Previously only the canary checking thread itself had its priority set
to SCHED_OTHER. Now all threads are traversed and adjusted.

ASTERISK-19867 #close
Reported by: Xavier Hienne

Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39

Creating ODBC SQL queries resulted in queries too large to fit into the
supplied buffer.  The resulting truncated buffer contained an invalid SQL
query.

* Made SQL query generation code use a thread storage buffer that can
increase in size as needed.

* Fixed bad multi-line warning messages.

ASTERISK-26263 #close
Reported by: Jeppe Ryskov Larsen

Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae

When receiving an SDP offer with multiple payloads for
the same format we would generate an answer with the first
payload, but during the payload crossover operation
(to set the payloads for receiving) we would remove all
payloads but the last. This would result in incoming
traffic being matched against the wrong format and outgoing
traffic being sent using the wrong payload.

This change makes it so that once a format has a payload
number put into the mapping all subsequent ones are ignored.
This ensures there is only ever one payload in the mapping
and that it is the payload placed into the answer SDP.

ASTERISK-26365 #close

Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60

A name server that returns "Server Failure" is indicating only that
the server couldn't process that particular request.  We should NOT
assume that the name server is incapable of serving other requests.

Here's the scenario we've been encountering...

* 2 local name servers configured in resolv.conf.
* An OPTIONS request causes a request for A and AAAA records to go out
  to both nameservers.
* The A responses both come back successfully resolved.
* Because of an issue at some upstream nameserver, the AAAA responses
  for that particular query come back as "SERVFAIL" from both local
  name servers.
* Both local servers are marked as bad and no further queries can be
  sent until the 60 second ttl expires.  Only previously cached results
  can be used.
* In this case, 60 seconds is just enough time for another OPTIONS
  request to go out to the same host so the cycle repeats.

We could set the bad ttl really low but that also affects REFUSED and
NOTAUTH which probably DO signal a real server issue.  Besides, even
a really low bad ttl would be an issue on a pbx.

Although we use our own resolver in 14 and master and don't have this
issue there, Teluu has merged this patch upstream so it's appropriate
to cherry-pick to 14 and master to keep pjproject consistent.


Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0

sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).

To use this, use a systemd unit with 'Type=notify' for Asterisk.

This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.

Also adds support for libsystemd detection in the configure script.

Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811

If sysinfo() is available, but not sysctl() or swapctl() the
printing code for swap buffer sizes is incorrectly omitted.
The above condition happens with musl c-library.

Fix #if rule to consider defined(HAVE_SYSINFO). And also
remove the redundant || defined(HAVE_SYSCTL) which was
incorrectly there to start with. Now swap information is
displayed only if an actual libc function to get it is
available.

This also fixes warnings previously seen with musl libc:

   [CC] asterisk.c -> asterisk.o
asterisk.c: In function 'handle_show_sysinfo':
asterisk.c:773:6: warning: variable 'totalswap' set but not used
 [-Wunused-but-set-variable]
  int totalswap = 0;
      ^~~~~~~~~
asterisk.c:770:11: warning: variable 'freeswap' set but not used
 [-Wunused-but-set-variable]
  uint64_t freeswap = 0;
           ^~~~~~~~

Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca

Currently when receiving video over RTP we store only
a calculated samples on the frame. When starting the video
it can take some time for this calculation to actually yield
a value as it requires constant changing timestamps. As well
if a video frame passes over multiple RTP packets this calculation
will fail as the timestamp is the same as the previous RTP
packet and the number of samples calculated will be 0.

This change preserves the timestamp on the frame and allows
it to pass through the core. When sending the video this timestamp
is used instead of a new one being calculated.

ASTERISK-26367 #close

Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd

ASTERISK-26375 #close

Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc

Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for
SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP
connections, allowing the TCP layer to handle the retransmits. Unfortunately,
this caused sessions to be terminated with a retransmit timeout becasue it
stopped at the point of the first retrans call.

This patch waits for the 64*T1 timer to expire instead.

