asterisk/asterisk.git
9 months agochan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS
Dan Cropp [Tue, 2 Apr 2019 19:42:44 +0000 (14:42 -0500)]
chan_pjsip:  Transmit REFER waits for the REFER result setting TRANSFERSTATUS

Previously, when a Transfer (REFER) was performed, chan_pjsip would set
the TRANSFERSTATUS to SUCCESS when the REFER was queued up.  This did not
reflect a successful/unsuccessful transfer the way chan_sip did.
Added a callback module to process the refer subscription information.

Now depends on res_pjsip_pubsub so call transfer progress can be monitored
and reported

ASTERISK-26968 #close
Reported-by: Dan Cropp

Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc

9 months agoMerge "CI: New way to determnine libdir"
George Joseph [Tue, 25 Jun 2019 14:07:12 +0000 (09:07 -0500)]
Merge "CI:  New way to determnine libdir"

9 months agoMerge "res_fax: gateway sends T.38 request to both endpoints if V.21 detected"
George Joseph [Mon, 24 Jun 2019 20:16:36 +0000 (15:16 -0500)]
Merge "res_fax: gateway sends T.38 request to both endpoints if V.21 detected"

9 months agoMerge "translate.c do not log WARNING on empty audio frame"
George Joseph [Fri, 21 Jun 2019 18:41:29 +0000 (13:41 -0500)]
Merge "translate.c do not log WARNING on empty audio frame"

9 months agoMerge "app_confbridge: Attended transfer event fixup"
Friendly Automation [Fri, 21 Jun 2019 16:24:35 +0000 (11:24 -0500)]
Merge "app_confbridge:  Attended transfer event fixup"

9 months agores_fax: gateway sends T.38 request to both endpoints if V.21 detected
Alexei Gradinari [Wed, 29 May 2019 22:54:16 +0000 (18:54 -0400)]
res_fax: gateway sends T.38 request to both endpoints if V.21 detected

According T.38 Gateway 'Use case 3'
https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway
T.38 Gateway should send T.38 negotiation request to called endpoint
if FAX preamble (using V.21 detector) generated by called endpoint.
But it does not, because fax_gateway_detect_v21 constructs T.38
negotiation request, but forwards it only to other channel,
not to the channel on which FAX preamble is detected.

Some SIP endpoints could be improperly configured to rely on the other side
to initiate T.38 re-INVITEs.

With this patch the T.38 Gateway tries to negotiate with both sides
by sending T.38 negotiation request to both endpoints supported T.38.

Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39

9 months agoCI: New way to determnine libdir
George Joseph [Wed, 19 Jun 2019 16:58:39 +0000 (10:58 -0600)]
CI:  New way to determnine libdir

We were using the presence of /usr/lib64 to determine where
shared libraries should be installed.  This only existed on
Redhat based systems and was safe.  If it existed, use it,
otherwise use /usr/lib.

Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT
NOT INCLUDE IT IN THE DEFAULT ld.so.conf.  So if anything is
installed there, it won't work.

The new method, just looks for $ID in /etc/os-release and if it's
centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib.

NOTE:  This applies only to the CI scripts.  Normal asterisk
build and install is not affected.

Change-Id: Iad66374b550fd89349bedbbf2b93f8edd195a7c3

9 months agotranslate.c do not log WARNING on empty audio frame
Alexei Gradinari [Fri, 14 Jun 2019 20:45:39 +0000 (16:45 -0400)]
translate.c do not log WARNING on empty audio frame

There is WARNING "no samples for ..." on each Playtones.
The function ast_playtones_start calls ast_activate_generator,
which calls ast_prod.
The function ast_prod calls ast_write with empty audio frame.
In this case it's spam log.

Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660

9 months agochan_dahdi: Address gcc9 issues
George Joseph [Mon, 17 Jun 2019 17:11:49 +0000 (11:11 -0600)]
chan_dahdi:  Address gcc9 issues

Fixed format-truncation issues in chan_dahdi.c and
sig_analog.c.  Since they're related to fields provided
by dahdi-tools we can't change the buffer sizes so we're just
checking the return from snprintf and printing an errior if we
overflow.

Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5

9 months agoapp_confbridge: Attended transfer event fixup
George Joseph [Mon, 10 Jun 2019 21:58:59 +0000 (15:58 -0600)]
app_confbridge:  Attended transfer event fixup

When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.

Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1

9 months agores_rtp_asterisk: Add support for DTLS packet fragmentation.
Joshua Colp [Tue, 11 Jun 2019 12:26:42 +0000 (09:26 -0300)]
res_rtp_asterisk: Add support for DTLS packet fragmentation.

This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.

This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.

ASTERISK-28018

Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06

9 months agoMerge "app_attended_transfer: new application AttendedTransfer"
George Joseph [Wed, 12 Jun 2019 15:44:06 +0000 (10:44 -0500)]
Merge "app_attended_transfer: new application AttendedTransfer"

9 months agoMerge "app_blind_transfer: new application BlindTransfer"
Friendly Automation [Wed, 12 Jun 2019 14:31:36 +0000 (09:31 -0500)]
Merge "app_blind_transfer: new application BlindTransfer"

9 months agoMerge "chan_pjsip.c: Check for channel and session to not be NULL in hangup"
George Joseph [Wed, 12 Jun 2019 13:50:01 +0000 (08:50 -0500)]
Merge "chan_pjsip.c: Check for channel and session to not be NULL in hangup"

9 months agoapp_attended_transfer: new application AttendedTransfer
Alexei Gradinari [Tue, 21 May 2019 19:12:55 +0000 (15:12 -0400)]
app_attended_transfer: new application AttendedTransfer

AttendedTransfer queues up attended transfer to the given extension.

