Log Message:
asterisk16: Update to 16.10.0

Changelog:
16.10.0:
New Features made in this release:

-----------------------------------
[ASTERISK-6863] -
		[patch] allow Asterisk to set high ToS bits as non-root on Linux
(Reported by Matt Addison)

Bugs fixed in this release:

-----------------------------------
[ASTERISK-28852] -
		Unprotected access to nochecksums variable, causes build failures
(Reported by Guido Falsi)
[ASTERISK-28846] -
		stream: Enforce formats immutability
(Reported by Joshua C. Colp)
[ASTERISK-28847] -
		ARI channels cuts the endpoint string over 80 characters
(Reported by sungtae kim)
[ASTERISK-28811] -
		Crash occurs when fax session switches from T.38 to audio
(Reported by Alexey Vasilyev)
[ASTERISK-28839] -
		Sporadic crashes with Segmentation fault
(Reported by Joeran Vinzens)
[ASTERISK-28835] -
		IPv6 addresses in SDP incorrectly formatted
(Reported by Daniel Heckl)
[ASTERISK-28372] -
		Asterisk REPLY Wrong Contact header port (TCP)
(Reported by Anton Satskiy)
[ASTERISK-24428] -
		Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for \ 
TLS) if the extern option variants aren't used
(Reported by sstream)
[ASTERISK-28838] -
		AST_MODULE_INFO requires, MODULEINFO does not mention
(Reported by Alexander Traud)
[ASTERISK-28841] -
		app_confbridge: Add support for disabling text messaging for a user
(Reported by Joshua C. Colp)
[ASTERISK-28837] -
		pjproject_bundled: Honor --without-pjproject.
(Reported by Alexander Traud)
[ASTERISK-28827] -
		res_rtp_asterisk: Loop when receive buffer is flushed by a received packet \ 
that is also in receive buffer with NACK
(Reported by nappsoft)
[ASTERISK-27195] -
		chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets
(Reported by Joshua Roys)
[ASTERISK-28826] -
		res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK
(Reported by nappsoft)
[ASTERISK-28812] -
		First DTMF is not get
(Reported by Bernard Merindol)
[ASTERISK-28758] -
		pjsip startup errors when using "with-ssl" configure option
(Reported by Patrick Wakano)
[ASTERISK-28824] -
		BuildSystem: Search for Python/C API when possibly needed only.
(Reported by Alexander Traud)
[ASTERISK-27717] -
		[patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7.
(Reported by Alexander Traud)
[ASTERISK-28798] -
		[patch] chan_sip: TCP/TLS client without server.
(Reported by Alexander Traud)
[ASTERISK-28817] -
		chan_pjsip: constant DTMF tone if RTP is not setup yet
(Reported by Kevin Harwell)
[ASTERISK-28819] -
		[patch] bridge_softmix_binaural: Show state in menuselect.
(Reported by Alexander Traud)
[ASTERISK-28816] -
		[patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
(Reported by Alexander Traud)
[ASTERISK-28818] -
		[patch] BuildSystem: Allow space in path.
(Reported by Alexander Traud)
[ASTERISK-28796] -
		func_channel: cannot read fields exten, context, userfield, channame from dialplan
(Reported by Sébastien Duthil)
[ASTERISK-28809] -
		[patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
(Reported by Alexander Traud)
[ASTERISK-28803] -
		[patch] chan_unistim: Avoid tautological warnings with clang.
(Reported by Alexander Traud)
[ASTERISK-28808] -
		[patch] test_stasis: Avoid always true warning with clang.
(Reported by Alexander Traud)
[ASTERISK-28056] -
		res_pjsip: Incorrect endpoint status after endpoint synchronization for a \ 
specific AOR
(Reported by Jason Hord)
[ASTERISK-28795] -
		channel: write to a stream on multi-frame writes
(Reported by Kevin Harwell)
[ASTERISK-28789] -
		test_utils: incorrectly printing error 'declined to load'
(Reported by Alexander Traud)
[ASTERISK-28788] -
		func_aes: incorrectly printing error 'declined to load'
(Reported by Alexander Traud)
[ASTERISK-28790] -
		Crash during conference call using confbridge and video
(Reported by Pascal Cadotte Michaud)
[ASTERISK-16676] -
		DAHDIRAS fails to properly initiate pppd unless asterisk is running as root
(Reported by Jaco Kroon)
[ASTERISK-21205] -
		[patch] dundi_read_result crash due to negative number
(Reported by Jaco Kroon)
[ASTERISK-28784] -
		res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
(Reported by Joshua C. Colp)
[ASTERISK-28743] -
		Asterisk is crashing if the 200 OK with SDP
(Reported by sungtae kim)
[ASTERISK-28783] -
		res_pjsip_session: Allow default non-audio streams to have reflected state
(Reported by Joshua C. Colp)
[ASTERISK-28774] -
		chan_pjsip's rtptimeout is erroneously triggered during direct-media \ 
(native_rtp) bridge
(Reported by Michael Neuhauser)
[ASTERISK-20325] -
		Comments in configs/func_odbc.conf.sample are not consistent with examples. \ 
Missing examples.
(Reported by Olivier Krief)
[ASTERISK-28780] -
		app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup
(Reported by Joshua C. Colp)
[ASTERISK-28773] -
		Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
(Reported by Torrey Searle)
[ASTERISK-28769] -
		DTLS Handshake Fails to Occur if ice_support is enabled but not used
(Reported by Torrey Searle)
[ASTERISK-28759] -
		A non negotiated rtp frame causes call disconnection when there is a SSRC change
(Reported by Paulo Vicentini)
[ASTERISK-26711] -
		func_enum: ENUM code wrong case
(Reported by Vitold)
[ASTERISK-23407] -
		Fix the FSF address in the headers of lots of pjproject files
(Reported by Jared Smith)
[ASTERISK-19460] -
		[patch] Function TXTCIDNAME never actually makes DNS calls and always returns \ 
an empty string
(Reported by George Joseph)