ASTERISK-19968

Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204

When performing DNS resolution the failover code present in
res_pjsip currently assumes that a request will always have
at least one viable address. In practice this is not true.
A domain may be used that has no records.

The code now checks that at least one address exists on the
request which prevents looping.

ASTERISK-26364 #close

Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c

The output of CLI "queue show" and AMI Queues action is truncated and
"failed to extend from 240 to 327" messages are generated if the queue
member and interface names are lengthy.

* Increase the string buffer size from 240 to 512 in order to accommodate
for more information fields added to the output since v1.8.

ASTERISK-26360 #close
Reported by: Richard Mudgett

Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d

Previously, the Contact was stored only on initial INVITE and on any
18X and 200. That meant that after re-INVITEs from *us* the Contact
could get updated, but after re-INVITEs from the *peer*, it did not.

This changeset fixes this inconsistency, properly allowing target
refreshes through re-INVITES (RFC3261, 12.2).

If your strictrtp setting allows it, this change allows you to switch
the source IP of a connected/calling device mid-call with a simple
re-INVITE from the new IP.

ASTERISK-26358 #close

Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435

Map the sip.conf general section legacy_useroption_parsing to the
new pjsip.conf global ignore_uri_user_options.

ASTERISK-26316
Reported by: Kevin Harwell

Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc

This implements the chan_sip legacy_useroption_parsing option but with a
better name.

* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.

ASTERISK-26316 #close
Reported by: Kevin Harwell

Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62

If you use the safe_asterisk script, it uses hardcoded defaults before
running configurable values from /etc/asterisk/startup.d. The hardcoded
default has TTY=9. Some containerized environments don't have such a
TTY, and safe_asterisk would stop.

The custom configuration from /etc/asterisk/startup.d/* isn't read until
after it stopped, so changing TTY in a custom config did not help.

This changeset changes safe_asterisk to continue if the TTY setting was
untouched and /dev/tty9 and /dev/vc/9 aren't found.

Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc

The code was incorrectly invoking the unidentified logic when
an endpoint had actually been identified, causing log messages
to be output.

ASTERISK-26349 #close

Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f

This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.

ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn

Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762

The endpoint identification PJSIP module is intended to identify which
endpoint an incoming request is from. If an endpoint is not identified,
then an artificial endpoint is used in its place when proceeding.

The problem is that the ACK request type is an exception to the rule.
The artificial endpoint is not used when processing an ACK. This results
in the possibility of having a NULL endpoint being used further on.

The reason ACK is an exception is an attempt not to spam security logs
with unidentified requests. Presumably, you've already logged the
unidentified request on the preceeding INVITE.

Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion
didn't cause an issue. A new change in 13.10 added endpoint ACL checking
shortly after endpoint identification. Because we are accessing a NULL
endpoint, this ACL check resulted in a crash.

The fix here is to be sure to retrieve the artificial endpoint for all
request types. ACKs still do not generate unidentified request security
events.

ASTERISK-26264 #close
Reported by nappsoft

AST-2016-006

Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703

In some scenarios dialog_initialize_rtp can be called multiple times on
the same dialog.  This can cause RTP instances to be leaked along with
multiple file descriptors for each instance.

This change makes it so the existing RTP instances are destroyed and
not overwritten, stopping the memory leak.

ASTERISK-26272 #close
patches:
  ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909)

Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73

* Eliminated RAII_VAR in get_outbound_endpoint().

* Simplify update_to() coding.  However, this function can only be a NoOp
because the To string can only be a URI and not a name-address formatted
string.

* Simplify update_from() coding.  Also fixed a code path modifying the
from string when the caller could still want to use the original string.

* Fixed msg_data_create() incompletely removing the "pjsip:" to then add
back the "sip:" string if needed.  The code didn't handle the "pjsip:sip:"
case because it left the colon after pjsip in the string.

Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db

Currently when you add global headers from the dialplan both
the header in the dialplan and the globally configured header
are added to the resulting SIP INVITE. This change makes it
so the headers in the dialplan take precedence and are the
only ones added.

Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad