ChangeLog

2021-07-22 22:10 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.5.1 Released.

2021-06-14 13:28 +0000 [6c1aec36cb]  Kevin Harwell <kharwell@sangoma.com>

	* AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS

	  If an SSL socket parent/listener was destroyed during the handshake,
	  depending on timing, it was possible for the handling callback to
	  attempt access of it after the fact thus causing a crash.

	  ASTERISK-29415 #close

	  Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d

2021-05-10 17:59 +0000 [98e0b536d7]  Kevin Harwell <kharwell@sangoma.com>

	* AST-2021-008 - chan_iax2: remote crash on unsupported media format

	  If chan_iax2 received a packet with an unsupported media format, for
	  example vp9, then it would set the frame's format to NULL. This could
	  then result in a crash later when an attempt was made to access the
	  format.

	  This patch makes it so chan_iax2 now ignores/drops frames received
	  with unsupported media format types.

	  ASTERISK-29392 #close

	  Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1

2021-04-28 07:36 +0000 [4a525a8971]  Joshua C. Colp <jcolp@sangoma.com>

	* AST-2021-007 - res_pjsip_session: Don't offer if no channel exists.

	  If a re-INVITE is received after we have sent a BYE request then it
	  is possible for no channel to be present on the session. If this
	  occurs we allow PJSIP to produce the offer instead. Since the call
	  is being hung up if it produces an incorrect offer it doesn't
	  actually matter. This also ensures that code which produces SDP
	  does not need to handle if a channel is not present.

	  ASTERISK-29381

	  Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042

2021-06-24 12:50 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.5.0 Released.

2021-06-17 14:44 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.5.0-rc1 Released.

2021-06-17 09:39 +0000 [0747162d4f]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.5.0
2021-06-16 08:50 +0000 [702e1d33b5]  George Joseph <gjoseph@digium.com>

	* res_pjsip_messaging: Overwrite user in existing contact URI

	  When the MessageSend destination is in the form
	  PJSIP/<number>@<endpoint> and the endpoint's contact
	  URI already has a user component, that user component
	  will now be replaced with <number> when creating the
	  request URI.

	  ASTERISK_29404

	  Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5

2021-03-16 11:45 +0000 [804788037e]  Bernd Zobl <b.zobl@commend.com>

	* res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter

	  Set preferred transport when querying the local address to use in
	  filter_on_tx_messages(). This prevents the module to erroneously select
	  the wrong transport if more than one transports of the same type (TCP or
	  TLS) are configured.

	  ASTERISK-29241

	  Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6

2021-06-10 09:34 +0000 [2b174a38fe]  Naveen Albert <asterisk@phreaknet.org>

	* pbx_builtins: Corrects SayNumber warning

	  Previously, SayNumber always emitted a warning if the caller hung up
	  during execution. Usually this isn't correct, so check if the channel
	  hung up and, if so, don't emit a warning.

	  ASTERISK-29475

	  Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594

2021-05-22 07:53 +0000 [6b67821098]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Prevent module unloading in-use module.

	  The scenario where a channel still has an associated datastore we
	  cannot unload since there is a function pointer to the destroy and fixup
	  functions in play.  Thus increase the module ref count whenever we
	  allocate a datastore, and decrease it during destroy.

	  In order to tighten the race that still exists in spite of this (below)
	  add some extra failure cases to prevent allocations in these cases.

	  Race:

	  If module ref is zero, an LOCK or TRYLOCK is invoked (near)
	  simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if
	  in such a case the datastore is created *prior* to unloading being set
	  to true (first step in module unload) then it's possible that the module
	  will unload with the destructor being called (and segfault) post the
	  module being unloaded.  The module will however wait for such locks to
	  release prior to unloading.

	  If post that we can recheck the module ref before returning the we can
	  (in theory, I think) eliminate the last of the race.  This race is
	  mostly theoretical in nature.

	  Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-22 07:29 +0000 [6f303335d3]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Add "dialplan locks show" cli command.

	  For example:

	  arthur*CLI> dialplan locks show
	  func_lock locks:
	  Name                                     Requesters Owner
	  uls-autoref                              0          (unlocked)
	  1 total locks listed.

	  Obviously other potentially useful stats could be added (eg, how many
	  times there was contention, how many times it failed etc ... but that
	  would require keeping the stats and I'm not convinced that's worth the
	  effort.  This was useful to troubleshoot some other issues so submitting
	  it.

	  Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-22 07:42 +0000 [a3df5d7de8]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Fix memory corruption during unload.

	  AST_TRAVERSE accessess current as current = current->(field).next ...
	  and since we free current (and ast_free poisons the memory) we either
	  end up on a ast_mutex_lock to a non-existing lock that can never be
	  obtained, or a segfault.

	  Incidentally add logging in the "we have to wait for a lock to release"
	  case, and remove an ineffective statement that sets memory that was just
	  cleared by ast_calloc to zero.

	  Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-22 07:48 +0000 [6bd741b77d]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Fix requesters counter in error paths.

	  In two places we bail out with failure after we've already incremented
	  the requesters counter, if this occured then it would effectively result
	  in unload to wait indefinitely, thus preventing clean shutdown.

	  Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-25 10:36 +0000 [a611a0cd42]  Naveen Albert <asterisk@phreaknet.org>

	* app_originate: Allow setting Caller ID and variables

	  Caller ID can now be set on the called channel and
	  Variables can now be set on the destination
	  using the Originate application, just as
	  they can be currently using call files
	  or the Manager Action.

	  ASTERISK-29450

	  Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66

2021-06-10 16:24 +0000 [26059f8616]  Sean Bright <sean.bright@gmail.com>

	* menuselect: Fix description of several modules.

	  The text description needs to be the last thing on the AST_MODULE_INFO
	  line to be pulled in properly by menuselect.

	  Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832

2021-05-23 19:20 +0000 [a40e58a4da]  Naveen Albert <asterisk@phreaknet.org>

	* app_confbridge: New ConfKick() application

	  Adds a new ConfKick() application, which may
	  be used to kick a specific channel, all channels,
	  or all non-admin channels from a specified
	  conference bridge, similar to existing CLI and
	  AMI commands.

	  ASTERISK-29446

	  Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b

2021-06-02 08:11 +0000 [6873c5f3e4]  Naveen Albert <asterisk@phreaknet.org>

	* sip_to_pjsip: Fix missing cases

	  Adds the "auto" case which is valid with
	  both chan_sip dtmfmode and chan_pjsip's
	  dtmf_mode, adds subscribecontext to
	  subscribe_context conversion, and accounts
	  for cipher = ALL being invalid.

	  ASTERISK-29459

	  Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2

2021-06-02 08:25 +0000 [99573f9540]  Naveen Albert <asterisk@phreaknet.org>

	* res_pjsip_dtmf_info: Hook flash

	  Adds hook flash recognition support
	  for application/hook-flash.

	  ASTERISK-29460

	  Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea

2021-05-20 09:51 +0000 [a861522467]  Naveen Albert <mail@interlinked.x10host.com>

	* app_confbridge: New option to prevent answer supervision

	  A new user option, answer_channel, adds the capability to
	  prevent answering the channel if it hasn't already been
	  answered yet.

	  ASTERISK-29440

	  Change-Id: I26642729d0345f178c7b8045506605c8402de54b

2021-04-22 13:07 +0000 [8e2672d2a4]  George Joseph <gjoseph@digium.com>

	* res_pjsip_messaging: Refactor outgoing URI processing

	   * Implemented the new "to" parameter of the MessageSend()
	     dialplan application.  This allows a user to specify
	     a complete SIP "To" header separate from the Request URI.

	   * Completely refactored the get_outbound_endpoint() function
	     to actually handle all the destination combinations that
	     we advertized as supporting.

	   * We now also accept a destination in the same format
	     as Dial()...  PJSIP/number@endpoint

	   * Added lots of debugging.

	  ASTERISK-29404
	  Reported by Brian J. Murrell

	  Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce

2021-05-16 10:21 +0000 [9106c9d1f1]  Naveen Albert <mail@interlinked.x10host.com>

	* func_math: Three new dialplan functions

	  Introduces three new dialplan functions, MIN and MAX,
	  which can be used to calculate the minimum or
	  maximum of up to two numbers, and ABS, an absolute
	  value function.

