.  In the old
code it would take over a minute to load and use all 8 of my cores.  This
new code takes 2-3 seconds and barely touches the CPU even while dealing
with all of the OPTIONS requests.

ASTERISK-26806

Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082

15 months agoMerge "bridge_softmix: Forward TEXT frames"
Jenkins2 [Fri, 27 Apr 2018 15:06:30 +0000 (10:06 -0500)]
Merge "bridge_softmix:  Forward TEXT frames"

15 months agoMerge "BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD."
Joshua Colp [Fri, 27 Apr 2018 00:09:16 +0000 (19:09 -0500)]
Merge "BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD."

15 months agoMerge "chan_ooh323: introduce localras config parameter"
Jenkins2 [Thu, 26 Apr 2018 16:42:06 +0000 (11:42 -0500)]
Merge "chan_ooh323: introduce localras config parameter"

15 months agoMerge "core: Remove unused/incomplete SDP modules."
Jenkins2 [Wed, 25 Apr 2018 20:49:25 +0000 (15:49 -0500)]
Merge "core: Remove unused/incomplete SDP modules."

15 months agoMerge "install_prereq: Add DragonFly BSD."
Jenkins2 [Wed, 25 Apr 2018 19:01:43 +0000 (14:01 -0500)]
Merge "install_prereq: Add DragonFly BSD."

15 months agocore: Remove unused/incomplete SDP modules.
Richard Mudgett [Fri, 22 Dec 2017 19:11:53 +0000 (13:11 -0600)]
core: Remove unused/incomplete SDP modules.

Change-Id: Icc28fbdc46f58e54a21554e6fe8b078f841b1f86

15 months agoMerge "streams: Add string metadata capability"
Joshua Colp [Wed, 25 Apr 2018 18:45:26 +0000 (13:45 -0500)]
Merge "streams: Add string metadata capability"

15 months agoMerge "menuselect: Add DragonFly BSD."
Jenkins2 [Wed, 25 Apr 2018 18:31:44 +0000 (13:31 -0500)]
Merge "menuselect: Add DragonFly BSD."

15 months agoMerge "chan_ooh323: fix ooManualProgress/ooManualRingback on ooh323 debuggin on"
Richard Mudgett [Wed, 25 Apr 2018 15:46:05 +0000 (10:46 -0500)]
Merge "chan_ooh323: fix ooManualProgress/ooManualRingback on ooh323 debuggin on"

15 months agoMerge "Build System: Add missing ASTMM_LIBC to flex output."
Jenkins2 [Wed, 25 Apr 2018 15:02:13 +0000 (10:02 -0500)]
Merge "Build System: Add missing ASTMM_LIBC to flex output."

15 months agoMerge "format_pcm: Correct behavior of fseek and ftell for G.722"
Jenkins2 [Wed, 25 Apr 2018 14:44:43 +0000 (09:44 -0500)]
Merge "format_pcm: Correct behavior of fseek and ftell for G.722"

15 months agoMerge "bridge_softmix: Fix some REMB bugs."
Joshua Colp [Wed, 25 Apr 2018 09:29:42 +0000 (04:29 -0500)]
Merge "bridge_softmix: Fix some REMB bugs."

15 months agoMerge "chan_ooh323: Fix cppcheck warnings"
Joshua Colp [Tue, 24 Apr 2018 23:54:36 +0000 (18:54 -0500)]
Merge "chan_ooh323: Fix cppcheck warnings"

15 months agotranslate: generic plc not filled in after translation
Kevin Harwell [Wed, 18 Apr 2018 20:59:38 +0000 (15:59 -0500)]
translate: generic plc not filled in after translation

If during translation a codec could not handle a given frame the translation
core would return NULL, thus not passing along the "missing" frame. Due to this
there was no frame to apply generic plc to, thus rendering it useless.

This patch makes it so the translation core produces an interpolated slin frame
in the cases where an attempt was made to translate to slin, but failed. This
interpolated frame is then passed along and can be used by the generic plc
algorithms to fill in the frame.