This application can be useful with Custom Dynamic Features.
For example to make attended transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_atxfer
TRANSFER_CONTEXT=my_transfer

[my_atxfer]
exten => s,1,AttendedTransfer(1234567890)
   same => n,Return()

[my_transfer]
include => default
;;;

This application also can be used to completly redefine Attended transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
atxfer => *7

[applicationmap]
custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_atxfer
TRANSFER_CONTEXT=my_transfer

[custom_atxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,AttendedTransfer(${dest})
   same => n,Return()

[my_transfer]
include => default
;;;

Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b

9 months agoMerge "cdr_pgsql: fix error in connection string"
Friendly Automation [Tue, 11 Jun 2019 13:03:09 +0000 (08:03 -0500)]
Merge "cdr_pgsql: fix error in connection string"

9 months agochan_pjsip.c: Check for channel and session to not be NULL in hangup
agupta [Thu, 6 Jun 2019 12:48:18 +0000 (18:18 +0530)]
chan_pjsip.c: Check for channel and session to not be NULL in hangup

We have seen some rare case of segmentation fault in hangup function
and we could notice that channel pointer was NULL.  Debug log shows
that there is a 200 OK answer and SIP timeout at the same time.  It
looks that while the SIP session was being destroyed due to timeout
call hangup due to answer event lead to race condition and channel
is being destroyed from two different places.  The check ensures we
check it not to be NULL before freeing it.

ASTERISK-25371

Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778

9 months agoapp_blind_transfer: new application BlindTransfer
Alexei Gradinari [Tue, 21 May 2019 19:53:47 +0000 (15:53 -0400)]
app_blind_transfer: new application BlindTransfer

BlindTransfer redirects all channels currently bridged to the
caller channel to the specified destination.

This application can be useful with Custom Dynamic Features.
For example to make blind transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_blindxfer

[my_blindxfer]
exten => s,1,BlindTransfer(1234567890,default)
   same => n,Return()
;;;

This application also can be used to completly redefine Blind transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
blindxfer =>

[applicationmap]
custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_blindxfer

[custom_blindxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,BlindTransfer(${dest},default)
   same => n,Return()
;;;

Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a

9 months agopbx_dundi: added IPv4/IPv6 dual bind support to DUNDi
Kirsty Tyerman [Tue, 8 Jan 2019 06:14:07 +0000 (16:14 +1000)]
pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi

ASTERISK-28234
Reported-by: Kirsty Tyerman

Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1

9 months agocdr_pgsql: fix error in connection string
Chris-Savinovich [Tue, 4 Jun 2019 17:41:33 +0000 (12:41 -0500)]
cdr_pgsql: fix error in connection string

Fixes an error occurring in function pgsql_reconnect() caused when value of
hostname is blank. Which in turn will cause the connection string to look
like this: "host= port=xx", which creates a sintax error. This fix now checks
if the corresponding values for host, port, dbname, and user are blank. Note
that since this is a reconnect function the database library will replace any
missing value pairs with default ones.

ASTERISK-28435

Change-Id: I0a921f99bbd265768be08cd492f04b30855b8423

9 months agoMerge "res_fax: fix segfault on inactive "reserved" fax session"
Friendly Automation [Tue, 4 Jun 2019 10:07:14 +0000 (05:07 -0500)]
Merge "res_fax: fix segfault on inactive "reserved" fax session"

9 months agoMerge "app_readexten: new option 'p' to stop reading on '#' key"
Friendly Automation [Mon, 3 Jun 2019 15:05:11 +0000 (10:05 -0500)]
Merge "app_readexten: new option 'p' to stop reading on '#' key"

9 months agoMerge "pjsip: replace 180 by 183 if SDP negotiation has completed"
George Joseph [Mon, 3 Jun 2019 14:36:43 +0000 (09:36 -0500)]
Merge "pjsip: replace 180 by 183 if SDP negotiation has completed"

9 months agoMerge "res_fax: add channel name to CLI 'fax show session'"
Friendly Automation [Mon, 3 Jun 2019 14:29:12 +0000 (09:29 -0500)]
Merge "res_fax: add channel name to CLI 'fax show session'"

9 months agores_fax: fix segfault on inactive "reserved" fax session
Alexei Gradinari [Tue, 28 May 2019 20:35:17 +0000 (16:35 -0400)]
res_fax: fix segfault on inactive "reserved" fax session

The change #10017 "Handle fax gateway being started more than once"
introdiced a bug which leads to segfault in res_fax_spandsp.

The res_fax_spandsp module does not support reserving sessions, so
fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE.

The fax_gateway_start does not create a real fax session if the fax session
is already present and the state is not AST_FAX_STATE_RESERVED.
But the "reserved" session created for res_fax_spandsp has state
AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting.

Then when fax_gateway_framehook is called and gateway T.38 state is
NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to
segfault, because session tech_pvt is not set, i.e. the tech session
was not initialized/started.

This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved"
session created for res_fax_spandsp will start.

This patch also adds extra check and log ERROR if tech_pvt is not set
before call tech->write.

ASTERISK-27981 #close

Change-Id: Ife3e65e5f18c902db2ff0538fccf7d28f88fa803

10 months agores_fax: add channel name to CLI 'fax show session'
Alexei Gradinari [Tue, 28 May 2019 22:15:40 +0000 (18:15 -0400)]
res_fax: add channel name to CLI 'fax show session'

This patch adds a channel name to output of CLI 'fax show session'
and also expands the channel name field up to 30 characters on
CLI 'fax show sessions'

Change-Id: Id059c43ff41811f5e76712b83fb63b8f246da953

10 months agobuild: Fix file format in CHANGES-staging.
Ben Ford [Fri, 24 May 2019 14:01:14 +0000 (09:01 -0500)]
build: Fix file format in CHANGES-staging.