	  ASTERISK-29431

	  Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d

2021-05-19 13:45 +0000 [26a38c4084]  Ben Ford <bford@digium.com>

	* STIR/SHAKEN: Add Date header, dest->tn, and URL checking.

	  STIR/SHAKEN requires a Date header alongside the Identity header, so
	  that has been added. Still on the outgoing side, we were missing the
	  dest->tn section of the JSON payload, so that has been added as well.
	  Moving to the incoming side, URL checking has been added to the public
	  cert URL to ensure that it starts with http.

	  https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

	  Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c

2021-05-24 13:38 +0000 [16e4a9d8cf]  Joshua C. Colp <jcolp@sangoma.com>

	* res_pjsip: On partial transport reload also move factories.

	  For connection oriented transports PJSIP uses factories to
	  produce transports. When doing a partial transport reload
	  we need to also move the factory of the transport over so
	  that anything referencing the transport (such as an endpoint)
	  has the factory available.

	  ASTERISK-29441

	  Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161

2021-05-20 08:18 +0000 [033c2a2283]  Naveen Albert <mail@interlinked.x10host.com>

	* func_volume: Add read capability to function.

	  Up until now, the VOLUME function has been write
	  only, so that TX/RX values can be set but not
	  read afterwards. Now, previously set TX/RX values
	  can be read later.

	  ASTERISK-29439

	  Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f

2021-04-13 02:57 +0000 [59d15c4c2a]  Evgenios_Greek <jone1984@hotmail.com>

	* stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing

	  When unsubscribing from an endpoint technology a FRACK
	  would occur due to incorrect reference counting. This fixes
	  that issue, along with some other issues.

	  Fixed a typo in get_subscription when calling ao2_find as it
	  needed to pass the endpoint ID and not the entire object.

	  Fixed scenario where a subscription would get returned when
	  it shouldn't have been when searching based on endpoint
	  technology.

	  A doulbe unreference has also been resolved by only explicitly
	  releasing the reference held by tech_subscriptions.

	  ASTERISK-28237 #close
	  Reported by: Lucas Tardioli Silveira

	  Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729

2021-05-20 02:15 +0000 [b21d4d1b87]  Joseph Nadiv <ynadiv@corpit.xyz>

	* res_pjsip.c: Support endpoints with domain info in username

	  In multidomain environments, it is desirable to create
	  PJSIP endpoints with the domain info in the endpoint name
	  in pjsip_endpoint.conf.  This resulted in an error with
	  registrations, NOTIFY, and OPTIONS packet generation.

	  This commit will detect if there is an @ in the endpoint
	  identifier and generate the URI accordingly so NOTIFY and
	  OPTIONS From headers will generate correctly.

	  ASTERISK-28393

	  Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619

2021-05-20 07:51 +0000 [3aed363716]  Joshua C. Colp <jcolp@sangoma.com>

	* res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates.

	  RTCP ICE candidates use a base address derived from the RTP
	  candidate. The port on the base address was not being updated to
	  the RTCP port.

	  This change sets the base port to the RTCP port and all is well.

	  ASTERISK-29433

	  Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040

2021-05-25 05:38 +0000 [60ed1847b8]  Joshua C. Colp <jcolp@sangoma.com>

	* asterisk: We've moved to Libera Chat!

	  Change-Id: I48c1933dd79b50ddc0a6793acec4754b4e95c575

2021-05-19 13:13 +0000 [0f8e2174a7]  Jeremy Lainé <jeremy.laine@m4x.org>

	* res_rtp_asterisk: make it possible to remove SOFTWARE attribute

	  By default Asterisk reports the PJSIP version in a SOFTWARE attribute
	  of every STUN packet it sends. This may not be desired in a production
	  environment, and RFC5389 recommends making the use of the SOFTWARE
	  attribute a configurable option:

	  https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2

	  This patch adds a `stun_software_attribute` yes/no option to make it
	  possible to omit the SOFTWARE attribute from STUN packets.

	  ASTERISK-29434

	  Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b

2021-04-15 10:43 +0000 [655ee680cd]  George Joseph <gjoseph@digium.com>

	* res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs

	  RFC7616 and RFC8760 allow more than one WWW-Authenticate or
	  Proxy-Authenticate header per realm, each with different digest
	  algorithms (including new ones like SHA-256 and SHA-512-256).
	  Thankfully however a UAS can NOT send back multiple Authenticate
	  headers for the same realm with the same digest algorithm.  The
	  UAS is also supposed to send the headers in order of preference
	  with the first one being the most preferred.  We're supposed to
	  send an Authorization header for the first one we encounter for a
	  realm that we can support.

