Add an AMI action which provides information on all
configured Contacts.

ASTERISK-27581

Change-Id: I2eed42c74bbc725fad26b8b33b1a5b3161950c73

Change-Id: I67ed9039bf3f132fb20ee7a750e0aef0f704d7d3

The type=identify endpoint identification method can match by IP address
and by SIP header.  However, the SIP header matching has limited
usefulness because you cannot specify the SIP header matching priority
relative to the IP address matching.  All the matching happens at the same
priority and the order of evaluating the identify sections is
indeterminate.  e.g., If you had two type=identify sections where one
matches by IP address for endpoint alice and the other matches by SIP
header for endpoint bob then you couldn't predict which endpoint is
matched when a request comes in that matches both.

* Extract the SIP header matching criteria into its own "header" endpoint
identification method so the user can specify the relative priority of the
SIP header and the IP address matching criteria in the global
endpoint_identifier_order option.  The "ip" endpoint identification method
now only matches by IP address.

ASTERISK-27491

Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095

* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.

Still need to investigate dependencies among modules I cannot compile.

Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf

We did this for TCP transports already but I'm not sure why we
didn't do it for TLS transports.

ASTERISK_27474 #not_final_fix

Change-Id: I5b1ef4b882f7b859e718236686b7898751dbb262

If any component of ast_config_AST_RECORDING_DIR is a symbolic link we
would incorrectly assume the ARI user was trying to escape the recording
path.  Create additional check to check the recording directory's
realpath, only deny access if both do not match.

This is needed by the testsuite when run by 'run-local'.

Change-Id: I9145e841865edadcb5f75cead3471ad06bbb56c0

clang 5.0 warned about this.

ASTERISK-27576

Change-Id: If41f400a51973c06cdb9b75462e535b616bfe385

* Endpoint identify_by documentation.
* IP/Header endpoint identifier documentation.

Change-Id: Id92f00b495acca7be945daf749d2abd7f76a0b5a

The requirement that "ip" must be in the endpoint identify_by list to
allow the type=identify method to identify the endpoint is not necessary.
The "ip" identifier method can match one and only one endpoint.  To even
work, the "ip" identifier method configuration must explicitly specify the
identified endpoint.  Therefore, why bother configuring the type=identify
identifier in the first place?  The requirement only adds the potential
for configuration errors for no benefit.  Even worse, those configuration
errors cannot be detected when the configuration loads.  The requirement
was introduced with the ASTERISK_27206 patch.

* Remove the code change that enforces the requiremnt.  Listing the "ip"
method in the identify_by value is simply documentation.

Change-Id: Ia057f92a33fb5d9f51dc5d5692e3d5ee1a6f2c11

* Extracted sip_endpoint_identifier_type2str() and
sip_endpoint_identifier_str2type() to simplify the calling functions.

* Fixed pjsip_configuration.c:ident_to_str() building the endpoint's
identify_by value string.

Change-Id: Ide876768a8d5d828b12052e2a75008b0563fc509

The AMI PJSIPShowEndpoint action could only list one IdentifyDetail AMI
event per endpoint.  However, there is no reason that multiple
type=identify sections cannot identify the same endpoint.

* Reworked format_ami_endpoint_identify() to generate as many
IdentifyDetail AMI events as there are matching identifiers.

Change-Id: Ie146792aef72d78e05416ab5b27bc552a30399db

Some WebRTC clients can't handle renegotiation with the addition of
streams that include an offer to bundle. They instead expect the
newly added streams to already be bundled. This change does such a thing
if WebRTC support is enabled on an endpoint.

ASTERISK-27566

Change-Id: I7fe9b7ac35a2798627d9c2c8369129f407af6461

In addition to being a micro-optimization (RAII_VAR has overhead), this
change improves output of REF_DEBUG.  Unfortunately when RAII_VAR calls
ao2_cleanup it does so from a generated _dtor_varname function.  For
example this caused _dtor_app to release a reference instead of
__stasis_app_unregister.

Change-Id: I4ce67120583a446babf9adeec678b71d37fcd9e5

Add an AMI action which provides information on all
configured Auths.

ASTERISK-27547

Change-Id: I1a88a75b38a2b1dd9d1de6c0307b20a3f584c817

If the dial bridge has been created it must be released by calling
ast_bridge_destroy, simply releasing the ao2 reference is not enough.

Also move stasis_app_control_shutdown earlier in unload to ensure the
bridge cannot be created or grabbed after the app_bridges container is
released.

Change-Id: I372302de94ca63876069e2585a049c5060e5e767

Instead of searching for bridge_id provided in an argument this function
always searched for BRIDGE_ALL first.  Rewrite this function to work
like the similar functions for channel and endpoint functions.

Change-Id: Ib5caca69e11727c5c8a7284a1d00621f40f1e60a

Some (normally optional) modules created notices, warnings, and even errors
in normal situations like (un)load. This cluttered the command-line interface
(CLI) on start and while stopping gracefully. However, when an user went for
the script './contrib/scripts/install_prereq', those modules get compiled-in
because their prerequisites were met at compile time. Furthermore, because of
ASTERISK_27475, the former talkative module 'res_curl' is built as side-effect.

ASTERISK-27553

Change-Id: I9f105f46d72553994e820679bfde3478a551b281

clang 5.0 warned about this.

ASTERISK-27557

Change-Id: I7cceaa88e147cbdf81a3a7beec5c1c20210fa41e

The ip_identify_apply() did not validate the configuration for simple
static configuration errors or deal well with address resolution errors.

* Added missing configuration validation checks.
* Fixed address resolution error handling.
* Demoted an error message to a warning since it does not fail applying
the identify object configuration.

