[asterisk-dev] asterisk-dev Digest, Vol 193, Issue 26
zhengbin83
zhengbin83 at 163.com
Sun Jan 10 21:16:00 CST 2021
> 2020年10月28日 上午5:12，asterisk-dev-request at lists.digium.com 写道：
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> Today's Topics:
>
> 1. Re: Problem with SDP session id in 200 OK during ReInvite
> (Michael Maier)
> 2. Re: Asterisk 18.0.0 Now Available (John Kiniston)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 27 Oct 2020 18:47:36 +0100
> From: Michael Maier <m1278468 at mailbox.org>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Problem with SDP session id in 200 OK
> during ReInvite
> Message-ID: <6e2ceaf7-b482-0880-9e2c-648c408f02dc at mailbox.org>
> Content-Type: text/plain; charset=utf-8
>
> Hello Joshua,
>
>
> On 27.10.20 at 10:07 Joshua C. Colp wrote:
>> On Mon, Oct 26, 2020 at 2:02 PM Michael Maier <m1278468 at mailbox.org> wrote:
>>
>>> Hello!
>>>
>>> I'm facing the problem, that *sometimes* the SDP session ID isn't
>>> incremented in the 200 OK, which asterisk sends as answer to a ReInvite it
>>> got from the peer (use case: session
>>> timer handling). This leads to broken calls, because the SDP session ID
>>> must be incremented if the session description has changed (the session
>>> description has changed).
>>>
>>> Modifying the SDP session ID is possible in
>>> res/res_pjsip/pjsip_message_filter.c / filter_on_tx_message() for SDPs
>>> contained in an Invite, which is created and sent by asterisk.
>>> At the moment, I'm already modifying the SDP session ID at this place,
>>> because of another problem:
>>>
>>
>> <snip>
>>
>>
>>> Maybe it's too late and the processing of the 200 OK doesn't hit this part
>>> at all?
>>>
>>> Or is there any other possibility to modify the SDP session ID contained
>>> in the 200 OK, that is sent by asterisk as an answer to a ReInvite?
>>>
>>
>> It should be modifiable there, it injects itself at the transaction layer
>> to be called for both requests and responses. I'm not sure under what
>> scenarios (if any) it would not be called.
>
> thanks for your estimation! Meanwhile, I hope I have found the problem now. I changed my workaround to always statically modify the SDP session id (the condition seemed to be the
> problem). But tests are going on.
>
>
> Thanks
> Michael
>
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 27 Oct 2020 14:13:53 -0700
> From: John Kiniston <johnkiniston at gmail.com>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: Re: [asterisk-dev] Asterisk 18.0.0 Now Available
> Message-ID:
> <CAFJQOGfXSK=hMA31AWaB24fZv34Gor4mntPS==V-4MXMLZ-TsA at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Is anyone else seeing menuconfig give the wrong description app_audiosocket
> and chan_audiosocket selections with this release?
>
> I've tried on two systems and I'm seeing the same thing, If I highlight
> app_audioosocket I get a description of AST_MODULE_INFO(
> and chan_audiosocket has a description of AST_MODULE_INFO(ASTERISK_GPL_KEY,
> AST_MODFLAG_LOAD_ORDER,
>
>
>
> It's not affecting me, Just a weird display thing.
>
> On Tue, Oct 20, 2020 at 5:02 AM Asterisk Development Team <
> asteriskteam at digium.com> wrote:
>
>> The Asterisk Development Team would like to announce the release of
>> Asterisk 18.0.0.
>> This release is available for immediate download at
>> https://downloads.asterisk.org/pub/telephony/asterisk
>>
>> The release of Asterisk 18.0.0 resolves several issues reported by the
>> community and would have not been possible without your participation.
>>
>> *Thank you!*
>>
>> The following issues are resolved in this release:
>>
>> *Security bugs fixed in this release:*
>> -----------------------------------
>>
>> - [ASTERISK-28589
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28589>] -
>>
>> chan_sip: Depending on configuration an INVITE can alter Addr of a peer
>> (Reported by Andrey V. T.)