	  The UAS can also send multiple realms, especially when it's a
	  proxy that has forked the request in which case the proxy will
	  aggregate all of the Authenticate headers and then send them all
	  back to the UAC.

	  It doesn't stop there though... Each realm can require a
	  different username from the others.  There's also nothing
	  preventing each digest algorithm from having a unique password
	  although I'm not sure if that adds any benefit.

	  So now... For each Authenticate header we encounter, we have to
	  determine if we support the digest algorithm and, if not, just
	  skip the header.  We then have to find an auth object that
	  matches the realm AND the digest algorithm or find a wildcard
	  object that matches the digest algorithm. If we find one, we add
	  it to the results vector and read the next Authenticate header.
	  If the next header is for the same realm AND we already added an
	  auth object for that realm, we skip the header. Otherwise we
	  repeat the process for the next header.

	  In the end, we'll have accumulated a list of credentials we can
	  pass to pjproject that it can use to add Authentication headers
	  to a request.

	  NOTE: Neither we nor pjproject can currently handle digest
	  algorithms other than MD5.  We don't even have a place for it in
	  the ast_sip_auth object. For this reason, we just skip processing
	  any Authenticate header that's not MD5.  When we support the
	  others, we'll move the check into the loop that searches the
	  objects.

	  Changes:

	   * Added a new API ast_sip_retrieve_auths_vector() that takes in
	     a vector of auth ids (usually supplied on a call to
	     ast_sip_create_request_with_auth()) and populates another
	     vector with the actual objects.

	   * Refactored res_pjsip_outbound_authenticator_digest to handle
	     multiple Authenticate headers and set the stage for handling
	     additional digest algorithms.

	   * Added a pjproject patch that allows them to ignore digest
	     algorithms they don't support.  This patch has already been
	     merged upstream.

	   * Updated documentation for auth objects in the XML and
	     in pjsip.conf.sample.

	   * Although res_pjsip_authenticator_digest isn't affected
	     by this change, some debugging and a testsuite AMI event
	     was added to facilitate testing.

	  Discovered during OpenSIPit 2021.

	  ASTERISK-29397

	  Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281

2021-04-14 09:44 +0000 [83c2a16b2e]  Joseph Nadiv <ynadiv@corpit.xyz>

	* res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml

	  RFC 4235 Section 4.1.6 describes XML elements that should be
	  sent to subscribed endpoints to identify the local and remote
	  participants in the dialog.

	  This patch adds this functionality to PJSIP by iterating through the
	  ringing channels causing the NOTIFY, and inserts the channel info
	  into the dialog so that information is properly passed to the endpoint
	  in dialog-info+xml.

	  ASTERISK-24601
	  Patch submitted: Joshua Elson
	  Modified by: Joseph Nadiv and Sean Bright
	  Tested by: Joseph Nadiv

	  Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b

2021-05-13 09:47 +0000 [bfc25e5de2]  Naveen Albert <mail@interlinked.x10host.com>

	* app_voicemail: Configurable voicemail beep

	  Hitherto, VoiceMail() played a non-customizable beep tone to indicate
	  the caller could leave a message. In some cases, the beep may not
	  be desired, or a different tone may be desired.

	  To increase flexibility, a new option allows customization of the tone.
	  If the t option is specified, the default beep will be overridden.
	  Supplying an argument will cause it to use the specified file for the tone,
	  and omitting it will cause it to skip the beep altogether. If the option
	  is not used, the default behavior persists.

	  ASTERISK-29349

	  Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280

2021-05-13 10:32 +0000 [0ad3504ce0]  Naveen Albert <mail@interlinked.x10host.com>

	* AMI: Add AMI event to expose hook flash events

	  Although Asterisk can receive and propogate flash events, it currently
	  provides no mechanism for doing anything with them itself.

	  This AMI event allows flash events to be processed by Asterisk.
	  Additionally, AST_CONTROL_FLASH is included in a switch statement
	  in channel.c to avoid throwing a warning when we shouldn't.