ASTERISK-27814 #close

Change-Id: I133d084da87adef913bf2ecc9c9240e3eaf4f40a

16 months agobridge_softmix: Fix sporadic incorrect video stream mapping.
Joshua Colp [Fri, 20 Apr 2018 12:40:01 +0000 (12:40 +0000)]
bridge_softmix: Fix sporadic incorrect video stream mapping.

When an externally initiated renegotiation occurred it was
possible for video streams to be incorrectly remapped,
resulting in no video flowing to some receivers.

This change ensures that only the video source sets up
mappings and also that removed streams do not have mappings
set up.

Change-Id: Iab05f2254df3606670774844bb0935f833d3a9b0

16 months agochan_ooh323: fix ooManualProgress/ooManualRingback on ooh323 debuggin on
Alexander Anikin [Fri, 20 Apr 2018 19:07:21 +0000 (22:07 +0300)]
chan_ooh323: fix ooManualProgress/ooManualRingback on ooh323 debuggin on

Call ooManualProgress/Ringback outside of ast_debug function
when ooh323 debugging is on

ASTERISK-27812 #close
ASTERISK-26893 #close
Reported by: Dimos, Marco Giordani

Change-Id: I5873762e4f05824e7b6e94a19dd4eb56adbbbb79

16 months agobridge_softmix: Fix some REMB bugs.
Joshua Colp [Thu, 19 Apr 2018 18:44:03 +0000 (18:44 +0000)]
bridge_softmix: Fix some REMB bugs.

This change fixes a bug where a REMB collector may be
freed twice, and also tweaks REMB combining such that if
there is no bitrate from anyone (or there are no sources)
we report 0 instead of using an old bitrate.

ASTERISK-27804

Change-Id: Ia9dc9c150043890ee7ff85e9cdec007f1a77fcfd

16 months agoBuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD.
Alexander Traud [Fri, 20 Apr 2018 12:13:12 +0000 (14:13 +0200)]
BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD.

ASTERISK-27639

Change-Id: I1347f3f2f3737010d0a80a5c30b5aaf71cf3ccb0

16 months agoBuildSystem: Add DragonFly BSD.
Alexander Traud [Fri, 20 Apr 2018 10:50:03 +0000 (12:50 +0200)]
BuildSystem: Add DragonFly BSD.

ASTERISK-27820

Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557

16 months agomenuselect: Add DragonFly BSD.
Alexander Traud [Fri, 20 Apr 2018 10:40:21 +0000 (12:40 +0200)]
menuselect: Add DragonFly BSD.

In DragonFly BSD, added libraries from ports are placed into /usr/local.
Therefore, this directory must be added for the preprocessor, compiler, and
linker.

Beside that, the script ./configure was updated:
* OSARCH list was outdated and not used, removed.
* AC_CANONICAL_BUILD was not used.
* _REENTRANT, this feature test macro is obsolete.

ASTERISK-27820

Change-Id: I186d88d99cfa4de6569888e12ac97bd2f441c422

16 months agoinstall_prereq: Add DragonFly BSD.
Alexander Traud [Fri, 20 Apr 2018 10:18:06 +0000 (12:18 +0200)]
install_prereq: Add DragonFly BSD.

ASTERISK-27820

Change-Id: I718ddb000fe5184b1bdc7759da67a370a7520144

16 months agoMerge ""confbridge show profile bridge" does not output "sfu" when video_mode is...
Jenkins2 [Thu, 19 Apr 2018 21:14:37 +0000 (16:14 -0500)]
Merge ""confbridge show profile bridge" does not output "sfu" when video_mode is sfu"

16 months agoMerge "chan_vpb: Avoid GNU old-style field designator extension."
Jenkins2 [Thu, 19 Apr 2018 19:40:53 +0000 (14:40 -0500)]
Merge "chan_vpb: Avoid GNU old-style field designator extension."