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 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_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 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 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."

3 years agoMerge "res_pjsip: Add "webrtc" configuration option"
Jenkins2 [Mon, 17 Jul 2017 20:16:30 +0000 (15:16 -0500)]
Merge "res_pjsip: Add "webrtc" configuration option"

3 years agoMerge "res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use."
Jenkins2 [Mon, 17 Jul 2017 19:54:22 +0000 (14:54 -0500)]
Merge "res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use."

3 years agobridge_softmix: Don't reorder streams on participant leaving.
Joshua Colp [Mon, 17 Jul 2017 12:19:59 +0000 (12:19 +0000)]
bridge_softmix: Don't reorder streams on participant leaving.

When a participant leaves a bridge while operating in SFU mode
their respective stream on every other participant needs to be
removed. Leaving the stream out of the new topology results in
every stream after it being moved and reordered. This causes
problems with clients. Instead simply mark the stream as removed
which leaves it in place in the SDP and doesn't reorder or touch
any other streams.


Change-Id: I4b3f840adcdf69b83842b0d8a737665ba0ef9cb1

3 years agoMerge "app_confbridge: Make sure name recordings are always removed from the filesystem"
Jenkins2 [Mon, 17 Jul 2017 14:35:08 +0000 (09:35 -0500)]
Merge "app_confbridge: Make sure name recordings are always removed from the filesystem"

3 years agoMerge "chan_iax2: On reload make sure to check for existing MWI subscription"
George Joseph [Mon, 17 Jul 2017 14:03:14 +0000 (09:03 -0500)]
Merge "chan_iax2: On reload make sure to check for existing MWI subscription"

3 years agoMerge "res/res_stasis_snoop: generate silence when audiohook returns null"
Jenkins2 [Mon, 17 Jul 2017 13:25:29 +0000 (08:25 -0500)]
Merge "res/res_stasis_snoop: generate silence when audiohook returns null"

3 years agobridge_softmix: Use removed stream spots when renegotiating.
Joshua Colp [Sun, 16 Jul 2017 17:31:35 +0000 (17:31 +0000)]
bridge_softmix: Use removed stream spots when renegotiating.

Streams are never truly removed in SDP, they still occupy
a location within the SDP. This location can be reused by
another stream if it so chooses.

This change takes advantage of this such that if a new stream
is needing to be added for a new participant any removed streams
are instead replaced first. This reduces the size of the SDP
and the number of streams.


Change-Id: I95cdcfd55cf47e02ea52abb5d94008db3fb68b1d

3 years agores_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use.
Joshua Colp [Sun, 16 Jul 2017 17:18:39 +0000 (17:18 +0000)]
res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use.

This change makes it so that if an RTCP packet is being sent
the RTP ICE component is used for sending if RTCP-MUX is in use.


Change-Id: I6200f611ede709602ee9b89501720c29545ed68b

3 years agoMerge "Sounds: Update for core sounds 1.6 release"
Jenkins2 [Fri, 14 Jul 2017 15:56:17 +0000 (10:56 -0500)]
Merge "Sounds: Update for core sounds 1.6 release"

3 years agoapp_confbridge: Make sure name recordings are always removed from the filesystem
Sergej Kasumovic [Fri, 14 Jul 2017 06:25:36 +0000 (08:25 +0200)]
app_confbridge: Make sure name recordings are always removed from the filesystem

This commit fixes two possible scenarios:

* When recording name and if during recording you hangup, file is never
removed. This is due to the fact file location is nulled.
* When recording name and if you hangup during thank-you prompt, file
is never removed.

ASTERISK-27123 #close

Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625

3 years agoMerge "res/res_pjsip_t38 ensure t38 requests get rejected quickly"
George Joseph [Fri, 14 Jul 2017 12:44:20 +0000 (07:44 -0500)]
Merge "res/res_pjsip_t38  ensure t38 requests get rejected quickly"

3 years agochan_iax2: On reload make sure to check for existing MWI subscription
Sergej Kasumovic [Fri, 14 Jul 2017 06:11:50 +0000 (08:11 +0200)]
chan_iax2: On reload make sure to check for existing MWI subscription

On every reload of chan_iax2 module, MWI subscription was added, which
results in additional taskprocessors being accumulated over time.

This commit fixes it by making sure we check for existing subscription

This was verified with 'core show taskprocessors' CLI command.

ASTERISK-27122 #close

Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9

3 years agores_pjsip: Add "webrtc" configuration option
Kevin Harwell [Mon, 10 Jul 2017 23:17:44 +0000 (18:17 -0500)]
res_pjsip: Add "webrtc" configuration option

This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:

  rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled

When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.

Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.

ASTERISK-27119 #close

Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd

3 years agoSounds: Update for core sounds 1.6 release
Rusty Newton [Thu, 13 Jul 2017 20:43:42 +0000 (15:43 -0500)]
Sounds: Update for core sounds 1.6 release

Added necessary lines to make the en_NZ language set selectable and to get
core sounds 1.6 pulled down.

ASTERISK-26807 #close
ASTERISK-25816 #close
ASTERISK-26274 #close

Change-Id: I84e4dd4696568cc1ba318d12ac4b075461d6eed4

3 years agoMerge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."
Jenkins2 [Thu, 13 Jul 2017 19:40:11 +0000 (14:40 -0500)]
Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."

3 years agocore: Add PARSE_TIMELEN support to ast_parse_arg and ACO.
Corey Farrell [Mon, 10 Jul 2017 19:04:58 +0000 (15:04 -0400)]
core: Add PARSE_TIMELEN support to ast_parse_arg and ACO.

This adds support for parsing timelen values from config files.  This
includes support for all flags which apply to PARSE_INT32.  Support for
this parser is added to ACO via the OPT_TIMELEN_T option type.

Fixes an issue where extra characters provided to ast_app_parse_timelen
were ignored, they now cause an error.

Testing is included.

ASTERISK-27117 #close

Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554

3 years agores_rtp_asterisk / res_pjsip: Add support for BUNDLE.
Joshua Colp [Fri, 30 Jun 2017 18:55:57 +0000 (18:55 +0000)]
res_rtp_asterisk / res_pjsip: Add support for BUNDLE.

BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.

This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.

For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.


Change-Id: I96c0920b9f9aca7382256484765a239017973c11

3 years agores/res_stasis_snoop: generate silence when audiohook returns null
Torrey Searle [Tue, 11 Jul 2017 14:55:36 +0000 (16:55 +0200)]
res/res_stasis_snoop: generate silence when audiohook returns null

Currently when rtp is paused, no packets are written to the
recorded audio file, causing the silence to be skipped and recording
not properly time aligned.  The read handler as been adapted to
return a silence frame of the correct size.

ASTERISK-27128 #close

Change-Id: I2d7f60650457860b9c70907b14426756b058a844

3 years agores/res_pjsip_t38 ensure t38 requests get rejected quickly
Torrey Searle [Thu, 22 Jun 2017 12:47:54 +0000 (14:47 +0200)]
res/res_pjsip_t38  ensure t38 requests get rejected quickly

arm the t38 webhook always, so we can correctly reject a
T38 negotiation request when t38 is disabled on a channel

Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d

3 years agocore: Add digit filtering to ast_waitfordigit_full
Corey Farrell [Wed, 12 Jul 2017 18:24:36 +0000 (14:24 -0400)]
core: Add digit filtering to ast_waitfordigit_full

This adds a parameter to ast_waitfordigit_full which can be used to only
stop waiting when certain expected digits are received.  Any unexpected
DTMF digits are simply ignored.

This also creates a new dialplan application WaitDigit.

ASTERISK-27129 #close

Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9

3 years agoapp_playback.c: Use the timezonename parameter
Holger Hans Peter Freyther [Tue, 11 Jul 2017 09:48:26 +0000 (11:48 +0200)]
app_playback.c: Use the timezonename parameter

In say_date_generic the timezonename parameter is passed but never
used. Fix it by passing it to the ast_localtime function.


Change-Id: I63106b8db10426d417d7275f22554a616e92fae4

3 years agobasic-pbx: Remove res_pjsip_multihomed from sample config
Sean Bright [Wed, 12 Jul 2017 20:07:42 +0000 (16:07 -0400)]
basic-pbx: Remove res_pjsip_multihomed from sample config

ASTERISK-27127 #close
Reported by: HZMI8gkCvPpom0tM

Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789

3 years agoMerge "app_stream_echo: misc bug fixes"
Joshua Colp [Wed, 12 Jul 2017 11:13:34 +0000 (06:13 -0500)]
Merge "app_stream_echo: misc bug fixes"

3 years agoMerge "res_rtp_asterisk: trigger source change control frame when dtls is established"
Joshua Colp [Wed, 12 Jul 2017 11:13:25 +0000 (06:13 -0500)]
Merge "res_rtp_asterisk: trigger source change control frame when dtls is established"

3 years agoMerge "res_musiconhold: Add kill_escalation_delay, kill_method to class"
Joshua Colp [Wed, 12 Jul 2017 10:48:01 +0000 (05:48 -0500)]
Merge "res_musiconhold:  Add kill_escalation_delay, kill_method to class"

3 years agoMerge "manager: Remove AMI "Queues" action."
Joshua Colp [Wed, 12 Jul 2017 09:25:45 +0000 (04:25 -0500)]
Merge "manager: Remove AMI "Queues" action."