	  ASTERISK-29380

	  Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81

2021-05-13 08:50 +0000 [7b82587dd6]  Naveen Albert <mail@interlinked.x10host.com>

	* chan_sip: Expand hook flash recognition.

	  Some ATAs send hook flash events as application/hook-flash, rather than a DTMF
	  event. Now, we also recognize hook-flash as a flash event.

	  ASTERISK-29370

	  Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725

2021-05-11 12:00 +0000 [6d5cac1d10]  Joshua C. Colp <jcolp@sangoma.com>

	* pjsip: Add patch for resolving STUN packet lifetime issues.

	  In some cases it was possible for a STUN packet to be destroyed
	  prematurely or even destroyed partially multiple times.

	  This patch provided by Teluu fixes the lifetime of these
	  packets and ensures they aren't partially destroyed multiple
	  times.

	  https://github.com/pjsip/pjproject/pull/2709

	  ASTERISK-29377

	  Change-Id: Ie842ad24ddf345e01c69a4d333023f05f787abca

2021-05-13 10:13 +0000 [283fa3a93b]  Naveen Albert <mail@interlinked.x10host.com>

	* main/file.c: Don't throw error on flash event.

	  AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
	  where it should be ignored. Adding this to the switch ensures a
	  warning isn't thrown on RFC2833 flash events, since nothing's amiss.

	  ASTERISK-29372

	  Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc

2021-05-12 21:20 +0000 [78d7862463]  Sean Bright <sean.bright@gmail.com>

	* chan_pjsip: Correct misleading trace message

	  ASTERISK-29358 #close

	  Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647

2021-04-26 17:00 +0000 [a84d34035a]  Ben Ford <bford@digium.com>

	* STIR/SHAKEN: Switch to base64 URL encoding.

	  STIR/SHAKEN encodes using base64 URL format. Currently, we just use
	  base64. New functions have been added that convert to and from base64
	  encoding.

	  The origid field should also be an UUID. This means there's no reason to
	  have it as an option in stir_shaken.conf, as we can simply generate one
	  when creating the Identity header.

	  https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

	  Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c

2021-05-11 12:26 +0000 [e0cbdfe063]  Ben Ford <bford@digium.com>

	* STIR/SHAKEN: OPENSSL_free serial hex from openssl.

	  We're getting the serial number of the certificate from openssl and
	  freeing it with ast_free(), but it needs to be freed with OPENSSL_free()
	  instead. Now we duplicate the string and free the one from openssl with
	  OPENSSL_free(), which means we can still use ast_free() on the returned
	  string.

	  https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

	  Change-Id: Ia6e1a4028c1933a0e1d204b769ebb9f5a11f00ab

2021-04-21 11:12 +0000 [5e6508b56f]  Ben Ford <bford@digium.com>

	* STIR/SHAKEN: Fix certificate type and storage.

	  During OpenSIPit, we found out that the public certificates must be of
	  type X.509. When reading in public keys, we use the corresponding X.509
	  functions now.

	  We also discovered that we needed a better naming scheme for the
	  certificates since certificates with the same name would cause issues
	  (overwriting certs, etc.). Now when we download a public certificate, we
	  get the serial number from it and use that as the name of the cached
	  certificate.

	  The configuration option public_key_url in stir_shaken.conf has also
	  been renamed to public_cert_url, which better describes what the option
	  is for.

	  https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

	  Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d

2021-04-22 13:07 +0000 [40bdfff73b]  George Joseph <gjoseph@digium.com>

	* Updates for the MessageSend Dialplan App

	  Enhancements:

	   * The MessageSend dialplan application now takes an optional
	     third argument that can set the message's "To" field on
	     outgoing messages.  It's an alternative to using the
	     MESSAGE(to) dialplan function.

	     NOTE: No channel driver currently implements this field.  A
	     follow-on commit for res_pjsip_messaging will implement it for
	     the chan_pjsip channel driver.

	   * To prevent confusion with the first argument, currently named
	     "to", it's been renamed to "destination". Its function,
	     creating the request URI, hasn't changed.

	   * The documentation for MessageSend was updated to be
	     more clear about the parameters and how they interact
	     the MESSAGE() dialplan function.