>>
>> - [ASTERISK-28580
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28580>] -
>>
>> Bypass SYSTEM write permission in manager action allows system commands
>> execution
>> (Reported by Eliel Sardañons)
>>
>> - [ASTERISK-28495
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28495>] -
>>
>> res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
>> (Reported by Alexei Gradinari)
>>
>> *New Features made in this release:*
>> -----------------------------------
>>
>> - [ASTERISK-6863
>> <https://issues.asterisk.org/jira/browse/ASTERISK-6863>] -
>>
>> [patch] allow Asterisk to set high ToS bits as non-root on Linux
>> (Reported by Matt Addison)
>>
>> - [ASTERISK-17491
>> <https://issues.asterisk.org/jira/browse/ASTERISK-17491>] -
>>
>> CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do
>> anything
>> (Reported by candrews)
>>
>> - [ASTERISK-28639
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28639>] -
>>
>> res_pjsip_endpoint_identifier_ip: Add ability to match on source port
>> (Reported by Sean Bright)
>>
>> - [ASTERISK-28614
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28614>] -
>>
>> app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only
>> "sending"
>> (Reported by lvl)
>>
>> - [ASTERISK-28613
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28613>] -
>>
>> func_curl: CURLOPT cannot set Content-Type header
>> (Reported by Martin Tomec)
>>
>> - [ASTERISK-28533
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28533>] -
>>
>> func_jitterbuffer: Add support for video synchronization
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-17808
>> <https://issues.asterisk.org/jira/browse/ASTERISK-17808>] -
>>
>> [patch] Unregister a realtime moh class
>> (Reported by Byron Clark)
>>
>> - [ASTERISK-28489
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28489>] -
>>
>> Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI
>> domain
>> (Reported by Stas Kobzar)
>>
>> *Bugs fixed in this release:*
>> -----------------------------------
>>
>> - [ASTERISK-29109
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29109>] -
>>
>> res_pjsip_session: Asterisk 18 does not progress calls due to codec
>> negotiation after upgrading from Asterisk 16
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-25665
>> <https://issues.asterisk.org/jira/browse/ASTERISK-25665>] -
>>
>> Duplicate logging in queue log for EXITEMPTY events
>> (Reported by Ove Aursand)
>>
>> - [ASTERISK-29043
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29043>] -
>>
>> app_queue: Leave empty sometimes not recorded as abandoned
>> (Reported by Kfir Itzhak)
>>
>> - [ASTERISK-29042
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29042>] -
>>
>> res_parking: Parker UUID is no longer copied
>> (Reported by Misha Vodsedalek)
>>
>> - [ASTERISK-28878
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28878>] -
>>
>> chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
>> (Reported by Joseph Ades)
>>
>> - [ASTERISK-29046
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29046>] -
>>
>> pbx: Deadlock when doing a reload, while simultaneously doing an
>> ExtensionState on a pattern match hint that ends up adding an extension
>> (Reported by Ramarajan)
>>
>> - [ASTERISK-29040
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29040>] -
>>
>> res_speech: Assertion on format
>> (Reported by Nickolay V. Shmyrev)
>>
>> - [ASTERISK-29001
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29001>] -
>>
>> chan_pjsip does not process or forward 181 responses
>> (Reported by Torrey Searle)
>>
>> - [ASTERISK-29034
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29034>] -
>>
>> Lastpause of realtime members is reseting
>> (Reported by Evandro César Arruda)
>>
>> - [ASTERISK-27273
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27273>] -
>>
>> app_voicemail: When a voicemail is marked as "Urgent", it is not sent by
>> email/processed by the mailcmd command
>> (Reported by Leandro Dardini)
>>
>> - [ASTERISK-29033
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29033>] -
>>
>> res_pjsip_session: Aggressively terminates session on failed re-INVITE
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28974
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28974>] -
>>
>> res_rtp_asterisk: T.140 messages have appended RTP string to each message
>> block.
>> (Reported by Thomas Johnson)
>>
>> - [ASTERISK-29011
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29011>] -
>>
>> chan_sip: ToHost property not cleared on reload
>> (Reported by Dennis)
>>
>> - [ASTERISK-29021
>> <https://issues.asterisk.org/jira/browse/ASTERISK-29021>] -
>>
>> [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions
>> (Reported by cmaj)
>>
>> - [ASTERISK-28927
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28927>] -
>>
>> Asterisk crash in music on hold
>> (Reported by David Cunningham)
>>
>> - [ASTERISK-28973
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28973>] -
>>
>> Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is
>> active (UDP transport with external_media_address)
>> (Reported by Michael Neuhauser)
>>
>> - [ASTERISK-28995
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28995>] -
>>
>> res_pjsip_registrar: Expires on statically configured contacts is not
>> correct
>> (Reported by tootai)
>>
>> - [ASTERISK-28987
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28987>] -
>>
>> BridgeCreated ARI event shows wrong video_mode info
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-28978
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28978>] -
>>
>> acl: named_acl rule misconfiguration results in segfault on reading rule
>> from realtime
>> (Reported by Andrew Yager)
>>
>> - [ASTERISK-28975
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28975>] -
>>
>> res_http_websocket: Text payload data doesn't necessary include trailing
>> zero
>> (Reported by Nickolay V. Shmyrev)
>>
>> - [ASTERISK-28951
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28951>] -
>>
>> Inconsistent behaviour queues.conf when there is (not) a [general] section
>> (Reported by Walter Doekes)
>>
>> - [ASTERISK-28965
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28965>] -
>>
>> res_pjsip: Apply outbound proxy to static contacts on AOR
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28930
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28930>] -
>>
>> ./configure --without-ssl build failure
>> (Reported by Jaco Kroon)
>>
>> - [ASTERISK-28957
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28957>] -
>>
>> chan_sip: chan_sip does not process 400 response to an INVITE.
>> (Reported by Frederic LE FOLL)
>>
>> - [ASTERISK-28886
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28886>] -
>>
>> chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
>> (Reported by Jared Smith)
>>
>> - [ASTERISK-28888
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28888>] -
>>
>> res_corosync: causes asterisk crash in huge distributed environment.
