On FreeBSD using the clang/llvm compiler build fails to build due
to the switch statement argument being a non integer type expression.
Switch to an if/else if/else construct to sidestep the issue.

ASTERISK-28536 #close

Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8

According to RFC3550, ALL RTCP packets must be sent in a compond packet
of at least two individual packets, including SR/RR and SDES. REMB,
FIR, and NACK were not following this format, and as a result, would
fail the packet check in ast_rtcp_interpret. This was found from writing
unit tests for RTCP. The browser would accept the way we were
constructing these RTCP packets, but when sending directly from one
Asterisk instance to another, the above mentioned problem would occur.

Change-Id: Ieb140e9c22568a251a564cd953dd22cd33244605

When modifying an already defined variable in some channel drivers they
add a new variable with the same name to the list, but that value is
never used, only the first one found.

Introduce ast_variable_list_replace() and use it where appropriate.

ASTERISK-23756 #close
Patches:
  setvar-multiplie.patch submitted by Michael Goryainov

Change-Id: Ie1897a96c82b8945e752733612ee963686f32839

This fix allows a realtime moh class to be unregistered from the command
line. This is useful when the contents of a directory referenced by a
realtime moh class have changed.
The realtime moh class is then reloaded on the next request and uses the
new directory contents.

ASTERISK-17808

Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce

Added unit tests for RTCP video stats. These tests include NACK, REMB,
FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR
tests are currently disabled due to a bug. We expect to receive a
compound packet, but the code sends this out as a single packet, which
the browser accepts, but makes Asterisk upset.

While writing these tests, I noticed an issue with NACK as well. Where
it is handling a received NACK request, it was reading in only the first
8 bits of following packets that were also lost. This has been changed
to the correct value of 16 bits.

Also made a minor fix to the data buffer unit test.

Change-Id: I56107c7411003a247589bbb6086d25c54719901b

ChanIsAvail() creates a temporary channel with ast_request() to test
resource availability. It should not generate a CDR when it hangs up
this temporary channel.

This patch disables CDR generation for the temporary channel with
ast_cdr_set_property().

ASTERISK-28527

Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1

When the remote ISDN party ends an ISDN call on a PRI link
(DISCONNECT), CHANNEL(hangupsource) information is not available.

chan_dahdi already contains an ast_set_hangupsource() in
__dahdi_exception() function but it seems that ISDN message processing
does not use this part of code.

Two other channel modules associate ast_queue_hangup() and
ast_set_hangupsource() functions calls:
- chan_pjsip in chan_pjsip_session_end() function,
- chan_sip in sip_queue_hangup_cause() function.
chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and
set_hangup_source_and_cause().

Thus, I propose to add ast_set_hangupsource() beside
ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and
chan_sip already do.

ASTERISK-28525

Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c

The Channel resource has a new sub-resource "externalMedia".
This allows an application to create a channel for the sole purpose
of exchanging media with an external server.  Once created, this
channel could be placed into a bridge with existing channels to
allow the external server to inject audio into the bridge or
receive audio from the bridge.
See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
for more information.

Change-Id: I9618899198880b4c650354581b50c0401b58bc46

The links in the deprecation notice were the shortened
variety but it makes better sense to show the unshortened
links as they're more descriptive.

I.E.
wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
rather than
wiki.asterisk.org/wiki/x/tAHOAQ

Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9

ASTERISK-28511

Change-Id: If0d58598ce14aad3c786a1c0127b5f7b200b737d

This change removes the assumption that a frame will always have
a src set on it. This assumption is incorrect.

Given a scenario where an RTP packet is received with no payload
the resulting audio frame will have no samples. If this frame goes
through a signed linear translation path an interpolated frame can
be created (if generic packet loss concealment is enabled) that has
minimal data on it, including no src. If this frame is given to a
translation path a crash will occur due to the lack of src.

ASTERISK-28499

Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8

After receiving a 200 OK with a declined stream in response to a T.38
initiated re-invite Asterisk would crash when attempting to dereference
a NULL session media object.

This patch checks to make sure the session media object is not NULL before
attempting to use it.

ASTERISK-28495
patches:
  ast-2019-004.patch submitted by Alexei Gradinari (license 5691)

Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572

Module res_adsi.so is deprecated, therefore it does not load by default.
Module not loaded causes it to yield a FAIL when tested by tests/test_utils.c.
This fix checks if the corresponding module is loaded at the start of the test,
and if not, it passes the test and exits with a message.

This fix is applied to all versions where the module is marked deprecated.

Change-Id: I52be64c8f6af222e15148a856d1f10cb113e1e94

On reading information about initial client packet unistim use dirty
implementation of destination ip address retrieval. This fix uses
CMSG_*(..) to get ip address and make clang compile without warning.