	   * With the rename of MessageSend's first parameter, and the fact
	     that message.c references <info> elements in chan_sip.c,
	     res_pjsip_messaging.c and res_xmpp, they each needed
	     documentation updates to use MessageDestinationInfo instead of
	     MessageToInfo.

	   * appdocsxml.dtd was updated to include a missing element
	     declaration for "dataType".  This was showing up as an error
	     in Eclipse's dtd editor.

	   * Despite the changes in this commit, there should be
	     no impact to current users of MessageSend.

	  Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a

2021-04-30 15:21 +0000 [78f518622d]  Sean Bright <sean.bright@gmail.com>

	* translate.c: Avoid refleak when checking for a translation path

	  Change-Id: Idbd61ff77545f4a78b06a5064b55112e774b70e6

2021-04-28 07:17 +0000 [8faed04b01]  Joshua C. Colp <jcolp@sangoma.com>

	* chan_local: Skip filtering audio formats on removed streams.

	  When a stream topology is provided to chan_local when dialing
	  it filters the audio formats down. This operation did not skip
	  streams which were removed (that have no formats) resulting in
	  calling being aborted.

	  This change causes such streams to be skipped.

	  ASTERISK-29407

	  Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6

2021-04-27 12:31 +0000 [95414fc918]  Sean Bright <sean.bright@gmail.com>

	* res_rtp_asterisk: More robust timestamp checking

	  We assume that a timestamp value of 0 represents an 'uninitialized'
	  timestamp, but 0 is a valid value. Add a simple wrapper to be able to
	  differentiate between whether the value is set or not.

	  This also removes the fix for ASTERISK~28812 which should not be
	  needed if we are checking the last timestamp appropriately.

	  ASTERISK-29030 #close

	  Change-Id: Ie70d657d580d9a1f2877e25a6ef161c5ad761cf7

2021-04-29 15:32 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.4.0-rc1 Released.

2021-04-29 10:25 +0000 [1949d828b7]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.4.0
2021-04-23 12:37 +0000 [d2dcd15bd8]  Sean Bright <sean.bright@gmail.com>

	* res_pjsip.c: OPTIONS processing can now optionally skip authentication

	  ASTERISK-27477 #close

	  Change-Id: I68f6715bba92a525149e35d142a49377a34a1193

2021-04-21 06:42 +0000 [dec44306cf]  Jean Aunis <jean.aunis@prescom.fr>

	* translate.c: Take sampling rate into account when checking codec's buffer size

	  Up/down sampling changes the number of samples produced by a translation.
	  This must be taken into account when checking the codec's buffer size.

	  ASTERISK-29328

	  Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055

2021-04-25 04:45 +0000 [c2f4925ee0]  Joshua C. Colp <jcolp@sangoma.com>

	* svn: Switch to https scheme.

	  Some versions of SVN seemingly don't follow the redirect
	  to https.

	  Change-Id: Ia7c76c18cb620bcf56f08e1211a7d80d321fe253

2021-04-20 08:42 +0000 [5f3d96a765]  George Joseph <gjoseph@digium.com>

	* res_pjsip:  Update documentation for the auth object

	  Change-Id: I2f76867ce02ec611964925159be099de83346e38

2021-04-02 07:21 +0000 [88aec107df]  George Joseph <gjoseph@digium.com>

	* bridge_channel_write_frame: Check for NULL channel

	  There is a possibility, when bridge_channel_write_frame() is
	  called, that the bridge_channel->chan will be NULL.  The first
	  thing bridge_channel_write_frame() does though is call
	  ast_channel_is_multistream() which had no check for a NULL
	  channel and therefore caused a segfault. Since it's still
	  possible for bridge_channel_write_frame() to write the frame to
	  the other channels in the bridge, we don't want to bail before we
	  call ast_channel_is_multistream() but we can just skip the
	  multi-channel stuff.  So...

	  bridge_channel_write_frame() only calls ast_channel_is_multistream()
	  if bridge_channel->chan is not NULL.

	  As a safety measure, ast_channel_is_multistream() now returns
	  false if the supplied channel is NULL.