Change-Id: I8b519607263fe88e8ce964f526a45359fd362b6e

If an endpoint identifier name in the endpoint_identifier_order list is a
prefix to the identifier we are registering, we could install it in the
wrong position of the list.

Assuming
endpoint_identifier_order=username,ip,anonymous

then registering the "ip_only" identifier would put the identifier in the
wrong position of the priority list.

* Fix incorrect strncmp() string prefix matching.

Change-Id: Ib8819ec4b811da8a27419fd93528c54d34f01484

Change-Id: I872060a30543776a176a316309602d924a23eb29

This function returns NULL if the module in question is not running.  I
did not change ast_module_ref as most callers do not check the result
and they always call ast_module_unref.

Make use of this function when running registered items from:
* app_stack API's
* bridge technologies
* CLI commands
* File formats
* Manager Actions
* RTP engines
* Sorcery Wizards
* Timing Interfaces
* Translators
* AGI Commands
* Fax Technologies

ASTERISK-20346 #close

Change-Id: Ia16fd28e188b2fc0b9d18b8a5d9cacc31df73fcc

The pjsip_msg_find_hdr function can return NULL. This patch adds a check
when searching for the sequence header to make sure a NULL pointer is never
de-referenced.

Change-Id: I19af23aeeded65be016be92360e8cb7ffe51fad2

Add an AMI action which provides information on all
configured AORs.

ASTERISK-27537

Change-Id: If8b990a00909e5b6c0f04a3b8dccd9903dc445eb

Per RFC 5245, the foundation specified with an ICE candidate can be up
to 32 characters but we are only allowing for 31.

ASTERISK-27498 #close
Reported by: Michele Prà

Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf

Those SIP messages that create dialogs require a contact header to be present.
If the contact header was missing from the message it could cause Asterisk to
crash.

This patch checks to make sure SIP messages that create a dialog contain the
contact header. If the message does not and it is required Asterisk now returns
a "400 Missing Contact header" response. Also added NULL checks when retrieving
the contact header that were missing as a "just in case".

ASTERISK-27480 #close

Change-Id: I1810db87683fc637a9e3e1384a746037fec20afe

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh

ASTERISK-24198 #close

Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31

Some variables are set and never changed, making them constant.  This
means that code in the 'false' block of the conditional is unreachable.

In chan_skinny and res_config_ldap I used preprocessor directive `#if 0`
as I'm unsure if the unreachable code could be enabled in the future.

Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059

When RTCP-MUX enabled. rtp->s is the same as rtcp->s, check this before
close the file descriptor. Close the FD twice will hangs the asterisk
under heavy load.

ASTERISK-27299 #close
Reported-by: Aaron An
Tested-by: AaronAn

Change-Id: I870a072d73fd207463ac116ef97100addbc0820a

Remove nearly all use of regex from ACO users.  Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
  callers use simple prefix based regex.  I haven't decided the best
  way to fix this in both 13/15 and master.

Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b

When adding shutdown refs for OPTIONAL_API components I accidentally
added it to the unload_module function in res_smdi.  Move it to
load_module.

Change-Id: I2b9da38fbc11ef78ea23dbb2df92b684be7f647c

res_hep_pjsip.so and res_hep_rtcp.so will still load and do a lot of
unnecessary work even if 'enabled' is set to 'no' in hep.conf.

Change-Id: I3eddfeea09c6b5bc7c641952ee0ae487fd09b64b

Never ignore contents of line when generating completion options.

Change-Id: I74389efdfea154019d3b56a9f381610614c044c8

We should not do flood detection on video RTP streams.  Video RTP streams
are very bursty by nature.  They send out a burst of packets to update the
video frame then wait for the next video frame update.  Really only audio
streams can be checked for flooding.  The others are either bursty or
don't have a set rate.

* Added code to selectively disable packet flood detection for video RTP
streams.

ASTERISK-27440

Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70

add_crypto_to_stream wasn't checking for a NULL
session->inv_session->neg before calling pjmedia_sdp_neg_get_state.
This was causing a crash if the negotiation hadn't already been
completed and asterisk was compiled with --enable-dev-mode.

Change-Id: I57c6229954a38145da9810fc18657bfcc4d9d0c9

Reset the samples counter to zero when we are done playing an
announcement so that we don't skip into the middle of the first file in
the playlist.

Also add the selected annoucement to the output of 'moh show classes.'

ASTERISK-24329 #close
Reported by: Thomas Frederiksen

Change-Id: I2a5f986a31279c981592f49391409ebf38d6f6d0

ASTERISK-19657 #close
Reported by: Matt Jordan III, Esq.

Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a

When the RTCP code was transitioned over to Stasis a code change
was made to keep track of how many reports are present. This count
controlled where report blocks were placed in the RTCP report.

If a compound RTCP packet was received this logic would incorrectly
place a report block in the wrong location resulting in a write
to an invalid location.

This change removes this counting logic and always places the report
block at the first position. If in the future multiple reports are
supported the logic can be extended but for now keeping a count
serves no purpose.

ASTERISK-27382
ASTERISK-27429

Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116

When a connected line update is sent to an endpoint we do not request
a specific stream topology to be used. Previously this resulted in the
configured stream topology being used which may actually differ from the
currently negotiated topology. PJSIP is helpful in this regard in that
it will fill in any missing streams with removed ones. This results in
our own state not matching the SDP, though, and we do not apply the
negotiated SDP.

This change tweaks the code to use the actively negotiated stream
topology if it is present with a fallback to the configured one. This
results in the SDP and the state having matching information and the
world is happy.

ASTERISK*27397

Change-Id: I7a57117f0183479e6884b7bf3a53bb8c7464f604