Change-Id: I9290115a1aaadb589eb1d02eaeb502eec01b31fa

FD_SET contains a conditional statement to protect against buffer
overruns.  The statement was overly complicated and prevented use
of the last array element of ast_fdset.  We now just verify the fd
is less than ast_FDMAX.

Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40

When terminating the threads thrashing a sorcery memory cache each
would be told to stop and then we would wait on them. During at
least one thrashing test this was problematic due to the specific
usage pattern in use. It would take some time for termination of the
thread to occur.

This would occur due to contention between the threads retrieving
and the threads updating the cache. As the retrieving threads are
given priority it may be some time before the updating threads
are able to proceed.

This change makes it so all threads are told to stop and then each
are joined to ensure they stop. This way all the threads should
stop at around the same time instead of waiting for one to stop,
the next to stop, then the next, and so on. As a result of this
the execution time for each thrash test is much closer to their
expected value than previously seen as well.

Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827

In case failed of command "realtime show pgsql status" show a message the data
of connection to more clear information in error.

Change-Id: Ia8e9e2400466606e7118f52a46e05df0719b6a29

Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98

A user cannot set new bridge options after the conference is created by
the first user.  Attempting to do so is documented as undefined behavior.

This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.

Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266

* changes:
  app_confbridge: Add ability to get the muted conference state.
  app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
  app_confbridge: Make non-admin users join a muted conference muted.

The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
these options, which can cause problems on systems with older OpenSSL
installations.

This commit adds a configure script check for those defines and will not
attempt to make use of those if they do not exist. We will print a
warning urging the user to upgrade their OpenSSL installation if those
defines are not present.

Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d

ps_systems needed disable_tcp_switch
ps_registrations needed line and endpoint

ASTERISK-25737 #close

Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19

Device state subscription lifetimes were governed by when the
subscription was established and unsubscribed from. However, it is
possible that at the time of unsubscription, there could be device state
events still in flight. When those device state events occur, the device
state callback could attempt to dereference a freed pointer. Crash.

This change ensures that the lifetime of the device state subscription
does not end until the underlying stasis subscription has confirmed that
its final message has been sent.

Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2

During ICE negotiation the IPs of the local interfaces are sent to the remote
peer as host candidates. In many cases Asterisk is behind a static one-to-one
NAT, so these host addresses will be internal IP addresses.

To help in hiding the topology of the internal network, this patch adds the
ability to override the host candidates by matching them against a
user-defined list of replacements.

Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f

This change exposes the configuration of various aspects of the TLS
support and sets the default to the modern standards.

The TLS cipher is now set to the best values according to the
Mozilla OpSec team, different TLS versions can now be disabled, and
the cipher order can be forced to be that of the server instead of
the client.

ASTERISK-24972 #close

Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8

Sending UDPTL packets to Asterisk with the right amount of missing
sequence numbers and enough redundant 0-length IFP packets, can make
Asterisk crash.

ASTERISK-25603 #close
Reported by: Walter Doekes

ASTERISK-25742 #close
Reported by: Torrey Searle

Change-Id: I97df8375041be986f3f266ac1946a538023a5255

Setting the sip.conf timert1 value to a value higher than 1245 can cause
an integer overflow and result in large retransmit timeout times.  These
large timeout times hold system file descriptors hostage and can cause the
system to run out of file descriptors.

NOTE: The default sip.conf timert1 value is 500 which does not expose the
vulnerability.

* The overflow is now detected and the previous timeout time is
calculated.

ASTERISK-25397 #close
Reported by: Alexander Traud

Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290

test_dlinklists doesn't need to NOTICE everyone that every macro worked.

res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or
provider was registered.

res_odbc was missing a newline at the end of one message.

Change-Id: I6c06361518ef3711821795e535acd439782a995e

A regression was introduced where searching for realtime PJSIP objects
by regex by starting the regex with a leading "^" would cause no items
to be returned.

This was due to a change which attempted to drop the requirement for a
leading "^" to be present due to how some CLI commands formulate their
regexes. However, the change, rather than simply eliminating the
requirement, caused any regexes that did begin with "^" to end up not
returning the expected results.

This change fixes the problem by inspecting the regex and formulating
the realtime query differently depending on if it begins with "^".

ASTERISK-25702 #close
Reported by Nic Colledge

Patches:
    realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691

Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693

The module res_xmpp does not accept usernames in the form used in component
mode (XEP-0114). In component mode there is no @something in the name.
In component mode the connection is now not dropped anymore.

If the xmpp server sends out a "stream" tag before handshake is finished,
the connection gets dropped in res_xmpp. Now this tag will be ignored and
the connection will be established.

After connecting there will be an exchange of presence states. This does
not work as expected in component mode. The responsible function
"xmpp_pak_presence" is left before the states get sent out. Sending
presence states in component mode is now moved to the top of the function.

ASTERISK-25735 #close

Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c

Fix some warnings found with clang.

Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd

A recent commit set qualify_timeout to Decimal which isn't supported.
This path corrects it to Float.

Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf

When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a)
AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect
asterisk to include the same value for its own ip in both cases a) and b),
but it seems a) produces a contact header like Contact:
<sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like
<sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf

My guess is that manager_sipnotify should call
ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does,
because after applying this patch, both cases a) and b) produce
the contact header that I expect: <sip:asterisk@192.168.1.227:8060>

Reported by: Stefan Engström
Tested by: Stefan Engström

Change-Id: I86af5e209db64aab82c25417de6c768fb645f476