>> (Reported by Università di Bologna - CESIA VoIP)
>>
>> - [ASTERISK-28954
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28954>] -
>>
>> StreamEcho() only returns 1 active stream
>> (Reported by Bill Kervaski)
>>
>> - [ASTERISK-28955
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28955>] -
>>
>> "setvar" doesn't work properly in dahdi-channels.conf
>> (Reported by Marin Odrljin)
>>
>> - [ASTERISK-28953
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28953>] -
>>
>> res_pjsip_session: Preserve stream label
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28942
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28942>] -
>>
>> res_sorcery_memory_cache: Individual object expiration behaves
>> unexpectedly with full backend caching
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28950
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28950>] -
>>
>> Stale code in app_queue to check untouched channel
>> (Reported by Walter Doekes)
>>
>> - [ASTERISK-28644
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28644>] -
>>
>> Stale comment in app_queue about ring_entry exception
>> (Reported by Walter Doekes)
>>
>> - [ASTERISK-28952
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28952>] -
>>
>> Queue wrapuptime sometimes not respected (based on stale lastcall time)
>> (Reported by Walter Doekes)
>>
>> - [ASTERISK-28938
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28938>] -
>>
>> core_unreal / core_local: Add support for multistream and re-negotiation
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28948
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28948>] -
>>
>> ARI channel create doesn't referencing the channel_id parameter
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-28939
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28939>] -
>>
>> res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28944
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28944>] -
>>
>> bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly
>> doesn't re-negotiation
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28923
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28923>] -
>>
>> T.38 Segfaults in chan_pjsip_queryoption
>> (Reported by Yury Kirsanov)
>>
>> - [ASTERISK-28940
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28940>] -
>>
>> /channels/create doesn't get any parameters from the body
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-28936
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28936>] -
>>
>> res_pjsip: crash when dialing non-sip uri
>> (Reported by Walter Doekes)
>>
>> - [ASTERISK-28900
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28900>] -
>>
>> res_fax: Double frame free when gateway in use with off-nominal format
>> usage
>> (Reported by Gregory Massel)
>>
>> - [ASTERISK-28929
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28929>] -
>>
>> pjproject_bundled: Honor --without-pjproject.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28932
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28932>] -
>>
>> res_pjsip_logger writing too big packets
>> (Reported by nappsoft)
>>
>> - [ASTERISK-28920
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28920>] -
>>
>> bridge show all causes crash
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-28921
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28921>] -
>>
>> Wrong return value check for fwrite when writing to pcap file
>> (Reported by nappsoft)
>>
>> - [ASTERISK-28794
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28794>] -
>>
>> res_pjsip: Crash when escaping during URI printing
>> (Reported by nappsoft)
>>
>> - [ASTERISK-28884
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28884>] -
>>
>> x-ast-orig-host not filtered out from request URI and To header
>> (Reported by nappsoft)
>>
>> - [ASTERISK-28871
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28871>] -
>>
>> res_pjsip_session: Unnecessary re-Invite on call answer
>> (Reported by Alexei Gradinari)
>>
>> - [ASTERISK-28903
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28903>] -
>>
>> res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28898
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28898>] -
>>
>> bridge_softmix: Conference bridge not passing silent rtp packets
>> (Reported by Jonathan Hunter)
>>
>> - [ASTERISK-28892
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28892>] -
>>
>> res_musiconhold: Module res_musiconhold throws false warning
>> (Reported by Nicholas John Koch)
>>
>> - [ASTERISK-28904
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28904>] -
>>
>> RTP ICE leaks the memory
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-26780
>> <https://issues.asterisk.org/jira/browse/ASTERISK-26780>] -
>>
>> res_pjsip: PJSIP Registration Fails when transport=transport-udp6
>> (Reported by Peter Sokolov)
>>
>> - [ASTERISK-28854
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28854>] -
>>
>> SIGSEGV when pjsip show history encounters IPV6 address
>> (Reported by Roger James)
>>
>> - [ASTERISK-28797
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28797>] -
>>
>> [patch] tcptls: Fix notice when TLS is enabled but not configured.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28804
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28804>] -
>>
>> [patch] app_osplookup.c: Avoid a format truncation.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28776
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28776>] -
>>
>> Non async-signal-safe syscalls used after fork before exec
>> (Reported by nappsoft)
>>
>> - [ASTERISK-28870
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28870>] -
>>
>> streams: One memory leak and one issue cloning streams
>> (Reported by George Joseph)
>>
>> - [ASTERISK-28829
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28829>] -
>>
>> app_queue: leaking stasis subscription when Redirecting call
>> (Reported by lvl)
>>
>> - [ASTERISK-25844
>> <https://issues.asterisk.org/jira/browse/ASTERISK-25844>] -
>>
>> app_queue: Ghost channels in "core show channels" output
>> (Reported by Etienne Lessard)
>>
>> - [ASTERISK-28859
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28859>] -
>>
>> pjsip: Increase maximum candidate count
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-22920
>> <https://issues.asterisk.org/jira/browse/ASTERISK-22920>] -
>>
>> Crash while Forwarding from TLS extension with CHANNEL args
>> secure_bridge_media and secure_bridge_signaling
>> (Reported by Shlomi Gutman)
>>
>> - [ASTERISK-28852
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28852>] -
>>
>> Unprotected access to nochecksums variable, causes build failures
