This patch fixes several issues reported by the lgtm code analysis tool:

https://lgtm.com/projects/g/asterisk/asterisk

Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:

* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards

Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25

During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.

ASTERISK-28578

Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f

ASTERISK-28086 #close

Change-Id: Ib3baadc89b9f0477a6f25a63861433812368c5ea

Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis

ASTERISK-28542 #close

Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8

This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.

Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.

ASTERISK-28512

Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2

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

When fax detection occurs on an outbound PJSIP channel the
redirect operation will result in a masquerade occurring and
the underlying channel on the session changing. The code
incorrectly relocked the new channel instead of the old
channel when returning. This resulted in the new channel
being locked indefinitely. The code now always acts on the
expected channel.

ASTERISK-28538

Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3

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

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 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

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

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

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

There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:

unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);

would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.

ASTERISK-28480

Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1

Change-Id: I94dc38920e6e77cc73062648f62fdd613d0d1452
Signed-off-by: Oron Peled <oron.peled@xorcom.com>

Change-Id: I452d6a853bcd8c6e194455b19e5e017713e9c0fe
Signed-off-by: Oron Peled <oron.peled@xorcom.com>

* This command show the MFC/R2 links

Change-Id: I213822e1b7ef9c05bd89a2ba62df8e0856ce9f84
Signed-off-by: Oron Peled <oron.peled@xorcom.com>

Change-Id: Ibcb2401515a58782a1488c0b9efbed201c3f3a17
Signed-off-by: Oron Peled <oron.peled@xorcom.com>

Otherwise, OpenR2 threads go crazy and consume almost all CPU resources

Change-Id: I10a41f617613fe7399c5bdced5c64a2751173f28
Signed-off-by: Oron Peled <oron.peled@xorcom.com>

Details:
  - The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2"
  - As a result, the memcmp() in dahdi_r2_get_link() always fails
  - This cause dahdi_r2_get_link() to create new link for every channel
    (instead of a new link for every ~30 channels)
  - With the fix, far less links are generated -- so we use far less threads

Change-Id: I7259dd6272f5e46e8a6c7f5bf3e8c2ec01b8c132
Signed-off-by: Oron Peled <oron.peled@xorcom.com>

The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.

If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.

This change removes this assumption.

ASTERISK-28465

Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87

The MWI core recently got some new API calls that make tracking MWI state
lifetime more reliable. This patch updates those modules that subscribe to
specific MWI topics to use the new API. Specifically, these modules now
subscribe to both MWI topics and MWI state.

ASTERISK-28442

Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd

Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous
patch introduced a variable of type unassigned long long which is 64-bits.
Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work
with 32-bit systems.

ASTERISK-28457

Change-Id: I9cef6b5f2d826fc5c93f2f6a1c997c4e3e6c93fe

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

A few more format truncation issues addressed.

Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a

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

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

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

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

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

ASTERISK-28412

Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e

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

When the dtmf_mode on an endpoint is configured as "auto_info"
Asterisk will produce an inband DTMF tone alongside an INFO
message when sending DTMF.

ASTERISK-28371

Change-Id: I1380b82f006e110a1b83fbb50c9873edd13a5d9a

The compiler complained about a couple of variables that weren't
initialized but were being used. Initializing them to NULL resolves the
warnings/errors.

ASTERISK-28362 #close

Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b

The next usage of PJSIP_PARSE_URI will crash asterisk
${PJSIP_PARSE_URI(tel:+1234567890,host)}
or
${PJSIP_PARSE_URI(192.168.1.1:5060,host)}

The function pjsip_parse_uri successfully parses then, but returns
struct pjsip_other_uri *.

This patch restricts parsing only SIP/SIPS URIs.

Change-Id: I16f255c2b86a80a67e9f9604b94b129a381dd25e

Passing negative intervals to the scheduler rips a hole in the
space-time continuum.

ASTERISK-25792 #close
Reported by: Paul Sandys

Change-Id: Ie706f21cee05f76ffb6f7d89e9c867930ee7bcd7

Add logical group at DAHDIChannel event
and create "dahdi_group" at CHANNEL function.

ASTERISK-28317

Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a

chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP.  This new flag allows chan_pjsip to have the same
behavior as chan_sip.

ASTERISK-28322 #close

Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a

strtok() uses a static buffer, making it not thread safe.

Also add a #define to cause a compile failure if strtok is used.

Change-Id: Icce265153e1e65adafa8849334438ab6d190e541

Currently, the pjsip show channelstats cli does not show channel's
stats after hits the invalid channel info. This makes hard to use
this cli. Changed to keep iterate after hits the invalid channel
info.

ASTERISK-28292

Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c