	  ASTERISK-29379
	  Reported-by: Vyrva Igor
	  Reported-by: Ross Beer

	  Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce

2021-04-01 10:38 +0000 [404533c149]  Sean Bright <sean.bright@gmail.com>

	* loader.c: Speed up deprecation metadata lookup

	  Only use an XPath query once per module, then just navigate the DOM for
	  everything else.

	  Change-Id: Ia0336a7185f9180ccba4b6f631a00f9a22a36e92

2021-04-01 08:39 +0000 [19eef2a6dc]  George Joseph <gjoseph@digium.com>

	* res_prometheus: Clone containers before iterating

	  The channels, bridges and endpoints scrape functions were
	  grabbing their respective global containers, getting the
	  count of entries, allocating metric arrays based on
	  that count, then iterating over the container.  If the
	  global container had new objects added after the count
	  was taken and the metric arrays were allocated, we'd run
	  out of metric entries and attempt to write past the end
	  of the arrays.

	  Now each of the scape functions clone their respective
	  global containers and all operations are done on the
	  clone.  Since the clone is stable between getting the
	  count and iterating over it, we can't run past the end
	  of the metrics array.

	  ASTERISK-29130
	  Reported-By: Francisco Correia
	  Reported-By: BJ Weschke
	  Reported-By: Sébastien Duthil

	  Change-Id: If0c8e40853bc0e9429f2ba9c7f5f358d90c311af

2021-03-10 09:03 +0000 [a9a9864478]  Joshua C. Colp <jcolp@sangoma.com>

	* loader: Output warnings for deprecated modules.

	  Using the information from the MODULEINFO XML we can
	  now output useful information at the end of module
	  loading for deprecated modules. This includes the
	  version it was deprecated in, the version it will be
	  removed in, and the replacement if available.

	  ASTERISK-29339

	  Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a

2021-03-22 15:22 +0000 [17c86dcfaa]  Kevin Harwell <kharwell@sangoma.com>

	* res_rtp_asterisk: Fix standard deviation calculation

	  For some input to the standard deviation algorithm extremely large,
	  and wrong numbers were being calculated.

	  This patch uses a new formula for correctly calculating both the
	  running mean and standard deviation for the given inputs.

	  ASTERISK-29364 #close

	  Change-Id: Ibc6e18be41c28bed3fde06d612607acc3fbd621f

2021-03-29 17:40 +0000 [0ad1ff8a72]  Kevin Harwell <kharwell@sangoma.com>

	* res_rtp_asterisk: Don't count 0 as a minimum lost packets

	  The calculated minimum lost packets represents the lowest number of
	  lost packets missed during an RTCP report interval. Zero of course
	  is the lowest, but the idea is that this value contain the lowest
	  number of lost packets once some have been missed.

	  This patch checks to make sure the number of lost packets over an
	  interval is not zero before checking and setting the minimum value.

	  Also, this patch updates the rtp lost packet test to check for
	  packet loss over several reports vs one.

	  Change-Id: I07d6e21cec61e289c2326138d6bcbcb3c3d5e008

2021-03-31 12:17 +0000 [1414b9cc57]  Kevin Harwell <kharwell@sangoma.com>

	* res_rtp_asterisk: Statically declare rtp_drop_packets_data object

	  This patch makes the drop_packets_data object static.

	  Change-Id: If4f9b21fa0c47d41a35b6b05941d978efb4da87b

2021-03-29 17:52 +0000 [b0d828f14a]  Joshua C. Colp <jcolp@sangoma.com>

	* res_rtp_asterisk: Only raise flash control frame on end.

	  Flash in RTP is conveyed the same as DTMF, just with a
	  specific digit. In Asterisk however we do flash as a
	  single control frame.

	  This change makes it so that only on end do we provide
	  the flash control frame to the core. Previously we would
	  provide a flash control frame on both begin and end,
	  causing flash to work improperly.

	  ASTERISK-29373

	  Change-Id: I1accd9c6e859811336e670e698bd8bd124f33226

2021-03-05 12:53 +0000 [b912b31853]  Kevin Harwell <kharwell@sangoma.com>

	* res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command

	  This patch makes it so when Asterisk is compiled in DEVMODE a CLI
	  command is available that allows someone to drop incoming RTP
	  packets. The command allows for dropping of packets once, or on a
	  timed interval (e.g. drop 10 packets every 5 seconds). A user can
	  also specify to drop packets by IP address.