One of the change files doesn't conform to the format that the release
scripts need in order to parse it.

Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c

10 months agochan_dahdi: add missing include.
Guido Falsi [Thu, 23 May 2019 14:44:07 +0000 (16:44 +0200)]
chan_dahdi: add missing include.

After some definitions have been moved to asterisk/mwi.h the files
channels/chan_dahdi.h channels/sig_pri.c are missing this new
include.

ASTERISK-28427 #close

Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91

10 months agoapp_readexten: new option 'p' to stop reading on '#' key
Alexei Gradinari [Fri, 17 May 2019 22:45:25 +0000 (18:45 -0400)]
app_readexten: new option 'p' to stop reading on '#' key

This patch adds the 'p' option.
The extension entered will be considered complete when a # is entered.

Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1

10 months agoMerge "res_rtp_asterisk: timestamp should be unsigned instead of signed int"
Friendly Automation [Thu, 23 May 2019 14:03:49 +0000 (09:03 -0500)]
Merge "res_rtp_asterisk: timestamp should be unsigned instead of signed int"

10 months agoMerge "pjproject-bundled: Add upstream timer fixes"
Friendly Automation [Wed, 22 May 2019 17:07:50 +0000 (12:07 -0500)]
Merge "pjproject-bundled:  Add upstream timer fixes"

10 months agoMerge "res_rtp_asterisk: Add ability to propose local address in ICE"
Friendly Automation [Wed, 22 May 2019 16:28:18 +0000 (11:28 -0500)]
Merge "res_rtp_asterisk:  Add ability to propose local address in ICE"

10 months agores_prometheus: Add metrics for PJSIP outbound registrations
Matt Jordan [Fri, 10 May 2019 14:36:01 +0000 (09:36 -0500)]
res_prometheus: Add metrics for PJSIP outbound registrations

When monitoring Asterisk instances, it's often useful to know when an
outbound registration fails, as this often maps to the notion of a trunk
and having a trunk fail is usually a "bad thing". As such, this patch
adds monitoring metrics that track the state of PJSIP outbound registrations.
It does this by looking for the Registry events coming across the Stasis
system topic, and publishing those as metrics to Prometheus. Note that
while this may support other outbound registration types (IAX2, SIP, etc.)
those haven't been tested. Your mileage may vary.

(And why are you still using IAX2 and SIP? It's 2019 folks. Get with the
program.)

This patch also adds Sorcery observers to handle modifications to the
underlying PJSIP outbound registration objects. This is useful when a
reload is triggered that modifies the properties of an outbound registration,
or when ARI push configuration is used and an object is updated or
deleted. Because we rely on properties of the registration object to
define the metric (label key/value pairs), we delete the relevant metric when
we notice that something has changed and wait for a new Stasis message to
arrive to re-create the metric.

ASTERISK-28403

Change-Id: If01420e38530fc20b6dd4aa15cd281d94cd2b87e

10 months agores_prometheus: Add CLI commands
Matt Jordan [Thu, 3 Jan 2019 16:28:28 +0000 (10:28 -0600)]
res_prometheus: Add CLI commands

This patch adds a few CLI commands to the res_prometheus module to aid
system administrators setting up and configuring the module. This includes:

* prometheus show status: Display basic statistics about the Prometheus
  module, including its essential configuration, when it was last scraped,
  and how long the scrape took. The last two bits of information are useful
  when Prometheus isn't generating metrics appropriately, as it will at
  least tell you if Asterisk has had its HTTP route hit by the remote
  server.

* prometheus show metrics: Dump the current metrics to the CLI. Useful for
  system administrators to see what metrics are currently available without
  having to cURL or go to Prometheus itself.

ASTERISK-28403

Change-Id: Ic09813e5e14b901571c5c96ebeae2a02566c5172

10 months agores_prometheus: Add Asterisk bridge metrics
Matt Jordan [Thu, 9 May 2019 14:41:49 +0000 (09:41 -0500)]
res_prometheus: Add Asterisk bridge metrics

This patch adds basic Asterisk bridge statistics to the res_prometheus
module. This includes:

* asterisk_bridges_count: The current number of bridges active on the
  system.

* asterisk_bridges_channels_count: The number of channels active in a
  bridge.

In all cases, enough information is provided with each bridge metric
to determine a unique instance of Asterisk that provided the data, along
with the technology, subclass, and creator of the bridge.

ASTERISK-28403

Change-Id: Ie27417dd72c5bc7624eb2a7a6a8829d7551788dc

10 months agores_prometheus: Add Asterisk endpoint metrics
Matt Jordan [Thu, 9 May 2019 14:41:02 +0000 (09:41 -0500)]
res_prometheus: Add Asterisk endpoint metrics

This patch adds basic Asterisk endpoint statistics to the res_prometheus
module. This includes:

* asterisk_endpoints_state: The current state (unknown, online, offline)
  for each defined endpoint.

* asterisk_endpoints_channels_count: The current number of channels
  associated with a given endpoint.

* asterisk_endpoints_count: The current number of defined endpoints.

In all cases, enough information is provided with each endpoint metric
to determine a unique instance of Asterisk that provided the data, as well
as the underlying technology and resource definition.

ASTERISK-28403

Change-Id: I46443963330c206a7d12722d08dcaabef672310e

10 months agores_rtp_asterisk: timestamp should be unsigned instead of signed int
Morten Tryfoss [Tue, 21 May 2019 16:29:05 +0000 (18:29 +0200)]
res_rtp_asterisk: timestamp should be unsigned instead of signed int

Using timestamp with signed int will cause timestamps exceeding max value
to be negative.
This causes the jitterbuffer to do passthrough of the packet.

ASTERISK-28421

Change-Id: I9dabd0718180f2978856c50f43aac4e52dc3cde9

10 months agores_prometheus: Add Asterisk channel metrics
Matt Jordan [Fri, 3 May 2019 00:45:27 +0000 (19:45 -0500)]
res_prometheus: Add Asterisk channel metrics

This patch adds basic Asterisk channel statistics to the res_prometheus
module. This includes:

* asterisk_calls_sum: A running sum of the total number of
  processed calls

* asterisk_calls_count: The current number of calls

* asterisk_channels_count: The current number of channels

* asterisk_channels_state: The state of any particular channel

* asterisk_channels_duration_seconds: How long a channel has existed,
  in seconds

In all cases, enough information is provided with each channel metric
to determine a unique instance of Asterisk that provided the data, as
well as the name, type, unique ID, and - if present - linked ID of each
channel.

ASTERISK-28403

Change-Id: I0db306ec94205d4f58d1e7fbabfe04b185869f59

10 months agopjproject/Makefile: Updates for Darwin compatible builds
Matt Jordan [Mon, 29 Apr 2019 15:10:35 +0000 (10:10 -0500)]
pjproject/Makefile: Updates for Darwin compatible builds

This patch fixes three compatibility issues for Darwin compatible builds:

(1) Use BSD compatible command line option for sed

For some versions of BSD sed, the -r command line option is unknown.
Both GNU and BSD sed support the -E command line option for enabling
extended regular expressions; as such, this patch replaces the -r
option with -E.

(2) Look for '_' in pjproject generated symbols

In Darwin comaptible systems, the symbols generated for pjproject may be
prefixed with an '_'. When exporting these to a symbol file, the invocation
to sed has to optionally look for a prefix of said '_' character.

(3) Use -all_load/-noall_load when linking

The flags -whole-archive/-no-whole-archive are not supported by the
linker, and must instead be replaced with -all_load/-noall_load.

Change-Id: I58121756de6a0560a6e49ca9d6bf9566a333cde3

10 months agoMerge "Add core Prometheus support to Asterisk"
Friendly Automation [Tue, 21 May 2019 15:11:04 +0000 (10:11 -0500)]
Merge "Add core Prometheus support to Asterisk"

10 months agoAdd core Prometheus support to Asterisk
Matt Jordan [Thu, 3 Jan 2019 16:28:28 +0000 (10:28 -0600)]
Add core Prometheus support to Asterisk

Prometheus is the defacto monitoring tool for containerized applications.
This patch adds native support to Asterisk for serving up Prometheus
compatible metrics, such that a Prometheus server can scrape an Asterisk
instance in the same fashion as it does other HTTP services.

The core module in this patch provides an API that future work can build
on top of. The API manages metrics in one of two ways:
(1) Registered metrics. In this particular case, the API assumes that
    the metric (either allocated on the stack or on the heap) will have
    its value updated by the module registering it at will, and not
    just when Prometheus scrapes Asterisk. When a scrape does occur,
    the metrics are locked so that the current value can be retrieved.
(2) Scrape callbacks. In this case, the API allows consumers to be
    called via a callback function when a Prometheus initiated scrape
    occurs. The consumers of the API are responsible for populating
    the response to Prometheus themselves, typically using stack
    allocated metrics that are then formatted properly into strings
    via this module's convenience functions.

These two mechanisms balance the different ways in which information is
generated within Asterisk: some information is generated in a fashion
that makes it appropriate to update the relevant metrics immediately;
some information is better to defer until a Prometheus server asks for
it.

Note that some care has been taken in how metrics are defined to
minimize the impact on performance. Prometheus's metric definition
and its support for nesting metrics based on labels - which are
effectively key/value pairs - can make storage and managing of metrics
somewhat tricky. While a naive approach, where we allow for any number
of labels and perform a lot of heap allocations to manage the information,
would absolutely have worked, this patch instead opts to try to place
as much information in length limited arrays, stack allocations, and
vectors to minimize the performance impacts of scrapes. The author of
this patch has worked on enough systems that were driven to their knees
by poor monitoring implementations to be a bit cautious.

Additionally, this patch only adds support for gauges and counters.
Additional work to add summaries, histograms, and other Prometheus
metric types may add value in the future. This would be of particular
interest if someone wanted to track SIP response types.

Finally, this patch includes unit tests for the core APIs.

ASTERISK-28403

Change-Id: I891433a272c92fd11c705a2c36d65479a415ec42

10 months agopjproject-bundled: Add upstream timer fixes
Joshua Colp [Mon, 20 May 2019 17:45:57 +0000 (14:45 -0300)]
pjproject-bundled:  Add upstream timer fixes

Fixed #2191:
  - Stricter double timer entry scheduling prevention.
  - Integrate group lock in SIP transport, e.g: for add/dec ref,
    for timer scheduling.

ASTERISK-28161
Reported-by: Ross Beer

Change-Id: I2e09aa66de0dda9414d8a8259a649c4d2d96a9f5

10 months agores_rtp_asterisk: Add ability to propose local address in ICE
George Joseph [Fri, 17 May 2019 23:44:37 +0000 (17:44 -0600)]
res_rtp_asterisk:  Add ability to propose local address in ICE

You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:

[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address

This causes both 192.168.1.1 and 1.2.3.4 to be advertized.

Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db

10 months agopjsip: replace 180 by 183 if SDP negotiation has completed
Alexei Gradinari [Mon, 13 May 2019 20:37:50 +0000 (16:37 -0400)]
pjsip: replace 180 by 183 if SDP negotiation has completed

The caller endpoint hears dead silence if a callee replies 180 (without SDP)
and the caller already received 183 (with SDP).
It happens because Asterisk sends 180 (WITH SDP) to the caller,
there are not incoming RTP packets from the callee
and Asterisk does not generate inband ringing,
so there are not any outgoing RTP packets to the caller.

This patch replaces 180 by 183 if SDP negotiation has completed,
as if the caller endpoint is configured with "inband_progress=yes".

In this case Asterisk will generate inband ringing untill Asterisk receive
incoming RTP packets from the callee.

ASTERISK-27994 #close

Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73

10 months agoMerge "res_rtp_asterisk: Fix sequence number cycling and packet loss count."
Friendly Automation [Wed, 15 May 2019 22:45:56 +0000 (17:45 -0500)]
Merge "res_rtp_asterisk: Fix sequence number cycling and packet loss count."

10 months agoMerge "conversions.c: Add conversions for largest max sized integer"
Friendly Automation [Wed, 15 May 2019 12:04:37 +0000 (07:04 -0500)]
Merge "conversions.c: Add conversions for largest max sized integer"

10 months agoMerge "Fixes for GCC 9"
Friendly Automation [Wed, 15 May 2019 11:29:03 +0000 (06:29 -0500)]
Merge "Fixes for GCC 9"

10 months agoMerge "build: Pass --fno-partial-inlining to third-party when appropriate"
Friendly Automation [Wed, 15 May 2019 10:45:36 +0000 (05:45 -0500)]
Merge "build: Pass --fno-partial-inlining to third-party when appropriate"

10 months agoMerge "pjsip_options.c: Allow immediate qualifies for new contacts."
Friendly Automation [Mon, 13 May 2019 19:11:08 +0000 (14:11 -0500)]
Merge "pjsip_options.c: Allow immediate qualifies for new contacts."

10 months agoFixes for GCC 9
George Joseph [Fri, 10 May 2019 15:48:28 +0000 (09:48 -0600)]
Fixes for GCC 9

Various fixes for issues caught by gcc 9.  Mostly snprintf
trying to copy to a buffer potentially too small.

ASTERISK-28412

Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e

10 months agores_rtp_asterisk: Fix sequence number cycling and packet loss count.
Joshua Colp [Wed, 8 May 2019 15:41:43 +0000 (15:41 +0000)]
res_rtp_asterisk: Fix sequence number cycling and packet loss count.

This change fixes two bugs which both resulted in the packet loss
count exceeding 65,000.

The first issue is that the sequence number check to determine if
cycling had occurred was using the wrong variable resulting in the
check never seeing that cycling has occurred, throwing off the
packet loss calculation. It now uses the correct variable.

The second issue is that the packet loss calculation assumed that
the received number of packets in an interval could never exceed
the expected number. In practice this isn't true due to delayed
or retransmitted packets. The expected will now be updated to
the received number if the received exceeds it.

ASTERISK-28379

Change-Id: If888ebc194ab69ac3194113a808c414b014ce0f6

10 months agopjsip_options.c: Allow immediate qualifies for new contacts.
Ben Ford [Tue, 7 May 2019 16:08:33 +0000 (11:08 -0500)]
pjsip_options.c: Allow immediate qualifies for new contacts.

When multiple endpoints try to register close together using the same
AOR with qualify_frequency set, one contact would qualify immediately
while the other contacts would have to wait out the duration of the
timer before being able to qualify. Changing the conditional to check
the contact container count for a non-zero value allows all contacts to
qualify immediately.

Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415

10 months agoconversions.c: Add conversions for largest max sized integer
Kevin Harwell [Mon, 6 May 2019 21:26:46 +0000 (16:26 -0500)]
conversions.c: Add conversions for largest max sized integer

Added a conversion for umax (largest maximum sized integer allowed). Adjusted
the other current conversion functions (uint and ulong) to be derivatives of
the umax conversion since they are simply subsets of umax.

Also made the negative check move the pointer on spaces since strtoumax does it
anyways.

Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08

10 months agostasis: Hangup channel for Local channel No such extension error
agupta [Fri, 3 May 2019 15:49:31 +0000 (21:19 +0530)]
stasis: Hangup channel for Local channel No such extension error

When we use early bridge with create and dial from stasis using Local channel
and the dialplan does not any entry the it is returned from core_local.c with
No such extension .

In such case asterisk locks up till the channel is not hangup with the error
Exceptionally long voice queue length

* Found that in such case app_control_dial fails on ast_call method and
  return -1
* Since it is called from stasis_app_send_command_async and return -1 does
  not cause resources to be freed and since no PBX exist it is not able to
  read from channel causing exceptionally long queue
* After putting this code found that the channel was releasing immediately
  and resources were freed.

ASTERISK-28399
Reported by: Abhay Gupta
Tested by: Abhay Gupta

Change-Id: I0a55c923fc6995559f808d63b9488762b4489318

10 months agobuild: Pass --fno-partial-inlining to third-party when appropriate
George Joseph [Fri, 3 May 2019 18:31:06 +0000 (12:31 -0600)]
build: Pass --fno-partial-inlining to third-party when appropriate

When the gcc version is >= 8.2.1, we were already setting the
--fno-partial-inlining flag for Asterisk source files to get around
a gcc bug but we weren't passing the flag down to the bundled
builds of pjproject and jansson.