ASTERISK-25592 #close
Reported-by: Alexander Traud

Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1

res_pjsip_mwi allows both solicited and unsolicited MWI subscription types.
While both can be set in the configuration for a given endpoint/aor, only
one is allowed. Precedence is given to unsolicited. Meaning if an endpoint/aor
is configured to allow both types then the solicited subscription is rejected
when it comes in. However, there is a configuration option to override that
behavior:

mwi_subscribe_replaces_unsolicited

When set to "yes" then when a solicited subscription comes in instead of
rejecting it Asterisk is suppose to replace the unsolicited one if it exists.
Prior to this patch there was a bug in Asterisk that allowed the solicted one
to be added, but did not remove the unsolicited. As a matter of fact a new
unsolicited subscription got added everytime a SIP register was received.
Over time this eventually could "flood" a phone with SIP notifies.

This patch fixes that behavior to now make it work as expected. If configured
to do so a solicited subscription now properly replaces the unsolicited one.
As well when an unsubscribe is received the unsolicited subscription is
restored. Logic was also put in to handle reloads, and any configuration changes
that might result from that. For instance, if a solicited subscription had
previously replaced an unsolicited one, but after reload it was configured to
not allow that then the solicited one needs to be shutdown, and the unsolicited
one added.

ASTERISK-28488

Change-Id: Iec2ec12d9431097e97ed5f37119963aee41af7b1

Current implementation of ast_channel_tech send_digit_begin hook uses
same function for tone playback as key press handler. This cause every
incoming dtmf send back to asterisk. In case of two unistim phones
connected to each other, it'll cause indefinite DTMF loop. Fix add
separate function for dtmf tone phone play.

Change-Id: I5795db468df552f0c89c7576b6b3858b26c4eab4

This patch fixes one-way oudio that users expirienced on
big-endian architechtires. RTP port number bytes was stored
in improper order and phone sent RTP to wrong RTP port.

Reported-by: Andrey Ionov
Change-Id: I9a9ca7f26e31a67bbbceff12923baa10dfb8a3be

ASTERISK-28511 #close

Change-Id: Idd07bf341e89ac999c7f5701d9b72b8a9cb11e82

Change-Id: Ic784be8500e5cb75dcb34bae9f03cfd93b6b34fb

The UnicastRTP channel driver provided by chan_rtp now accepts
"<hostname>:<port>" as an alternative to "<ip_address>:<port>"
in the destination. The first AAAA (preferred) or A record resolved
will be used as the destination. The lookup is synchronous so beware
of possible dialplan delays if you specify a hostname.

Change-Id: Ie6f95b983a8792bf0dacc64c7953a41032dba677

The new function takes in a pointer to an ast_sockaddr structure,
a hostname and an optional port and then dispatches parallel
"AAAA" and "A" record queries.  If an "AAAA" record is returned,
it's parsed into the ast_sockaddr structure along with the port
if it was supplied.  If no "AAAA" record was returned, the
first "A" record returned (if any) is parsed instead.

This is a synchronous call.  If you need asynchronous lookups,
use ast_dns_query_set_resolve_async and roll your own.

Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95

NEC SIP Station interface with authenticated registration only supports cnonce
up to 32 characters.  In Linux, PJSIP would generate 36 character cnonce
which included hyphens.  Teluu developed this patch adding a compile time
setting to default to not include the hyphens.  They felt it best to still
generate the UUID and strip the hyphens.
They have indicated it will be part of PJSIP 2.10.

ASTERISK-28509
Reported-by: Dan Cropp

Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470

Given the following request path and 2 handler paths...
Request: /channels/externalMedia
Handler: /channels/{channelId}      "wildcard"
Handler: /channels/externalmedia    "non-wildcard"

...if /channels/externalMedia was registered as a handler after
/channels/{channelId} as shown above, the request would automatically
match the wildcard handler and attempt to parse "externalMedia" into
the channelId variable which isn't what was intended.  It'd work
if the non-wildard entry was defined in rest-api/api-docs/channels.json
before the wildcard entry but that makes the json files
order-dependent which isn't a good thing.

To combat this issue, the search loop saves any wildcard match but
continues looking for exact matches at the same level.  If it finds
one, it's used.  If it hasn't found an exact match at the end of
the current level, the wildcard is used.  Regardless, after
searching the current level, the wildcard is cleared so it won't
accidentally match for a different object or a higher level.

BTW, it's currently not possible for more than 1 wildcard entry
to be defined for a level.  For instance, there couldn't be:
Handler: /channels/{channelId}
Handler: /channels/{channelName}
We wouldn't know which one to match.

Change-Id: I574aa3cbe4249c92c30f74b9b40e750e9002f925