When fixing ASTERISK~24212, a change was done so a scheduled callback could not
be removed while it was running. The caller of ast_sched_del would have to wait.

However, when the caller of ast_sched_del is the callback itself (however wrong
this might be), this new check would cause a deadlock: it would wait forever
for itself.

This changeset introduces an additional check: if ast_sched_del is called
by the callback itself, it is immediately rejected (along with an ERROR log and
a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the
after-ast_sched_del-refcall function is only run if ast_sched_del returned
success.

This should fix the following spurious race condition found in chan_sip:
- thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE)
- thread 2: run sip_poke_peer_now
- thread 2: blank out sched-ID (too soon!)
- thread 1: set sched-ID (too late!)
- thread 2: try to delete the currently running sched-ID

After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor
excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and
other madness) should occur.

(Thanks Richard Mudgett for reviewing/improving this "scary" change.)

Note that this change does not fix the observed race condition: unlocked
access to peer->pokeexpire (and potentially other scheduled items in chan_sip),
causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the
deadlock go away. And in the observed case, it will not have adverse affects
(like memory leaks) because the scheduled item is removed through a different
path.

ASTERISK-28282

Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856

The DTLS MTU variable is not dependent on pjproject and should
not exist in its block.

Change-Id: I7e97d64dc192f2ac81bfe2b72b8229d321c7d026

We now check that a body exists and it has a length > 0 before
attempting to process it.

ASTERISK-28447
Reported-by: Gil Richard

Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f

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

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

Where possble, hostname and port has been added to error
messages, mostly on the server side.

ASTERISK-26006
Reported by: Oleksandr Natalenko

Change-Id: Iff4f897277bc36ce8c5b493b71d0a4a7b74e62f0

Most SSL/TLS error messages coming from pjproject now have either
the peer address:port or peer hostname, depending on what was
available at the time and code location where the error was
generated.

ASTERISK-28444
Reported by: Bernhard Schmidt

Change-Id: I41770e8a1ea5e96f6e16b236692c4269ce1ba91e

Relies on https://github.com/asterisk/third-party/pull/4

Change-Id: Iec9cad42cb4ae109a86a3d4dae61e8bce4424ce3

A few more format truncation issues addressed.

Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a

Given the non-default configuration of enabling ICE support on an
endpoint that does not result in an ICE negotiation occurring the
ICE candidates would be leaked.

This change makes it so that the ICE candidates are only retrieved
if ICE negotiation is occurring.

ASTERISK-28460

Change-Id: I7b3f76f031c41fb8a3dc3ef1a84b77e2a8cb969f

According T.38 Gateway 'Use case 3'
https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway
T.38 Gateway should send T.38 negotiation request to called endpoint
if FAX preamble (using V.21 detector) generated by called endpoint.
But it does not, because fax_gateway_detect_v21 constructs T.38
negotiation request, but forwards it only to other channel,
not to the channel on which FAX preamble is detected.

Some SIP endpoints could be improperly configured to rely on the other side
to initiate T.38 re-INVITEs.

With this patch the T.38 Gateway tries to negotiate with both sides
by sending T.38 negotiation request to both endpoints supported T.38.

Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39

We were using the presence of /usr/lib64 to determine where
shared libraries should be installed.  This only existed on
Redhat based systems and was safe.  If it existed, use it,
otherwise use /usr/lib.

Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT
NOT INCLUDE IT IN THE DEFAULT ld.so.conf.  So if anything is
installed there, it won't work.

The new method, just looks for $ID in /etc/os-release and if it's
centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib.

NOTE:  This applies only to the CI scripts.  Normal asterisk
build and install is not affected.

Change-Id: Iad66374b550fd89349bedbbf2b93f8edd195a7c3

There is WARNING "no samples for ..." on each Playtones.
The function ast_playtones_start calls ast_activate_generator,
which calls ast_prod.
The function ast_prod calls ast_write with empty audio frame.
In this case it's spam log.

Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660

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

When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.

Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1

This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.

This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.

ASTERISK-28018

Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06

AttendedTransfer queues up attended transfer to the given extension.

This application can be useful with Custom Dynamic Features.
For example to make attended transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_atxfer
TRANSFER_CONTEXT=my_transfer

[my_atxfer]
exten => s,1,AttendedTransfer(1234567890)
   same => n,Return()

[my_transfer]
include => default
;;;

This application also can be used to completly redefine Attended transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
atxfer => *7

[applicationmap]
custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_atxfer
TRANSFER_CONTEXT=my_transfer

[custom_atxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,AttendedTransfer(${dest})
   same => n,Return()

[my_transfer]
include => default
;;;

Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b

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

BlindTransfer redirects all channels currently bridged to the
caller channel to the specified destination.

This application can be useful with Custom Dynamic Features.
For example to make blind transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_blindxfer

[my_blindxfer]
exten => s,1,BlindTransfer(1234567890,default)
   same => n,Return()
;;;

This application also can be used to completly redefine Blind transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
blindxfer =>

[applicationmap]
custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_blindxfer

[custom_blindxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,BlindTransfer(${dest},default)
   same => n,Return()
;;;

Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a

Fixes an error occurring in function pgsql_reconnect() caused when value of
hostname is blank. Which in turn will cause the connection string to look
like this: "host= port=xx", which creates a sintax error. This fix now checks
if the corresponding values for host, port, dbname, and user are blank. Note
that since this is a reconnect function the database library will replace any
missing value pairs with default ones.

ASTERISK-28435

Change-Id: I0a921f99bbd265768be08cd492f04b30855b8423

The change #10017 "Handle fax gateway being started more than once"
introdiced a bug which leads to segfault in res_fax_spandsp.

The res_fax_spandsp module does not support reserving sessions, so
fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE.

The fax_gateway_start does not create a real fax session if the fax session
is already present and the state is not AST_FAX_STATE_RESERVED.
But the "reserved" session created for res_fax_spandsp has state
AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting.

Then when fax_gateway_framehook is called and gateway T.38 state is
NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to
segfault, because session tech_pvt is not set, i.e. the tech session
was not initialized/started.

This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved"
session created for res_fax_spandsp will start.

This patch also adds extra check and log ERROR if tech_pvt is not set
before call tech->write.

ASTERISK-27981 #close

Change-Id: Ife3e65e5f18c902db2ff0538fccf7d28f88fa803