ASTERISK-28392

Change-Id: I99ede9bc35408ecd096f7d5369e8192d3dc75704

10 months agoMerge "app_confbridge: Add "all" variants of REMB behavior."
Friendly Automation [Fri, 3 May 2019 15:54:13 +0000 (10:54 -0500)]
Merge "app_confbridge: Add "all" variants of REMB behavior."

10 months agoMerge "stasis: Only place stasis created and dialed channels into dial bridge."
Joshua Colp [Fri, 3 May 2019 15:50:38 +0000 (10:50 -0500)]
Merge "stasis: Only place stasis created and dialed channels into dial bridge."

10 months agoMerge "stasis: Call callbacks when imparting fails"
Friendly Automation [Fri, 3 May 2019 15:13:33 +0000 (10:13 -0500)]
Merge "stasis: Call callbacks when imparting fails"

10 months agoMerge "rtp: Add support for transport-cc in receiver direction."
Friendly Automation [Fri, 3 May 2019 15:06:43 +0000 (10:06 -0500)]
Merge "rtp: Add support for transport-cc in receiver direction."

10 months agores_pjsip: Check return from pjsip_parse_uri calls
George Joseph [Thu, 2 May 2019 18:29:49 +0000 (12:29 -0600)]
res_pjsip:  Check return from pjsip_parse_uri calls

Updated ast_sip_create_rdata_with_contact and registrar_find_contact
to check the return from pjsip_parse_uri before attempting to
use the uri returned.

ASTERISK-28402
Reported-by: Ross Beer

Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7

10 months agostasis: Only place stasis created and dialed channels into dial bridge.
agupta [Tue, 30 Apr 2019 14:21:46 +0000 (19:51 +0530)]
stasis: Only place stasis created and dialed channels into dial bridge.

The dial bridge is meant to hold channels which have been created
and dialed in stasis. It handles the frames coming from them and raises
the appropriate events.

It was possible for the code to mistakenly place calls which came
from the dialplan into the dial bridge if they were not in an
answered state. These channels are not outgoing channels and
should not be placed into the dial bridge.

The code now checks to ensure that only stasis created channels are
placed into the dial bridge by checking that a PBX does not exist
on the channel.

ASTERISK-27756

Change-Id: Ideee69ff06c9a0b31f7ed61165f5c055f51d21b6

10 months agostasis: Call callbacks when imparting fails
Holger Hans Peter Freyther [Wed, 10 Apr 2019 04:30:25 +0000 (05:30 +0100)]
stasis: Call callbacks when imparting fails

After a bridge has been deleted the stasis control will depart
the channel and might attempt to re-add it to the dial bridge.

The later can fail and this can lead to a situation that the stasis
control is unlinked but the after_bridge_cb_failed cb is executed trying
to access a dangling control object.

Fix it by calling the after_cb's before bridge_channel_impart_signal.

ASTERISK-26718

Change-Id: Ib4e8f70d7a21bd54afe3cb51cc6717ef7c355496

10 months agoapp_confbridge: Add "all" variants of REMB behavior.
Joshua Colp [Tue, 30 Apr 2019 11:22:44 +0000 (11:22 +0000)]
app_confbridge: Add "all" variants of REMB behavior.

When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.

This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.

ASTERISK-28401

Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c

10 months agortp: Add support for transport-cc in receiver direction.
Joshua Colp [Tue, 23 Apr 2019 10:00:43 +0000 (10:00 +0000)]
rtp: Add support for transport-cc in receiver direction.

The transport-cc draft is a mechanism by which additional information
about packet reception can be provided to the sender of packets so
they can do sender side bandwidth estimation. This is accomplished
by having a transport specific sequence number and an RTCP feedback
message. This change implements this in the receiver direction.

For each received RTP packet where transport-cc is negotiated we store
the time at which the RTP packet was received and its sequence number.
At a 1 second interval we go through all packets in that period of time
and use the stored time of each in comparison to its preceding packet to
calculate its delta. This delta information is placed in the RTCP
feedback message, along with indicators for any packets which were not
received.

The browser then uses this information to better estimate available
bandwidth and adjust accordingly. This may result in it lowering the
available send bandwidth or adjusting how "bursty" it can be.

ASTERISK-28400

Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc

10 months agoMerge "app_queue: Set correct value by default for shared_lastcall"
Friendly Automation [Tue, 30 Apr 2019 21:45:48 +0000 (16:45 -0500)]
Merge "app_queue: Set correct value by default for shared_lastcall"

10 months agoMerge "mwi core: Move core MWI functionality into its own files"
Friendly Automation [Tue, 30 Apr 2019 15:41:10 +0000 (10:41 -0500)]
Merge "mwi core: Move core MWI functionality into its own files"

10 months agoMerge "app_amd: Fix infinite loop on silent calls"
Friendly Automation [Tue, 30 Apr 2019 15:07:30 +0000 (10:07 -0500)]
Merge "app_amd: Fix infinite loop on silent calls"

10 months agoMerge "stasis: Fix crash at shutdown."
Friendly Automation [Tue, 30 Apr 2019 10:44:53 +0000 (05:44 -0500)]
Merge "stasis: Fix crash at shutdown."

10 months agoapp_amd: Fix infinite loop on silent calls
agupta [Tue, 4 Dec 2018 08:10:15 +0000 (13:40 +0530)]
app_amd: Fix infinite loop on silent calls

The total time logic will now be executed on calls which
do not pass any media.

ASTERISK-28143

Change-Id: I24726bd29d7e467fc721ca265363417234b22855

10 months agoapp_queue: Set correct value by default for shared_lastcall
Rodrigo Ramírez Norambuena [Mon, 29 Apr 2019 16:13:07 +0000 (12:13 -0400)]
app_queue: Set correct value by default for shared_lastcall

There a long history here:

In commit dd1e62c095c has introduce by default shared_lastcall = true by
default but this now only happen is there not [general] directive in
queues.conf

After that, the commit 4b50e3f1ee84ae29da6d9eb3cfd9896a49d2394b fix the
sample file.