>> (Reported by Guido Falsi)
>>
>> - [ASTERISK-28848
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28848>] -
>>
>> app_fax: Compile.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28846
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28846>] -
>>
>> stream: Enforce formats immutability
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28847
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28847>] -
>>
>> ARI channels cuts the endpoint string over 80 characters
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-28811
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28811>] -
>>
>> Crash occurs when fax session switches from T.38 to audio
>> (Reported by Alexey Vasilyev)
>>
>> - [ASTERISK-28839
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28839>] -
>>
>> Sporadic crashes with Segmentation fault
>> (Reported by Joeran Vinzens)
>>
>> - [ASTERISK-28835
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28835>] -
>>
>> IPv6 addresses in SDP incorrectly formatted
>> (Reported by Daniel Heckl)
>>
>> - [ASTERISK-28372
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28372>] -
>>
>> Asterisk REPLY Wrong Contact header port (TCP)
>> (Reported by Anton Satskiy)
>>
>> - [ASTERISK-24428
>> <https://issues.asterisk.org/jira/browse/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
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28838>] -
>>
>> AST_MODULE_INFO requires, MODULEINFO does not mention
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28841
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28841>] -
>>
>> app_confbridge: Add support for disabling text messaging for a user
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28837
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28837>] -
>>
>> pjproject_bundled: Honor --without-pjproject.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28827
>> <https://issues.asterisk.org/jira/browse/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
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27195>] -
>>
>> chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets
>> (Reported by Joshua Roys)
>>
>> - [ASTERISK-28826
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28826>] -
>>
>> res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK
>> (Reported by nappsoft)
>>
>> - [ASTERISK-28812
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28812>] -
>>
>> First DTMF is not get
>> (Reported by Bernard Merindol)
>>
>> - [ASTERISK-28758
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28758>] -
>>
>> pjsip startup errors when using "with-ssl" configure option
>> (Reported by Patrick Wakano)
>>
>> - [ASTERISK-28824
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28824>] -
>>
>> BuildSystem: Search for Python/C API when possibly needed only.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-27717
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27717>] -
>>
>> [patch] BuildSystem: In NetBSD, the Python Programming Language is
>> python-2.7.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28817
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28817>] -
>>
>> chan_pjsip: constant DTMF tone if RTP is not setup yet
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28819
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28819>] -
>>
>> [patch] bridge_softmix_binaural: Show state in menuselect.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28816
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28816>] -
>>
>> [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28818
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28818>] -
>>
>> [patch] BuildSystem: Allow space in path.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28809
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28809>] -
>>
>> [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28796
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28796>] -
>>
>> func_channel: cannot read fields exten, context, userfield, channame from
>> dialplan
>> (Reported by Sébastien Duthil)
>>
>> - [ASTERISK-28803
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28803>] -
>>
>> [patch] chan_unistim: Avoid tautological warnings with clang.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28808
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28808>] -
>>
>> [patch] test_stasis: Avoid always true warning with clang.
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28056
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28056>] -
>>
>> res_pjsip: Incorrect endpoint status after endpoint synchronization for a
>> specific AOR
>> (Reported by Jason Hord)
>>
>> - [ASTERISK-28795
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28795>] -
>>
>> channel: write to a stream on multi-frame writes
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28789
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28789>] -
>>
>> test_utils: incorrectly printing error 'declined to load'
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28788
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28788>] -
>>
>> func_aes: incorrectly printing error 'declined to load'
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28790
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28790>] -
>>
>> Crash during conference call using confbridge and video
>> (Reported by Pascal Cadotte Michaud)
>>
>> - [ASTERISK-16676
>> <https://issues.asterisk.org/jira/browse/ASTERISK-16676>] -
>>
>> DAHDIRAS fails to properly initiate pppd unless asterisk is running as root
>> (Reported by Jaco Kroon)
>>
>> - [ASTERISK-21205
>> <https://issues.asterisk.org/jira/browse/ASTERISK-21205>] -
>>
>> [patch] dundi_read_result crash due to negative number
>> (Reported by Jaco Kroon)
>>
>> - [ASTERISK-28784
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28784>] -
>>
>> res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28743
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28743>] -
>>
>> Asterisk is crashing if the 200 OK with SDP
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-28783
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28783>] -
>>
>> res_pjsip_session: Allow default non-audio streams to have reflected state
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28774
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28774>] -
>>
>> chan_pjsip's rtptimeout is erroneously triggered during direct-media
>> (native_rtp) bridge
>> (Reported by Michael Neuhauser)
>>
>> - [ASTERISK-20325
>> <https://issues.asterisk.org/jira/browse/ASTERISK-20325>] -
>>
>> Comments in configs/func_odbc.conf.sample are not consistent with
>> examples. Missing examples.
