A patch made to the master branch (Now the 14 branch) inadvertently made
libsrtp a required dependency in order to compile Asterisk.  Rather than
create dummy defines to substitute for the defines supplied by libsrtp
when libsrtp is not available, most of the code in sdp_srtp.c is moved
into res_srtp.c.  This gets more code out of Asterisk's core that isn't
used when SRTP is not available.  This also makes another inadvertent
required dependency on libsrtp by Asterisk's core unlikely.

ASTERISK-26253 #close
Reported by: Ben Merrills

Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7

If both channels which should be masqueraded
are in the same serializer:
1st channel will be locked waiting condition 'complete'
2nd channel will be locked waiting condition 'suspended'

On heavy load system a chance that both channels will be in
the same serializer 'pjsip/distibutor' is very high.

To reproduce compile res_pjsip/pjsip_distributor.c with
DISTRIBUTOR_POOL_SIZE=1

Steps to reproduce:
1. Party A calls Party B (bridged call 'AB')
2. Party B places Party A on hold
3. Party B calls Voicemail app (non-bridged call 'BV')
4. Party B attended transfers Party A to voicemail using REFER.
5. When asterisk masquerades calls 'AB' and 'BV',
   a deadlock is happened.

This patch adds a suspension indicator to the taskprocessor.
When a session suspends/unsuspends the serializer
it sets the indicator to the appropriate state.
The session checks the suspension indicator before
suspend the serializer.

ASTERISK-26145 #close

Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b

The extensions table defined two columns (id and priority) as primary key
autoincrement columns. However only one is allowed when defining the primary
key.

This patch removes the autoincrement attribute from the priority column since
it does not need to be as such and really should not have been on there in the
first place.

This patch also removes 'context', 'exten', and 'priority' from the primary key
index and creates a new combined unique contraint index on them.

ASTERISK-26183 #close

Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b

When an RTCP packet is sent or received, res_rtp_asterisk generates a
Stasis event that contains the RTCP report as well as the local and
remote addresses that the report pertains to.

The addresses are determined using ast_find_ourip(). For the local
address, this will typically result in a lookup of the hostname of the
server, and then a DNS lookup of that hostname. If you do not have the
host in /etc/hosts, then this results in a full DNS lookup, which can
potentially block for some time.

This is especially problematic when performing RTCP reads, since those
are done on the same thread responsible for reading and writing media.

This patch addresses the issue by performing a lookup of the local
address when RTCP is allocated. We then use this cached local address
for the Stasis events when necessary.

ASTERISK-26280 #close
Reported by Mark Michelson

Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556

Modules must define AST_MODULE_SELF_SYM to be used as the name of a
generated function.  This produces a friendly error when it's not
defined.

ASTERISK-26278 #close

Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b

The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.

This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.

This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.

ASTERISK-26230 #close

Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a

On heavy loaded system with IMAP or DB storage,
'app_voicemail' taskprocessor queue could reach 500 scheduled tasks.
It could happen when the IMAP or DB server dies or is unreachable.
It could happen on startup when there are many (thousands)
realtime endpoints configured with unsolicited mwi.
If the taskprocessor queue reaches the high water level
then the alert is triggered and pjsip stops processing new requests
until the queue reaches the low water level to clear the alert.

This patch adds 2 new 'general' configuration options
to tune taskprocessor alert levels:
'tps_queue_high' - Taskprocessor high water alert trigger level.
'tps_queue_low' - Taskprocessor low water clear alert level

ASTERISK-26229 #close

Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8

This change replaces the custom unload process for the outbound
publish module with the common serializer shutdown group.

ASTERISK-25217 #close

Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6

This file was out of sync with the current ARI definitions.

Change-Id: Ie7cb7d6d3c2eeb9cc9d683ca87b43b117e713d0a

This ensures startup is canceled due to allocation failures from the
following initializations.
* channel.c: ast_channels_init
* config_options.c: aco_init

ASTERISK-26265 #close

Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611

The regular expression would match causing the code that handled
the line if it was merely a comment to never get executed.

Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819

This changes the use of an empty regex for both res_sorcery_config
and res_sorcery_memory to "." instead. This is a more compatible
regular expression which also works on FreeBSD.

ASTERISK-26206 #close

Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388

ASTERISK-26256 #close

Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058

Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe

SILK is a bit of a hog when it comes to using up our limited number of
dynamic payload types in the RTP engine. By freeing up four slots, it
allows for other codecs to potentially take the place.

Now, codec_silk.so will dynamically use the payload slots in the RTP
engine when it loads.

A better fix would be make RTP dynamic payload types actually
dynamic. However, at this stage of Asterisk 14 development, this is a
risky move that would be imprudent.

Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612

Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5

This adds a two strings to ast_exten.  name to go with exten and
cidmatch_display to go with cidmatch.  The new fields contain input used
to add the extension in the first place.  The existing fields now
contain stripped input that excludes insignificant spaces and dashes.
These stripped fields should always be used for comparisons.  The
unstripped fields should normally be used for display, but displaying
stripped values will not cause runtime errors.

Note the actual string is only stored twice if it contains dashes.  If
no dashes are found then both 'char *' fields point to the same memory.
So this change has a minimum effect on memory usage.

The existing functions ast_get_extension_name and
ast_get_extension_cidmatch return unstripped values as they did before
this change.  Other similar bugs likely still exist where unstripped
extensions are saved outside pbx.c then passed back in.

ASTERISK-26233 #close

Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f

We can allow dangerous functions when adding a hint since altering
dialplan is itself a privileged activity.  Otherwise, we could never
execute dangerous functions.

ASTERISK-25996 #close
Reported by: Andrew Nagy

Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba

This patch fixes the issue in pjsip_tx_data_dec_ref()
when tx_data_destroy can be called more than once,
and checks if invalid value (e.g. NULL) is passed to.

This patch updates array limit checks and docs
in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability().

Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40