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is no longer valid.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

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


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.


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 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'.


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.


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.


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.


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."

23 months 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."

23 months 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."

23 months 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."

23 months 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

23 months 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

23 months 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

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

Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da

23 months 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

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

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


Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f

23 months 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

23 months 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

23 months 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

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

Change-Id: I56ed530633a642633b18383821069e806c92ae82

23 months 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

23 months 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

23 months 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

23 months 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

23 months 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

23 months 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

23 months 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

23 months 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."

23 months 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."

23 months 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"

23 months 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."

23 months 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."

23 months 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."

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

23 months 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

23 months 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"

23 months 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

23 months 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."

23 months 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"

23 months 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"

23 months 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

23 months 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

23 months 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

23 months 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

23 months 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

23 months 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

23 months 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"

23 months 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"

23 months 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."

23 months 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

23 months 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

23 months 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"

23 months 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"

23 months 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."

23 months 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."

23 months 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

2 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

2 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"

2 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."

2 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."

2 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"

2 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."

2 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

2 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"

2 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"

2 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"

2 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

2 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

2 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"

2 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

2 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"

2 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

2 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

2 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

2 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."

2 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

2 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

2 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

2 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

2 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

2 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

2 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

2 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"