We'll need to keep the same setting if there a general or not section in
configuration file since the shared_lastcall is by a long time in
sample files as default value to 'no'.

Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c

11 months agostasis: Fix crash at shutdown.
Ben Ford [Tue, 23 Apr 2019 14:47:45 +0000 (09:47 -0500)]
stasis: Fix crash at shutdown.

When compiling in dev mode, stasis statistics are enabled and can cause
a crash at shutdown due to the following:
- Containers are freed
- Topics and subscriptions remain
- When those topics and subscriptions are deallocated, they go to do
  things with the container

This changes the containers to global ao2 objects, and whenever needed
in the code, a reference must be obtained and checked before any
operations can be done.

ASTERISK-28353 #close

Change-Id: Ie7d5e907fcfcb4d65bd36d5e4eb923126fde8d33

11 months agoapp_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings
Antoni Goldstein [Fri, 29 Mar 2019 14:04:46 +0000 (14:04 +0000)]
app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings

Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.

Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.

ASTERISK-28363

Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1

11 months agomwi core: Move core MWI functionality into its own files
Kevin Harwell [Tue, 9 Apr 2019 19:48:22 +0000 (14:48 -0500)]
mwi core: Move core MWI functionality into its own files

There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:

main/mwi.h
main/mwi.c

Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.

Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0

11 months agoMerge "core/buildsystem: check the actual compiler being version"
Friendly Automation [Tue, 23 Apr 2019 20:23:55 +0000 (15:23 -0500)]
Merge "core/buildsystem: check the actual compiler being version"

11 months agoMerge "main/stasis.c: Added detail info for stasis show app cli"
Friendly Automation [Tue, 23 Apr 2019 18:18:07 +0000 (13:18 -0500)]
Merge "main/stasis.c: Added detail info for stasis show app cli"

11 months agocore/buildsystem: check the actual compiler being version
Guido Falsi [Sun, 7 Apr 2019 16:36:55 +0000 (18:36 +0200)]
core/buildsystem: check the actual compiler being version

Make compiler check use the output of the actual compiler being
used as reported by the CC variable, instead of unconditionally
running the "gcc" binary.  Also only run the check if the compiler
is gcc or a cross-compile gcc.

ASTERISK-28374

Change-Id: Icaacf6d93686ad21076878aa1504a23b4fc9d0f4

11 months agores_indications: Fix indications remove command autocomplete
Lucas Mendes [Fri, 19 Apr 2019 14:33:49 +0000 (16:33 +0200)]
res_indications: Fix indications remove command autocomplete

We changed the validation of autocomplete parameter in the "indications
remove" command to avoid continue the execution of the command after
asking for autocomplete out of range parameters.

ASTERISK-28391
Reported by: lmendes86

Change-Id: I92b24131fd02f2e3c7fec966eea6f7a663310d40

11 months agoMerge "loader: support for permanent dlopen()"
Friendly Automation [Fri, 19 Apr 2019 14:06:03 +0000 (09:06 -0500)]
Merge "loader: support for permanent dlopen()"

11 months agoMerge "res_pjsip: Added a norefersub configuration setting"
Joshua Colp [Fri, 19 Apr 2019 13:30:14 +0000 (08:30 -0500)]
Merge "res_pjsip:  Added a norefersub configuration setting"

11 months agores_remb_modifier: Propertly initialize bitrate to 0.0
George Joseph [Wed, 17 Apr 2019 19:45:26 +0000 (13:45 -0600)]
res_remb_modifier:  Propertly initialize bitrate to 0.0

...and return the frame unaltered if bitrate can't be determined.

Change-Id: Ib2175ab84f85a3d7060d31625f5a2c7fbcc2ba4c

11 months agoMerge "res_mwi_devstate: Specify AST_MODFLAG_LOAD_ORDER to enable load priority"
Friendly Automation [Thu, 18 Apr 2019 10:44:01 +0000 (05:44 -0500)]
Merge "res_mwi_devstate: Specify AST_MODFLAG_LOAD_ORDER to enable load priority"

11 months agores_pjsip: Added a norefersub configuration setting
Dan Cropp [Mon, 8 Apr 2019 22:04:48 +0000 (17:04 -0500)]
res_pjsip:  Added a norefersub configuration setting

Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.

res_pjsip_refer:  Configures PJSIP norefersub capability accordingly.

Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.

This is useful for Cisco switches that do not follow RFC4488.

ASTERISK-28375 #close
Reported-by: Dan Cropp

Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9

11 months agomain/stasis.c: Added detail info for stasis show app cli
sungtae kim [Wed, 10 Apr 2019 00:09:10 +0000 (02:09 +0200)]
main/stasis.c: Added detail info for stasis show app cli

Currently, the "stasis show app" cli doesn't give detail
of subscription/subscriber information.
Added more printings to show details.

ASTERISK-28378

Change-Id: If25a6f14fe4f622bfb37462e891333da1fdf875f

11 months agoMerge "pbx.c: Ignore dashes in extensions when using extenpatternmatchnew"
Friendly Automation [Tue, 16 Apr 2019 16:38:52 +0000 (11:38 -0500)]
Merge "pbx.c: Ignore dashes in extensions when using extenpatternmatchnew"

11 months agoMerge "app_voicemail: Don't split mailbox options on comma"
Friendly Automation [Tue, 16 Apr 2019 16:37:09 +0000 (11:37 -0500)]
Merge "app_voicemail: Don't split mailbox options on comma"

11 months agores_mwi_devstate: Specify AST_MODFLAG_LOAD_ORDER to enable load priority
Sean Bright [Tue, 16 Apr 2019 15:58:40 +0000 (11:58 -0400)]
res_mwi_devstate: Specify AST_MODFLAG_LOAD_ORDER to enable load priority

Suggested by abelbeck on the issue tracker.

