We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.

Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0

This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.

A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.

The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.

ASTERISK-26732 #close
Reported by Dan Jenkins

Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5

When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered

ASTERISK-26869 #close

Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525

A caller can leave the Queue() application after being bridged with a
member in a few ways:

  * Caller or member hangup
  * Caller is transferred somewhere else (blind or atx)
  * Caller is externally redirected elsewhere

The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.

This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.

ASTERISK-26400 #close
Reported by: Etienne Lessard

Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334

This change ensures that if no header_match option is set on an
identify an error message is not output stating the option is set
to an invalid value.

ASTERISK-26863

Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a

This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.

Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.

ASTERISK-26863 #close

Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79)

Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range.  e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.

The buffer overwrite is fixed two ways in this patch.

1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens.  Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set.  Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.

2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.

ASTERISK-26668

Change-Id: I555d97411140e47e0522684062d174fbe32aa84a

When using a non-dynamic peer address, build_peer() invalidates the
peer address structure by setting the address family to unspecified.
However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup()
will not amend the peer address if the cache is still valid, resulting
in peer connectivity failures.
To fix this, we call ast_dnsmgr_refresh() instead.

ASTERISK-26865

Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082

In some scenarios, such as when there may not be a terminal (such as
inside a Docker container), curl will apparently direct the progress bar
to stdout. This can cause extra data to be appended to a file curl'd
down to stdout, resulting in md5 verification failures.

This patch removes the progress bar, and tells curl to download the file
silently.

ASTERISK-26872 #close

Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c

* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.

Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48

This patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.

ASTERISK-26717 #close

Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124

During module loading of func_devstate, Asterisk emits the current
device state of all Custom device states currently stored in the AstDB.
This was erroneously including a new line character ('\n') to the end of
the device state, causing two new lines to be emitted in
DeviceStateChange AMI events.

Note that this only happened for those device state changes that
occurred during startup. Regular device state changes for Custom device
states are handled elsewhere, and did not have the newline.

ASTERISK-26643 #close
Reported by: Roman Bedros
Tested by: Matt Jordan
patches:
  ami_devstate.diff uploaded by Roman Bedros (License 6842)

Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93

This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.

Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:

"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."

ASTERISK-25237 #close

Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70

Tabs > spaces. Always.

Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1

When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.

This change just adds a check that the channel exists on the
session before querying it.

ASTERISK-26857

Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01

Bundled pjproject should now only rebuild if one of the menuselect
"Compiler Flags" options changes.

Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43

* cli_commands.c Fixed CLI output

ASTERISK-26822 #close

Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01

* res_musiconhold.c: Ensure the general section is not treated as
a moh class.

ASTERISK-26353 #close

Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d

Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def