3 years agoMerge "Avoid setting maxfiles for a remote asterisk"
Joshua Colp [Wed, 12 Jul 2017 09:24:43 +0000 (04:24 -0500)]
Merge "Avoid setting maxfiles for a remote asterisk"

3 years agoMerge "http.c: Reduce log spam"
Jenkins2 [Wed, 12 Jul 2017 00:42:10 +0000 (19:42 -0500)]
Merge "http.c:  Reduce log spam"

3 years agobridge/core_unreal: Fix SFU bugs with forwarding frames.
Joshua Colp [Tue, 11 Jul 2017 19:33:44 +0000 (19:33 +0000)]
bridge/core_unreal: Fix SFU bugs with forwarding frames.

This change fixes a few things uncovered during SFU testing.

1. Unreal channels incorrectly forwarded video frames when
no video stream was present on them. This caused a crash when
they were read as the core requires a stream to exist for the
underlying media type. The Unreal channel will now ensure a
stream exists for the media type before forwarding the frame
and if no stream exists then the frame is dropped.

2. Mapping of frames during bridging from the stream number of
the underlying channel to the stream number of the bridge was
done in the wrong location. This resulted in the frame getting
dropped. This mapping now occurs on reading of the frame from
the channel.

3. Bridging was using the wrong ast_read function resulting in
it living in a non-multistream world.

4. In bridge_softmix when adding new streams to existing channels
the wrong stream topology was copied resulting in no streams
being added.

Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8

3 years agores_musiconhold: Add kill_escalation_delay, kill_method to class
George Joseph [Tue, 11 Jul 2017 12:26:27 +0000 (06:26 -0600)]
res_musiconhold:  Add kill_escalation_delay, kill_method to class

By default, when res_musiconhold reloads or unloads, it sends a HUP
signal to custom applications (and all descendants), waits 100ms,
then sends a TERM signal, waits 100ms, then finally sends a KILL
signal.  An application which is interacting with an external
device and/or spawns children of its own may not be able to exit
cleanly in the default times, expecially if sent a KILL signal, or
if it's children are getting signals directly from

* To allow extra time, the 'kill_escalation_delay'
  class option can be used to set the number of milliseconds
  res_musiconhold waits before escalating kill signals, with the
  default being the current 100ms.

* To control to whom the signals are sent, the "kill_method" class
  option can be set to "process_group" (the default, existing
  behavior), which sends signals to the application and its
  descendants directly, or "process" which sends signals only to the
  application itself.

Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b

3 years agomanager: Remove AMI "Queues" action.
Benjamin Keith Ford [Wed, 5 Jul 2017 17:44:18 +0000 (12:44 -0500)]
manager: Remove AMI "Queues" action.

When performing the "Queues" action via AMI, it outputs the same
text that the Asterisk CLI outputs when running a "queue show"
command, which does not conform with the AMI spec. "QueueStatus"
already does what the "Queues" action should do, so instead of
correcting the output, the "Queues" action will be removed and
"QueueStatus" should be used instead.

ASTERISK-27073 #close
Reported by: Brian

Change-Id: Id11743859758255b69cc3a557750d7a56c6d16f8

3 years agoAvoid setting maxfiles for a remote asterisk
Tzafrir Cohen [Mon, 3 Jul 2017 12:30:37 +0000 (15:30 +0300)]
Avoid setting maxfiles for a remote asterisk

Setting maxfiles (maximum number of open files) has no practical
effect on a remote asterisk (rasterisk, rasterisk -x).

It has an ill effect of printing an extra message, which
may be annoying in case of -x.

ASTERISK-27105 #close

Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2

3 years agohttp.c: Reduce log spam
George Joseph [Wed, 5 Jul 2017 20:31:43 +0000 (14:31 -0600)]
http.c:  Reduce log spam

Messages like "fwrite() failed: Connection reset by peer" are no
help whatsoever, especially since they can be caused simply by a
client disconnecting.

* Make those WARNINGs DEBUGs.
* Check the return from ast_iostream_printf of headers.

Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b

3 years agoMerge "res_pjsip: Fix crash with from_user containing invalid characters."
Jenkins2 [Tue, 11 Jul 2017 12:08:39 +0000 (07:08 -0500)]
Merge "res_pjsip: Fix crash with from_user containing invalid characters."

3 years agoMerge "json.c: Add backtrace log to find 'Invalid UTF-8 string' errors"
Jenkins2 [Mon, 10 Jul 2017 16:41:17 +0000 (11:41 -0500)]
Merge "json.c: Add backtrace log to find 'Invalid UTF-8 string' errors"

3 years agoMerge "res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock."
Jenkins2 [Mon, 10 Jul 2017 16:19:16 +0000 (11:19 -0500)]
Merge "res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock."

3 years agoMerge "bridge_native_rtp.c: Fix direct media video RTP instance ACL check."
Jenkins2 [Mon, 10 Jul 2017 15:53:11 +0000 (10:53 -0500)]
Merge "bridge_native_rtp.c: Fix direct media video RTP instance ACL check."

3 years agores_pjsip: Fix crash with from_user containing invalid characters.
Benjamin Keith Ford [Fri, 7 Jul 2017 16:19:13 +0000 (11:19 -0500)]
res_pjsip: Fix crash with from_user containing invalid characters.

If the from_user field contains certain characters (like @, {, ^, etc.),
PJSIP will return a null value for the URI when attempting to parse it.
This causes a crash when trying to dial out through a trunk that contains
these invalid characters in its from_user field.

This change checks the configuration and ensures that an endpoint will
not be created if the from_user contains an invalid character. It also
adds a null check to the PJSIP URI parsing as a backup.

ASTERISK-27036 #close
Reported by: Maxim Vasilev

Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0

3 years agoMerge "app_queue: Add priority to AMI QueueStatus"
George Joseph [Mon, 10 Jul 2017 14:50:37 +0000 (09:50 -0500)]
Merge "app_queue: Add priority to AMI QueueStatus"

3 years agojson.c: Add backtrace log to find 'Invalid UTF-8 string' errors
Richard Mudgett [Wed, 28 Jun 2017 00:27:43 +0000 (19:27 -0500)]
json.c: Add backtrace log to find 'Invalid UTF-8 string' errors

Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929

3 years agoMerge "app_voicemail: Cleanup ODBC connection handling"
Joshua Colp [Fri, 7 Jul 2017 21:38:21 +0000 (16:38 -0500)]
Merge "app_voicemail: Cleanup ODBC connection handling"

3 years agoMerge "core: Remove 'Data Retrieval API'"
Jenkins2 [Fri, 7 Jul 2017 20:42:56 +0000 (15:42 -0500)]
Merge "core: Remove 'Data Retrieval API'"

3 years agores_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock.
Richard Mudgett [Wed, 5 Jul 2017 18:39:45 +0000 (13:39 -0500)]
res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock.

When a message is received on the TURN socket, the code processing the
message needs to call into the ICE/STUN session for further processing.
This code path locks the TURN group lock then the ICE/STUN group lock.  In
another thread an ICE/STUN timer can fire off to send a keep alive message
over the TURN socket.  In this code path, the ICE/STUN group lock is
obtained then the TURN group lock is obtained to send the packet.  A
classic deadlock case if the group locks are not the same.

* Made TURN get created using the ICE/STUN session's group lock.

NOTE: I was originally concerned that the ICE/STUN session can get
recreated by ice_reset_session() for an event like RTCP multiplexing
causing a change during SDP negotiation.  In this case the TURN group lock
would become different.  However, TURN is also recreated as part of the
ICE/STUN recreation in ice_create() when all known ICE candidates are
added to the new ICE session.  While the ICE/STUN and TURN sessions are
being recreated there is a period where the group locks could be

ASTERISK-27023 #close
    res_rtp_asterisk-turn-deadlock-fix.patch (license #6502)
        patch uploaded by Michael Walton (modified)

Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9

3 years agoFix alembic branches
George Joseph [Thu, 6 Jul 2017 10:55:17 +0000 (04:55 -0600)]
Fix alembic branches

Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187

3 years agoMerge "channel: Clear channel flag in error branch."
Joshua Colp [Wed, 5 Jul 2017 23:46:10 +0000 (18:46 -0500)]
Merge "channel: Clear channel flag in error branch."

3 years agoMerge "pjproject_bundled: Allow passing configure options to bundled"
Jenkins2 [Wed, 5 Jul 2017 22:59:39 +0000 (17:59 -0500)]
Merge "pjproject_bundled:  Allow passing configure options to bundled"