>> (Reported by Olivier Krief)
>>
>> - [ASTERISK-28780
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28780>] -
>>
>> app_mixmonitor: Memory leak due to race condition between AMI MixMonitor
>> and hangup
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28773
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28773>] -
>>
>> Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
>> (Reported by Torrey Searle)
>>
>> - [ASTERISK-28769
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28769>] -
>>
>> DTLS Handshake Fails to Occur if ice_support is enabled but not used
>> (Reported by Torrey Searle)
>>
>> - [ASTERISK-28759
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28759>] -
>>
>> A non negotiated rtp frame causes call disconnection when there is a SSRC
>> change
>> (Reported by Paulo Vicentini)
>>
>> - [ASTERISK-26711
>> <https://issues.asterisk.org/jira/browse/ASTERISK-26711>] -
>>
>> func_enum: ENUM code wrong case
>> (Reported by Vitold)
>>
>> - [ASTERISK-23407
>> <https://issues.asterisk.org/jira/browse/ASTERISK-23407>] -
>>
>> Fix the FSF address in the headers of lots of pjproject files
>> (Reported by Jared Smith)
>>
>> - [ASTERISK-19460
>> <https://issues.asterisk.org/jira/browse/ASTERISK-19460>] -
>>
>> [patch] Function TXTCIDNAME never actually makes DNS calls and always
>> returns an empty string
>> (Reported by George Joseph)
>>
>> - [ASTERISK-28766
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28766>] -
>>
>> PJSIP blind transfer not completed after using Proceeding()
>> (Reported by lvl)
>>
>> - [ASTERISK-28764
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28764>] -
>>
>> res_rtp_asterisk: Improve NACK support and seqno handling
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28755
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28755>] -
>>
>> SIP/Stasis: SIP headers not transmitted in the "variables" field
>> (Reported by Jean Aunis - Prescom)
>>
>> - [ASTERISK-28685
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28685>] -
>>
>> check_expr2: linking (when hardening) and cross-compiling troubles
>> (Reported by Sebastian Kemper)
>>
>> - [ASTERISK-28754
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28754>] -
>>
>> ASTERISK-28738 Causes Audio Issue After Hold
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-28697
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28697>] -
>>
>> res_pjsip: Named ACL does not update on reload if changed
>> (Reported by Timothy Vanderaerden)
>>
>> - [ASTERISK-28746
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28746>] -
>>
>> res_pjsip_outbound_registration keeps retrying the first entry in a SRV
>> record set
>> (Reported by George Joseph)
>>
>> - [ASTERISK-28716
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28716>] -
>>
>> ICE: pjnath shouldn't wait for ICE to complete before allowing sending
>> (Reported by Benjamin Keith Ford)
>>
>> - [ASTERISK-28738
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28738>] -
>>
>> Incorrect state machine used when MOH_PASSTHRU is used
>> (Reported by Torrey Searle)
>>
>> - [ASTERISK-28742
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28742>] -
>>
>> res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28735
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28735>] -
>>
>> Realtime MoH Unknown format '' -- defaulting to SLIN
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-28730
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28730>] -
>>
>> res_pjsip_session: Fix out of order session refreshes
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-26955
>> <https://issues.asterisk.org/jira/browse/ASTERISK-26955>] -
>>
>> pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited
>> by "[]" Rejected
>> (Reported by Peter Sokolov)
>>
>> - [ASTERISK-28718
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28718>] -
>>
>> chan_sip: Returns 403 if RTP ports are depleted, should return 503
>> (Reported by Walter Doekes)
>>
>> - [ASTERISK-28713
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28713>] -
>>
>> res_stasis_playback: Error building JSON
>> (Reported by Sébastien Duthil)
>>
>> - [ASTERISK-28714
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28714>] -
>>
>> REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759)
>> Causes Segfaults
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-26082
>> <https://issues.asterisk.org/jira/browse/ASTERISK-26082>] -
>>
>> res_pjsip_messaging: MessageSend Content-Type can't be changed
>> (Reported by Alex)
>>
>> - [ASTERISK-28423
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28423>] -
>>
>> ARI causes STASIS Deadlock
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-28679
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28679>] -
>>
>> stasis application is destroyed after its creation
>> (Reported by Francois Blackburn)
>>
>> - [ASTERISK-25421
>> <https://issues.asterisk.org/jira/browse/ASTERISK-25421>] -
>>
>> PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when
>> sending
>> (Reported by Dmitriy Serov)
>>
>> - [ASTERISK-28686
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28686>] -
>>
>> chan_sip strictrtp=yes fails when media source is changed: no audio
>> (Reported by Walter Doekes)
>>
>> - [ASTERISK-28139
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28139>] -
>>
>> RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
>> (Reported by Paul Brooks)
>>
>> - [ASTERISK-28677
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28677>] -
>>
>> CDR billsec is always 0 for transferred calls
>> (Reported by Maciej Michno)
>>
>> - [ASTERISK-28702
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28702>] -
>>
>> chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
>> (Reported by Andrew Siplas)
>>
>> - [ASTERISK-24484
>> <https://issues.asterisk.org/jira/browse/ASTERISK-24484>] -
>>
>> Update documentation for statsd module - usage requirements unclear
>> (Reported by Dan Jenkins)
>>
>> - [ASTERISK-28706
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28706>] -
>>
>> silk 24hHz doesn't show up in 'core show translation' output
>> (Reported by Sean Bright)
>>
>> - [ASTERISK-28695
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28695>] -
>>
>> core: minmemfree watermark uses free RAM, not available RAM
>> (Reported by Kevin Flyn)
>>
>> - [ASTERISK-28693
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28693>] -
>>
>> chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the
>> dialplan
>> (Reported by Frank Matano)
>>
>> - [ASTERISK-23739
>> <https://issues.asterisk.org/jira/browse/ASTERISK-23739>] -
>>
>> [patch]Segfault forwarding voicemail with ODBC storage enabled and
>> realtime voicemail_data is used
>> (Reported by Stas Kobzar)
>>
>> - [ASTERISK-27622
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27622>] -
>>
>> empty voicemail.conf required for ARA (realtime) voicemail to leave message
>> (Reported by Jim Van Meggelen)
>>
>> - [ASTERISK-21794
>> <https://issues.asterisk.org/jira/browse/ASTERISK-21794>] -
>>
>> CLI command 'realtime update2' syntax failure when using according to
>> usage help
>> (Reported by Cedric BASSAGET)
>>
>> - [ASTERISK-28349
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28349>] -
>>
>> Pause reason not reported in QueueMember AMI event
>> (Reported by Niksa Baldun)
>>
>> - [ASTERISK-25429
>> <https://issues.asterisk.org/jira/browse/ASTERISK-25429>] -
>>
>> res_pjsip_endpoint_identifier_ip: Document support for hostnames
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-27775
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27775>] -
>>
>> res_pjsip_notify: Multiple Event headers can be present instead of just one
>> (Reported by AvayaXAsterisk)
>>
>> - [ASTERISK-28682
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28682>] -
>>
>> app_record: Lack of `beep` audio file causes application to return error
>> and hangup
>> (Reported by Corey Farrell)
>>
>> - [ASTERISK-28507
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28507>] -
>>
>> Wiki docs missing for MessageWaiting
>> (Reported by David M. Lee)
>>
>> - [ASTERISK-27759
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27759>] -
>>
>> res_pjsip_pubsub: Subscription persistence does not preserve XML version
>> number
>> (Reported by Bryan Nelson)
>>
>> - [ASTERISK-28605
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28605>] -
>>
>> chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show
>> span X
>> (Reported by Dirk Wendland)
>>
>> - [ASTERISK-28633
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28633>] -
>>
>> stasis bridge topic leak
>> (Reported by Joeran Vinzens)
>>
>> - [ASTERISK-28492
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28492>] -
>>
>> pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
>> (Reported by Jean-Denis Girard)
>>
>> - [ASTERISK-28562
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28562>] -
>>
>> SIP WSS message not processed until next frame arrives
>> (Reported by Robert Sutton)
>>
>> - [ASTERISK-28667
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28667>] -
>>
>> Asterisk ignores parsing of config files if a Byte order mark is present
>> (Reported by Robin Leffmann)
>>
>> - [ASTERISK-28625
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28625>] -
>>
>> Playback of local files impacted by large media cache
>> (Reported by Kevin Reeves)
>>
>> - [ASTERISK-27243
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27243>] -
>>
>> contrib: valgrind.supp doesn't suppress what it's supposed to due to
>> invalid syntax
>> (Reported by Richard Kenner)
>>
>> - [ASTERISK-28664
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28664>] -
>>
>> "trustrpid" is misspelled in sip_to_pjsip.py
>> (Reported by Pascal Cadotte Michaud)
>>
>> - [ASTERISK-28636
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28636>] -
>>
>> app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
>> (Reported by Frederic LE FOLL)
>>
>> - [ASTERISK-28604
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28604>] -
>>
>> app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
>> (Reported by George Joseph)
>>
>> - [ASTERISK-28659
>> <https://issues.asterisk.org/jira/browse/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
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28660>] -
>>
>> res_fax: wrap Asterisk initiated negotiation with config option
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28626
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28626>] -
>>
>> Missing arguments in PJSIP_CONTACT function documentation
>> (Reported by Pascal Cadotte Michaud)
>>
>> - [ASTERISK-28609
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28609>] -
>>
>> Memory Leak in res_rtp_asterisk.c
>> (Reported by Ted G)
>>
>> - [ASTERISK-28651
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28651>] -
>>
>> chan_sip logs errors on tx to non-existent TCP connections
>> (Reported by Jaco Kroon)
>>
>> - [ASTERISK-28502
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28502>] -
>>
>> chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-28641
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28641>] -
>>
>> res_pjsip Segfaults when realtime configuration to an AOR points to a not
>> existent AOR
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-28647
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28647>] -
>>
>> chan_sip: RTP frames not transmitted after emitting a COLP
>> (Reported by Jean Aunis - Prescom)
>>
>> - [ASTERISK-28637
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28637>] -
>>
>> chan_sip+native_bridge_rtp: directmedia compatibility check failure when
>> negociated ptime is not default ptime.
>> (Reported by Frederic LE FOLL)
>>
>> - [ASTERISK-28445
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28445>] -
>>
>> res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when
>> TEST_FRAMEWORK enabled
>> (Reported by Bernhard Schmidt)
>>
>> - [ASTERISK-28631
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28631>] -
>>
>> res_parking: Doesn't park when parkee and parker are the same
>> (Reported by Ross Beer)
>>
>> - [ASTERISK-28621
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28621>] -
>>
>> Enforce T.38 error correction mode at 200 ok received
>> (Reported by Salah Ahmed)
>>
>> - [ASTERISK-28624
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28624>] -
>>
>> res_pjsip_outbound_registration: add SRV failover
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28608
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28608>] -
>>
>> app_amd: Use time calculation to calculate timeout
>> (Reported by Michael Cargile)
>>
>> - [ASTERISK-28615
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28615>] -
>>
>> chan_dahdi: PRI span status may stay "Down, Active" after a short alarm
>> (Reported by Frederic LE FOLL)
>>
>> - [ASTERISK-28576
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28576>] -
>>
>> res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't
>> match
>> (Reported by Joshua Elson)
>>
>> - [ASTERISK-26481
>> <https://issues.asterisk.org/jira/browse/ASTERISK-26481>] -
>>
>> FILE function grabs garbage along with read data when target line has no
>> newline