Improvements made in this release:

-----------------------------------
[ASTERISK-28853] -
		Missing include on FreeBSD
(Reported by Guido Falsi)
[ASTERISK-28813] -
		func_volume: Allow decimal numbers as parameter to improve granularity
(Reported by Jean Aunis - Prescom)
[ASTERISK-27946] -
		dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't
(Reported by Joshua Elson)
[ASTERISK-28782] -
		Add support for Content-Disposition header in multi-part INVITES
(Reported by Torrey Searle)
[ASTERISK-28787] -
		res_pjsip_session: Decide more intelligently when to add video
(Reported by Joshua C. Colp)

16.9.0:
Bugs fixed in this release:
-----------------------------------

    [ASTERISK-28766] -

	 	PJSIP blind transfer not completed after using Proceeding()
(Reported by lvl)

    [ASTERISK-28685] -

	 	check_expr2: linking (when hardening) and cross-compiling troubles
(Reported by Sebastian Kemper)

    [ASTERISK-28764] -

	 	res_rtp_asterisk: Improve NACK support and seqno handling
(Reported by Joshua C. Colp)

    [ASTERISK-28755] -

	 	SIP/Stasis: SIP headers not transmitted in the "variables" field
(Reported by Jean Aunis - Prescom)

    [ASTERISK-28754] -

	 	ASTERISK-28738 Causes Audio Issue After Hold
(Reported by Ross Beer)

    [ASTERISK-28697] -

	 	res_pjsip: Named ACL does not update on reload if changed
(Reported by Timothy Vanderaerden)

    [ASTERISK-28746] -

	 	res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set
(Reported by George Joseph)

    [ASTERISK-28716] -

	 	ICE: pjnath shouldn't wait for ICE to complete before allowing sending
(Reported by Benjamin Keith Ford)

    [ASTERISK-28738] -

	 	Incorrect state machine used when MOH_PASSTHRU is used
(Reported by Torrey Searle)

    [ASTERISK-28742] -

	 	res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
(Reported by Kevin Harwell)

    [ASTERISK-28735] -

	 	Realtime MoH Unknown format '' -- defaulting to SLIN
(Reported by Ross Beer)

    [ASTERISK-28730] -

	 	res_pjsip_session: Fix out of order session refreshes
(Reported by Joshua C. Colp)

    [ASTERISK-28718] -

	 	chan_sip: Returns 403 if RTP ports are depleted, should return 503
(Reported by Walter Doekes)

    [ASTERISK-28719] -

	 	Cannot remove defaultrule from queue using realtime queues
(Reported by EDV O-TON)

    [ASTERISK-28713] -

	 	res_stasis_playback: Error building JSON
(Reported by Sébastien Duthil)

    [ASTERISK-28714] -

	 	REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes \ 
Segfaults
(Reported by Ross Beer)

    [ASTERISK-26082] -

	 	res_pjsip_messaging: MessageSend Content-Type can't be changed
(Reported by Alex)

    [ASTERISK-28423] -

	 	ARI causes STASIS Deadlock
(Reported by Ross Beer)

    [ASTERISK-28679] -

	 	stasis application is destroyed after its creation
(Reported by Francois Blackburn)

    [ASTERISK-25421] -

	 	PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending
(Reported by Dmitriy Serov)

    [ASTERISK-28686] -

	 	chan_sip strictrtp=yes fails when media source is changed: no audio
(Reported by Walter Doekes)

    [ASTERISK-28139] -

	 	RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
(Reported by Paul Brooks)

    [ASTERISK-26955] -

	 	pjsip: SIP Packets with Via "received=" Containing IPv6 Address \ 
Delimited by "[]" Rejected
(Reported by Peter Sokolov)

Improvements made in this release:
-----------------------------------

    [ASTERISK-28750] -

	 	TLS/SSL Key too small error
(Reported by Martin Zeh)

    [ASTERISK-28733] -

	 	stream: Add support for adding/removing streams during SFU/calls
(Reported by Joshua C. Colp)

    [ASTERISK-24798] -

	 	Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor
(Reported by xrobau)

    [ASTERISK-28726] -

	 	install_prereq script uses the interactive mode when installing aptitude
(Reported by Sylvain Afchain)

16.8.0:
 New Features made in this release:

-----------------------------------
[ASTERISK-17491] -
		CURLOPT() needs a "followlocation" parameter / "maxredirs" \ 
doesn't do anything
(Reported by candrews)
[ASTERISK-28639] -
		res_pjsip_endpoint_identifier_ip: Add ability to match on source port
(Reported by Sean Bright)

Bugs fixed in this release:

-----------------------------------
[ASTERISK-28679] -
		stasis application is destroyed after its creation
(Reported by Francois Blackburn)
[ASTERISK-28423] -
		ARI causes STASIS Deadlock
(Reported by Ross Beer)
[ASTERISK-28714] -
		REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes \ 
Segfaults
(Reported by Ross Beer)
[ASTERISK-28677] -
		CDR billsec is always 0 for transferred calls
(Reported by Maciej Michno)
[ASTERISK-28702] -
		chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
(Reported by Andrew Siplas)
[ASTERISK-28706] -
		silk 24hHz doesn't show up in 'core show translation' output
(Reported by Sean Bright)
[ASTERISK-24484] -
		Update documentation for statsd module - usage requirements unclear
(Reported by Dan Jenkins)
[ASTERISK-28695] -
		core: minmemfree watermark uses free RAM, not available RAM
(Reported by Kevin Flyn)
[ASTERISK-28693] -
		chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan
(Reported by Frank Matano)
[ASTERISK-23739] -
		[patch]Segfault forwarding voicemail with ODBC storage enabled and realtime \ 
voicemail_data is used
(Reported by Stas Kobzar)
[ASTERISK-27622] -
		empty voicemail.conf required for ARA (realtime) voicemail to leave message
(Reported by Jim Van Meggelen)
[ASTERISK-28349] -
		Pause reason not reported in QueueMember AMI event
(Reported by Niksa Baldun)
[ASTERISK-21794] -
		CLI command 'realtime update2' syntax failure when using according to usage help
(Reported by Cedric BASSAGET)
[ASTERISK-25429] -
		res_pjsip_endpoint_identifier_ip: Document support for hostnames
(Reported by Joshua C. Colp)
[ASTERISK-27775] -
		res_pjsip_notify: Multiple Event headers can be present instead of just one
(Reported by AvayaXAsterisk)
[ASTERISK-28682] -
		app_record: Lack of `beep` audio file causes application to return error and hangup
(Reported by Corey Farrell)
[ASTERISK-28507] -
		Wiki docs missing for MessageWaiting
(Reported by David M. Lee)
[ASTERISK-27759] -
		res_pjsip_pubsub: Subscription persistence does not preserve XML version number
(Reported by Bryan Nelson)
[ASTERISK-28605] -
		chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X
(Reported by Dirk Wendland)
[ASTERISK-28633] -
		stasis bridge topic leak
(Reported by Joeran Vinzens)
[ASTERISK-28492] -
		pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
(Reported by Jean-Denis Girard)
[ASTERISK-28562] -
		SIP WSS message not processed until next frame arrives
(Reported by Robert Sutton)
[ASTERISK-27243] -
		contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax
(Reported by Richard Kenner)
[ASTERISK-28497] -
		func_odbc: truncating Unicode string on readsql
(Reported by Boris P. Korzun)
[ASTERISK-28647] -
		chan_sip: RTP frames not transmitted after emitting a COLP
(Reported by Jean Aunis - Prescom)
[ASTERISK-28667] -
		Asterisk ignores parsing of config files if a Byte order mark is present
(Reported by Robin Leffmann)
[ASTERISK-28664] -
		"trustrpid" is misspelled in sip_to_pjsip.py
(Reported by Pascal Cadotte Michaud)
[ASTERISK-28604] -
		app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
(Reported by George Joseph)
[ASTERISK-28659] -
		res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create \ 
additional streams and offer does not have them
(Reported by nappsoft)
[ASTERISK-28660] -
		res_fax: wrap Asterisk initiated negotiation with config option
(Reported by Kevin Harwell)
[ASTERISK-28636] -
		app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
(Reported by Frederic LE FOLL)
[ASTERISK-28626] -
		Missing arguments in PJSIP_CONTACT function documentation
(Reported by Pascal Cadotte Michaud)
[ASTERISK-28609] -
		Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)
[ASTERISK-28651] -
		chan_sip logs errors on tx to non-existent TCP connections
(Reported by Jaco Kroon)
[ASTERISK-28502] -
		chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
(Reported by Ross Beer)
[ASTERISK-28625] -
		Playback of local files impacted by large media cache
(Reported by Kevin Reeves)

Improvements made in this release:

-----------------------------------
[ASTERISK-28710] -
		Should be able to disable the /httpstatus URI in the built-in HTTP server
(Reported by Sean Bright)
[ASTERISK-28638] -
		Simplify dialplan for Dial, Page, and ChanIsAvail
(Reported by cmaj)
[ASTERISK-28673] -
		GET FULL VARIABLE documentation clarification
(Reported by Jonathan Harris)
[ASTERISK-28658] -
		app_confbridge: Add support for setting maximum sample rate
(Reported by Joshua C. Colp)