ASTERISK~28384
Reported by: abelbeck

Change-Id: Icee0fff2b58dbfaa80f2b68270fe69dfb0463fc0

11 months agoMerge "build: Revise CHANGES and UPGRADE.txt handling."
Benjamin Keith Ford [Tue, 16 Apr 2019 15:52:51 +0000 (10:52 -0500)]
Merge "build: Revise CHANGES and UPGRADE.txt handling."

11 months agoMerge "res_ael: Use Gosub in for loop expressions"
Joshua Colp [Tue, 16 Apr 2019 13:11:28 +0000 (08:11 -0500)]
Merge "res_ael: Use Gosub in for loop expressions"

11 months agoMerge "ARI: Run 'make ari-stubs'"
Joshua Colp [Tue, 16 Apr 2019 12:29:45 +0000 (07:29 -0500)]
Merge "ARI:  Run 'make ari-stubs'"

11 months agoMerge "res_ael: Fix pattern matching against literal '+'"
Joshua Colp [Tue, 16 Apr 2019 12:25:40 +0000 (07:25 -0500)]
Merge "res_ael: Fix pattern matching against literal '+'"

11 months agoCI: Move test group config files to Jenkins
George Joseph [Fri, 12 Apr 2019 16:32:44 +0000 (10:32 -0600)]
CI: Move test group config files to Jenkins

One of the downaides of having things like test configuration
in the git repo is that it can't be changed at runtime.  You have
to create a review for the changes and merge it mefore it will
take effect.

This review moves the data currently held in
tests/CI/periodic-dailyTestGroups.json and
tests/CI/gateTestGroups.json into a Jenkins Config File attached
to the job definitions.  This allows us to alter it from the
Jenkins UI at runtime.  The original files stay in the repo
as documentation.

Change-Id: I14b9702f6285ce1fb2420287ba0e7d3b59109763

11 months agoapp_voicemail: Don't split mailbox options on comma
Sean Bright [Sat, 13 Apr 2019 18:36:56 +0000 (14:36 -0400)]
app_voicemail: Don't split mailbox options on comma

Because the per-mailbox options are the last thing on a line, don't look
for or stomp on any subsequent commas.

ASTERISK-27935 #close
Reported by: Sébastien Duthil

Change-Id: I07b2eb4a33c303d0c7114d5b906f8c067c60a153

11 months agoMerge "res_ael: Create consistent label names across reloads"
George Joseph [Fri, 12 Apr 2019 19:16:31 +0000 (14:16 -0500)]
Merge "res_ael: Create consistent label names across reloads"

11 months agoMerge "pbx.c: Properly parse labels with leading digits"
George Joseph [Fri, 12 Apr 2019 19:16:10 +0000 (14:16 -0500)]
Merge "pbx.c: Properly parse labels with leading digits"

11 months agopbx.c: Ignore dashes in extensions when using extenpatternmatchnew
Sean Bright [Fri, 12 Apr 2019 14:33:57 +0000 (10:33 -0400)]
pbx.c: Ignore dashes in extensions when using extenpatternmatchnew

Because hyphens are not matched literally in Asterisk dialplan, we need
to ignore them in our candidate extensions as well.

ASTERISK-17695 #close
Reported by: test011

Change-Id: I227f02301577b1633e8a55b9fe9dc149935c03f0

11 months agoMerge "app_voicemail: Cleanup stale lock files on module load"
Friendly Automation [Fri, 12 Apr 2019 15:00:40 +0000 (10:00 -0500)]
Merge "app_voicemail: Cleanup stale lock files on module load"

11 months agoMerge "chan_ooh323: fix h323 log file path"
Friendly Automation [Fri, 12 Apr 2019 14:19:15 +0000 (09:19 -0500)]
Merge "chan_ooh323: fix h323 log file path"

11 months agoapp_voicemail: Cleanup stale lock files on module load
Sean Bright [Tue, 9 Apr 2019 15:10:12 +0000 (11:10 -0400)]
app_voicemail: Cleanup stale lock files on module load

If Asterisk crashes while a VM directory is locked, lock files in the VM
spool directory will not get properly cleaned up. We now clear them on
module load.

ASTERISK-20207 #close
Reported by: Steven Wheeler

Change-Id: If40ccd508e2f6e5ade94dde2f0bcef99056d0aaf

11 months agoARI: Run 'make ari-stubs'
George Joseph [Fri, 12 Apr 2019 12:33:10 +0000 (06:33 -0600)]
ARI:  Run 'make ari-stubs'

An earlier contributor apparently forgot to run 'make ari-stubs'
before committing after making ARI model changes.

Change-Id: I7813e5638e2821d11f4b968dc2aeab4f725190a6

11 months agores_ael: Create consistent label names across reloads
Sean Bright [Thu, 11 Apr 2019 20:48:49 +0000 (16:48 -0400)]
res_ael: Create consistent label names across reloads

Reset the internal counter that the AEL2 compiler uses for unique label
names before compiling. This keeps dialplan labels consistent across
reloads assuming the AEL2 has not changed.

ASTERISK-17799 #close
Reported by: Kirill Katsnelson

Change-Id: I30b3cc887d1ee0644d3f341e2fef16f525d7fae5