>> (Reported by Jonathan Harris)
>>
>> - [ASTERISK-28618
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28618>] -
>>
>> bridge_softmix: hold not cleared when joining a softmix bridge
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28616
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28616>] -
>>
>> parking: Deadlock when multi call parking
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28572
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28572>] -
>>
>> Memory leaks in res_calendar_exchange and res_calendar_icalendar
>> (Reported by Yoooooo Ha)
>>
>> - [ASTERISK-28585
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28585>] -
>>
>> ari/resource_events: Crash in event session cleanup
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28590
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28590>] -
>>
>> utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid
>> argument"
>> (Reported by Speed Dial Dave)
>>
>> - [ASTERISK-28578
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28578>] -
>>
>> race condition on pjsip channelstats command
>> (Reported by Salah Ahmed)
>>
>> - [ASTERISK-28571
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28571>] -
>>
>> cdr_pgsql: accesses obsolete (and finally removed) column
>> (Reported by Christoph Moench-Tegeder)
>>
>> - [ASTERISK-28575
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28575>] -
>>
>> MWI Send Notify Crash on 16.6
>> (Reported by Joshua Elson)
>>
>> - [ASTERISK-28574
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28574>] -
>>
>> pjproject fails to build on 16.6.0, works on 16.5
>> (Reported by Niklas Larsson)
>>
>> - [ASTERISK-28561
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28561>] -
>>
>> Asterisk Deadlocks
>> (Reported by Aheliotech)
>>
>> - [ASTERISK-28086
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28086>] -
>>
>> chan_pjsip: Crash when initiating PlayDTMF over AMI
>> (Reported by Jeremiah Gadd)
>>
>> - [ASTERISK-28552
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28552>] -
>>
>> res_pjsip_mwi: Frack during unload on unsolicited_mwi container
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28566
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28566>] -
>>
>> CDR backend unload problem during active call(s)
>> (Reported by Marian Piater)
>>
>> - [ASTERISK-28553
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28553>] -
>>
>> stasis.c: Crash during unload
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28544
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28544>] -
>>
>> Wrong contact representation in ipv6 mode
>> (Reported by Jørgen H)
>>
>> - [ASTERISK-28534
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28534>] -
>>
>> Segmentation fault when there is no priority for an extension
>> (Reported by Timothy Vanderaerden)
>>
>> - [ASTERISK-28463
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28463>] -
>>
>> res_pjsip_path: Crash when invalid contact is configured
>> (Reported by Juan Martin)
>>
>> - [ASTERISK-28521
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28521>] -
>>
>> pjsip: Memory Leak
>> (Reported by Mark)
>>
>> - [ASTERISK-28523
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28523>] -
>>
>> Asterisk 16.5.0 Memory leak
>> (Reported by Cyril Ramière)
>>
>> - [ASTERISK-28536
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28536>] -
>>
>> Asterisk release candidates fail to build on FreeBSD
>> (Reported by Guido Falsi)
>>
>> - [ASTERISK-28538
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28538>] -
>>
>> chan_pjsip: Deadlock on fax detection
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28497
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28497>] -
>>
>> func_odbc: truncating Unicode string on readsql
>> (Reported by Boris P. Korzun)
>>
>> - [ASTERISK-23756
>> <https://issues.asterisk.org/jira/browse/ASTERISK-23756>] -
>>
>> setvar directive when used in template and a child of said template,
>> results in duplicate variable names
>> (Reported by Michael Goryainov)
>>
>> - [ASTERISK-28527
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28527>] -
>>
>> ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf
>> (Reported by Frederic LE FOLL)
>>
>> - [ASTERISK-28525
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28525>] -
>>
>> chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up
>> (Reported by Frederic LE FOLL)
>>
>> - [ASTERISK-28511
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28511>] -
>>
>> codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32
>> (Reported by Ruddy G)
>>
>> - [ASTERISK-28499
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28499>] -
>>
>> translate: Crash when frame does not have a "src" field set
>> (Reported by Gregory Massel)
>>
>> - [ASTERISK-25592
>> <https://issues.asterisk.org/jira/browse/ASTERISK-25592>] -
>>
>> chan_unistim: Clang Warning: variable sized type not at end of a struct
>> (Reported by Alexander Traud)
>>
>> - [ASTERISK-28488
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28488>] -
>>
>> pjsip mwi: n+1 sip notify's sent on re-register
>> (Reported by Chris Savinovich)
>>
>> - [ASTERISK-28509
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28509>] -
>>
>> PJSIP cnonce generated on Linux contains 36 characters, NEC only supports
>> up to 32 characters
>> (Reported by Dan Cropp)
>>
>> - [ASTERISK-28505
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28505>] -
>>
>> app_voicemail/IMAP: segfault in leave_voicemail because not checking
>> mailstream
>> (Reported by Alexei Gradinari)
>>
>> - [ASTERISK-28487
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28487>] -
>>
>> compile menuselect on gentoo
>> (Reported by Kilburn)
>>
>> - [ASTERISK-28472
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28472>] -
>>
>> Asterisk occasionally passes a NULL as srtp->session to
>> srtp_protect/unprotect causing SEGV
>> (Reported by Jonas Swiatek)
>>
>> - [ASTERISK-28498
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28498>] -
>>
>> cel / cdr: Event times may be incorrect
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28480
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28480>] -
>>
>> json integer overflow in ssrc and timestamp
>> (Reported by Salah Ahmed)
>>
>> - [ASTERISK-28228
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28228>] -
>>
>> res_pjsip: pjsip show contacts prints double entries
>> (Reported by Ian Jones)
>>
>> - [ASTERISK-28483
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28483>] -
>>
>> packet lost on UDPTL wrap around
>> (Reported by Torrey Searle)
>>
