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

ASTERISK-27547

Change-Id: I1a88a75b38a2b1dd9d1de6c0307b20a3f584c817

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

This patch does three things associated with the initial incoming INVITE
request URI.

1) Add access to the full initial incoming INVITE request URI.

2) We were not setting DNID on incoming PJSIP channels.  The DNID is the
user portion of the initial incoming INVITE Request-URI.  The value is
accessed by reading CALLERID(dnid).

3) Fix CHANNEL(pjsip,target_uri) documentation.

* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).

* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.

* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.

* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.

ASTERISK-27478

Change-Id: I512e60d1f162395c946451becb37af3333337b33

Support for these protocols was added in the same commit as the 'proto'
field, so we can safely use the same ./configure check.

For reference: https://trac.pjsip.org/repos/changeset/4968

Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac

Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d

A couple of places were setting the status to "UNKNOWN" when qualifies were
being disabled. Instead this should be set to the "CREATED" status that
represents when a contact is given (uri available), but the qualify frequency
is set to zero so we don't know the status.

This patch updates the relevant places with "CREATED". It also updates the
"CREATED" status description (value shown in CLI/AMI/ARI output) to a value
of "NonQualified"/"NonQual" as this description is hopefully less confusing.

ASTERISK-27467

Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89

Use the new ast_cli_completion_add() function to improve completion
performance for commands like 'pjsip show endpoint.'

Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348

Using the LIKE operator requires a full table scan of 'astdb', whereas a
comparison operation is able to use the primary key index.

This patch adds a new function to the AstDB API for quick prefix matches
and updates res_sorcery_astdb to utilize it. This showed substantial
performance improvement in my test environment.

Related to ASTERISK~26806, but does not completely resolve it.

Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1

Change-Id: I25348c386a222bb704aff07f54375108a6402906

There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.

Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7

res_stasis was missing AST_MODFLAG_LOAD_ORDER.  Set res_stasis and
res_speech to start at (AST_MODPRI_APP_DEPEND - 1) so they are ready for
dependent modules.

Change-Id: I27f4f3810a95b6be8a5bfbf62be2ace6bfab6ff3

For both dynamic and static contacts it was possible that potential AOR
changes were not being applied to all contacts. This was because the qualify
and schedule code was only retrieving AOR's, and contacts with frequencies
greater than zero.

For instance the following could happen: and AOR/contact has a frequency of 5,
it then gets set to 0, and then a reload occurs. All scheduled OPTIONS are
stopped, a list of AOR's is retrieved with frequency > 0, but none are
selected since in this scenario all are 0. The contact for the one previously
set to 5 though does not get updated, so it's status remains "AVAILABLE".

This patch makes it so all contacts (static and dynamic) are selected, and
appropriately updated if need be.

ASTERISK-27467 #close

Change-Id: I7a920170f89c683af9505d4723a44fc6841decdb

Dynamic contacts were not being properly updated on reload. As a matter of
fact any changes to the AOR that a dynamic contact was associated with were
not being applied.

On reload, this patch makes it so for each dynamic contact, the associated
AOR is now retrieved and the AOR's fields are applied to the contact.

ASTERISK-27467

Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d

The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value.  This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object().  i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.

Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935

More complicated direct media reinvite negotiations can result in longer
delays before direct media flows.  The strictrtp learning timeout time
was too short.  One log showed that the first RTP packet came in just
after three seconds.

* Increase the strictrtp learning timeout time from 1.5 to 5 seconds.

ASTERISK-27453

Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c

This change makes the presence of the GMIME_MAJOR_VERSION
definition optional, as not all versions of gmime actually
define it.

ASTERISK-27454

Change-Id: I01d99590045971ed6787899147170a5954077238

The patch for ASTERISK_24560 inverted a test checking if the bridge name
is being updated to a different name.

* Fix the test to return "Changing bridge name is not implemented" when
someone attempts to change the bridge name.

ASTERISK-27445

Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9

Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now,
srflx is gathered and sent via SDP, even when Asterisk is enabled for
Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used.

ASTERISK-27437

Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1

res_parking had an inplicit load_pri of 0 meaning it was one of the very
first modules loaded after modules with global symbols.  Set it to
AST_MODPRI_DEVSTATE_PROVIDER as it provides device state for parking
lots.

Change-Id: I297b6fb3ff6993ec004e667b22a74f5925906259

res_mwi_external_ami specified AST_MODFLAG_LOAD_ORDER but didn't set
load_pri, resulting in an actual load priority of 0.  This module only
provides AMI actions so it has no reason to load early.

Change-Id: I82987fcf10d3ea42716b2f9df915b16687fd5839

Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri
AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT.

Change-Id: I0123258eafce324249433a69df15a85cc16e509f

Mac doesn't like the comparison of -1 to an enum, so store the result of
ast_sip_str_to_dtmf to an int so we can check for the negative return
value.  ast_sip_str_to_dtmf returns an int so this is only delaying the
implicit type cast.

Change-Id: I0c262c1719ee951aae1f437d733a301cf5f8ad29

Some net-snmp builds do not provide the RONLY declare, only
NETSNMP_OLDAPI_RONLY.  Map RONLY to NETSNMP_OLDAPI_RONLY to get around
this error.

Change-Id: Ida5c7ad9406515825485c4d3b4a34fd6ad0da577

It's impossible for gwtimeout or fdtimeout to be less than 0 because
they are unsigned int's.  Remove checks and unreachable branches.

Change-Id: Ib2286960621e6ee245e40013c84986143302bc78