	  Change-Id: I25fa7ae9bad6ed68e273bbcccf0ee51cae6e7024

2021-03-30 06:59 +0000 [65a4a3a4e6]  Joshua C. Colp <jcolp@sangoma.com>

	* res_pjsip: Give error when TLS transport configured but not supported.

	  Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b

2021-03-05 12:47 +0000 [15de2f1727]  Kevin Harwell <kharwell@sangoma.com>

	* time: Add timeval create and unit conversion functions

	  Added a TIME_UNIT enumeration, and a function that converts a
	  string to one of the enumerated values. Also, added functions
	  that create and initialize a timeval object using a specified
	  value, and unit type.

	  Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392

2021-03-24 08:38 +0000 [35302efe73]  Sean Bright <sean.bright@gmail.com>

	* app_queue: Add alembic migration to add ringinuse to queue_members.

	  ASTERISK-28356 #close

	  Change-Id: I53a1bfdd3113d620bea88349019173a2f3f0ae39

2021-03-28 10:47 +0000 [be3153346b]  Sean Bright <sean.bright@gmail.com>

	* modules.conf: Fix more differing usages of assignment operators.

	  I missed the changes in 18 and master in the previous review.

	  ASTERISK-24434 #close

	  Change-Id: Ieb132b2a998ce96daa9c9acf26535a974b895876

2021-03-24 10:52 +0000 [bbfb8f2b9d]  Ben Ford <bford@digium.com>

	* logger.conf.sample: Add more debug documentation.

	  Change-Id: Iff0e713f2120d8dce8e1e26924b99ed17f9d9dff

2021-03-23 17:24 +0000 [31364fa4c8]  Sean Bright <sean.bright@gmail.com>

	* queues.conf.sample: Correct 'context' documentation.

	  ASTERISK-24631 #close

	  Change-Id: I8bf8776906a72ee02f24de6a85345940b9ff6b6f

2021-03-23 15:15 +0000 [e27fa9eceb]  Sean Bright <sean.bright@gmail.com>

	* app_queue.c: Remove dead 'updatecdr' code.

	  Also removed the sample documentation, and some oddly-placed
	  documentation about the timeout argument to the Queue() application
	  itself. There is a large section on the timeout behavior below.

	  ASTERISK-26614 #close

	  Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217

2021-03-19 09:11 +0000 [a0009c807e]  Mark Murawski <markm@intellasoft.net>

	* logger: Console sessions will now respect logger.conf dateformat= option

	  The 'core' console (ie: asterisk -c) does read logger.conf and does
	  use the dateformat= option.

	  Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
	  and uses a hard coded dateformat option for printing received verbose messages:
	    main/logger.c: static char dateformat[256] = "%b %e %T"

	  This change will load logger.conf for each remote console session and
	  use the dateformat= option to set the per-line timestamp for verbose messages

	  Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
	  ASTERISK-25358: #close
	  Reported-by: Igor Liferenko

2021-03-19 15:57 +0000 [4393207751]  Sean Bright <sean.bright@gmail.com>

	* app_queue.c: Don't crash when realtime queue name is empty.

	  ASTERISK-27542 #close

	  Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a

2021-03-18 11:14 +0000 [c78d0ce429]  George Joseph <gjoseph@digium.com>

	* res_pjsip_session: Make reschedule_reinvite check for NULL topologies

	  When the check for equal topologies was added to reschedule_reinvite()
	  it was assumed that both the pending and active media states would
	  actually have non-NULL topologies.  We since discovered this isn't
	  the case.

	  We now only test for equal topologies if both media states have
	  non-NULL topologies.  The logic had to be rearranged a bit to make
	  sure that we cloned the media states if their topologies were
	  non-NULL but weren't equal.

	  ASTERISK-29215

	  Change-Id: I61313cca7fc571144338aac826091791b87b6e17

2021-03-19 04:56 +0000 [55c467eab1]  Joshua C. Colp <jcolp@sangoma.com>

	* app_queue: Only send QueueMemberStatus if status changes.

	  If a queue member was updated with the same status multiple
	  times each time a QueueMemberStatus event would be sent
	  which would be a duplicate of the previous.

	  This change makes it so that the QueueMemberStatus event is