>> *Improvements made in this release:*
>> -----------------------------------
>>
>> - [ASTERISK-28959
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28959>] -
>>
>> res_pjsip: Added option for disable rport parameter set
>> (Reported by sungtae kim)
>>
>> - [ASTERISK-28958
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28958>] -
>>
>> Continue reading string when ping received by websocket
>> (Reported by Nickolay V. Shmyrev)
>>
>> - [ASTERISK-28945
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28945>] -
>>
>> AMI SendText - add Content-Type parameter
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28949
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28949>] -
>>
>> res_http_websocket: Add masking to websocket client
>> (Reported by Moises Silva)
>>
>> - [ASTERISK-28899
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28899>] -
>>
>> Upgrade Asterisk to bundled pjproject 2.10
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28895
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28895>] -
>>
>> res_pjsip_logger: Add tons'o'functionality
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28896
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28896>] -
>>
>> ari: Add support for specifying variables on channel create
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28879
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28879>] -
>>
>> pjproject has race conditions in it's build system
>> (Reported by Guido Falsi)
>>
>> - [ASTERISK-28866
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28866>] -
>>
>> third-party/pjproject/configure.m4 contains bashisms
>> (Reported by Guido Falsi)
>>
>> - [ASTERISK-28853
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28853>] -
>>
>> Missing include on FreeBSD
>> (Reported by Guido Falsi)
>>
>> - [ASTERISK-28832
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28832>] -
>>
>> chan_mobile creates PCMA streams that make some VoIP clients crash or not
>> render received audio
>> (Reported by Peter Turczak)
>>
>> - [ASTERISK-28813
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28813>] -
>>
>> func_volume: Allow decimal numbers as parameter to improve granularity
>> (Reported by Jean Aunis - Prescom)
>>
>> - [ASTERISK-28777
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28777>] -
>>
>> Codec Negotiation: add outgoing_call_offer_prefs option
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-27946
>> <https://issues.asterisk.org/jira/browse/ASTERISK-27946>] -
>>
>> dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it
>> shouldn't
>> (Reported by Joshua Elson)
>>
>> - [ASTERISK-28782
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28782>] -
>>
>> Add support for Content-Disposition header in multi-part INVITES
>> (Reported by Torrey Searle)
>>
>> - [ASTERISK-28787
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28787>] -
>>
>> res_pjsip_session: Decide more intelligently when to add video
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28756
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28756>] -
>>
>> Codec Negotiation: add incoming_call_offer_pref option
>> (Reported by Kevin Harwell)
>>
>> - [ASTERISK-28750
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28750>] -
>>
>> TLS/SSL Key too small error
>> (Reported by Martin Zeh)
>>
>> - [ASTERISK-28733
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28733>] -
>>
>> stream: Add support for adding/removing streams during SFU/calls
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-24798
>> <https://issues.asterisk.org/jira/browse/ASTERISK-24798>] -
>>
>> Documentation - Clarify That Format Is Set By File Name Extension In
>> MixMonitor
>> (Reported by xrobau)
>>
>> - [ASTERISK-28726
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28726>] -
>>
>> install_prereq script uses the interactive mode when installing aptitude
>> (Reported by Sylvain Afchain)
>>
>> - [ASTERISK-28710
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28710>] -
>>
>> Should be able to disable the /httpstatus URI in the built-in HTTP server
>> (Reported by Sean Bright)
>>
>> - [ASTERISK-28484
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28484>] -
>>
>> Add AudioSocket support
>> (Reported by Seán C. McCord)
>>
>> - [ASTERISK-28638
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28638>] -
>>
>> Simplify dialplan for Dial, Page, and ChanIsAvail
>> (Reported by cmaj)
>>
>> - [ASTERISK-28673
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28673>] -
>>
>> GET FULL VARIABLE documentation clarification
>> (Reported by Jonathan Harris)
>>
>> - [ASTERISK-28629
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28629>] -
>>
>> [patch] Add an "inhibitCOLP" flag to the bridges REST API
>> (Reported by Jean Aunis - Prescom)
>>
>> - [ASTERISK-28658
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28658>] -
>>
>> app_confbridge: Add support for setting maximum sample rate
>> (Reported by Joshua C. Colp)
>>
>> - [ASTERISK-28602
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28602>] -
>>
>> res_pjsip_outbound_registration: Maximum retries reached
>> (Reported by Daniel)
>>
>> - [ASTERISK-28586
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28586>] -
>>
>> Typo in README-SERIOUSLY.bestpractices.md
>> (Reported by Sam Banks)
>>
>> - [ASTERISK-22192
>> <https://issues.asterisk.org/jira/browse/ASTERISK-22192>] -
>>
>> [patch] Allow voicemail forwards with ODBC backend when format differs
>> from attachfmt column
>> (Reported by cmaj)
>>
>> - [ASTERISK-28567
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28567>] -
>>
>> Problem with ASTERISK-20207: Asterisk should clear out any .lock files in
>> the voice mail directory on startup.
>> (Reported by Michael)
>>
>> - [ASTERISK-28542
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28542>] -
>>
>> [patch] add the ability for asterisk to generate on-hold re-invites
>> (Reported by Torrey Searle)
>>
>> - [ASTERISK-28512
>> <https://issues.asterisk.org/jira/browse/ASTERISK-28512>] -
>>
>> Add pass-through support for H.265 (HEVC) codec
>> (Reported by Florian Floimair)
>>
>> For a full list of changes in this release, please see the ChangeLog:
>> https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
>>
>> *Thank you for your continued support of Asterisk!*
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
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> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
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> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
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