[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 13.17.0-rc1 Now Available

[Sylvio Jollenbeck]

The Asterisk Development Team would like to announce the first release
candidate of Asterisk 13.17.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.17.0-rc1 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 candidate:

*Bugs fixed in this release:*
-----------------------------------
- [ASTERISK-25665 <https://issues.asterisk.org/jira/browse/ASTERISK-25665>]
- Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
- [ASTERISK-27074 <https://issues.asterisk.org/jira/browse/ASTERISK-27074>]
- core_local: local channel data not being properly unref'ed and unlocked
(Reported by Kevin Harwell)
- [ASTERISK-27075 <https://issues.asterisk.org/jira/browse/ASTERISK-27075>]
- bridge: stuck channel(s) after failed attended transfer
(Reported by Kevin Harwell)
- [ASTERISK-24052 <https://issues.asterisk.org/jira/browse/ASTERISK-24052>]
- app_voicemail reloads result in leaked IMAP sockets.
(Reported by Louis Jocelyn Paquet)
- [ASTERISK-27060 <https://issues.asterisk.org/jira/browse/ASTERISK-27060>]
- Comment typo format_g729.c
(Reported by Matthew Fredrickson)
- [ASTERISK-27026 <https://issues.asterisk.org/jira/browse/ASTERISK-27026>]
- res_ari: Crash when no ari.conf configuration file exists
(Reported by Ronald Raikes)
- [ASTERISK-27041 <https://issues.asterisk.org/jira/browse/ASTERISK-27041>]
- Core/PBX: [patch] Deadlock between dialplan execution and application
unregistration
(Reported by Frederic LE FOLL)
- [ASTERISK-27057 <https://issues.asterisk.org/jira/browse/ASTERISK-27057>]
- Seg Fault in ast_sorcery_object_get_id at sorcery.c
(Reported by Ryan Smith)
- [ASTERISK-27024 <https://issues.asterisk.org/jira/browse/ASTERISK-27024>]
- nat/external_media settings ignored in 14.4.1
(Reported by Christopher van de Sande)
- [ASTERISK-27046 <https://issues.asterisk.org/jira/browse/ASTERISK-27046>]
- res_pjsip_transport_websocket: segfault in get_write_timeout
(Reported by Jørgen H)
- [ASTERISK-27022 <https://issues.asterisk.org/jira/browse/ASTERISK-27022>]
- res_rtp_asterisk: Incorrect SSRC change for RTCP component
(Reported by Michael Walton)
- [ASTERISK-26923 <https://issues.asterisk.org/jira/browse/ASTERISK-26923>]
- bridging: T.38 request is lost when channels are added to bridge
(Reported by Torrey Searle)
- [ASTERISK-27053 <https://issues.asterisk.org/jira/browse/ASTERISK-27053>]
- res_pjsip_refer/session: Calls dropped during transfer
(Reported by Kevin Harwell)
- [ASTERISK-27052 <https://issues.asterisk.org/jira/browse/ASTERISK-27052>]
- Asterisk build process fails with flag --with-pjproject-bundled with curl
download command and slow network
(Reported by alex)
- [ASTERISK-27039 <https://issues.asterisk.org/jira/browse/ASTERISK-27039>]
- chan_pjsip: Device state is idle when channel from endpoint is in early
media
(Reported by Joshua Colp)
- [ASTERISK-26996 <https://issues.asterisk.org/jira/browse/ASTERISK-26996>]
- chan_pjsip: Flipping between codecs
(Reported by Michael Maier)
- [ASTERISK-26281 <https://issues.asterisk.org/jira/browse/ASTERISK-26281>]
- chan_pjsip would send INVITE to 'Unreachable' endpoints
(Reported by Jacek Konieczny)
- [ASTERISK-26973 <https://issues.asterisk.org/jira/browse/ASTERISK-26973>]
- bridge: Crash when freeing frame and snooping
(Reported by Michel R. Vaillancourt)
- [ASTERISK-19291 <https://issues.asterisk.org/jira/browse/ASTERISK-19291>]
- Background in realtime
(Reported by Andrew Nowrot)
- [ASTERISK-27025 <https://issues.asterisk.org/jira/browse/ASTERISK-27025>]
- channel / meetme: Fix missing parentheses
(Reported by Joshua Colp)
- [ASTERISK-27021 <https://issues.asterisk.org/jira/browse/ASTERISK-27021>]
- GET /recordings/stored returns 500 Internal Server Error
(Reported by Tim Morgan)
- [ASTERISK-24858 <https://issues.asterisk.org/jira/browse/ASTERISK-24858>]
- [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel
platform when using slin codec
(Reported by Frankie Chin)
- [ASTERISK-23951 <https://issues.asterisk.org/jira/browse/ASTERISK-23951>]
- Asterisk attempts and fails to build format_mp3 even if mp3lib was not
downloaded
(Reported by Tzafrir Cohen)
- [ASTERISK-25294 <https://issues.asterisk.org/jira/browse/ASTERISK-25294>]
- srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)
- [ASTERISK-23839 <https://issues.asterisk.org/jira/browse/ASTERISK-23839>]
- AGI - RECORD FILE - documentation doesn't describe BEEP argument
(Reported by Rusty Newton)
- [ASTERISK-22432 <https://issues.asterisk.org/jira/browse/ASTERISK-22432>]
- Async AGI crashes Asterisk when issuing "set variable" command without
args
(Reported by Antoine Pitrou)
- [ASTERISK-25662 <https://issues.asterisk.org/jira/browse/ASTERISK-25662>]
- Malformed AGI 520 Usage response
(Reported by Tony Mountifield)
- [ASTERISK-25101 <https://issues.asterisk.org/jira/browse/ASTERISK-25101>]
- DTLS configuration can not be specified in the general section -
documentation
(Reported by Ben Langfeld)
- [ASTERISK-27008 <https://issues.asterisk.org/jira/browse/ASTERISK-27008>]
- res_format_attr_h264: SDP parse fails if fmtp optional parameters have a
space
(Reported by John Harris)
- [ASTERISK-26399 <https://issues.asterisk.org/jira/browse/ASTERISK-26399>]
- app_queue: Agent not called when caller is parked
(Reported by wushumasters)
- [ASTERISK-26400 <https://issues.asterisk.org/jira/browse/ASTERISK-26400>]
- app_queue: Queue member stops being called after AMI "Redirect" action
for queues with wrapuptime
(Reported by Etienne Lessard)
- [ASTERISK-26715 <https://issues.asterisk.org/jira/browse/ASTERISK-26715>]
- app_queue: Member will not receive any new calls after doing a transfer
if wrapuptime = greater than 0 and using Local channel
(Reported by David Brillert)
- [ASTERISK-26975 <https://issues.asterisk.org/jira/browse/ASTERISK-26975>]
- app_queue: Non-zero wrapup time can cause agents not to receive queue
calls after transfer queue call
(Reported by Lorne Gaetz)
- [ASTERISK-27012 <https://issues.asterisk.org/jira/browse/ASTERISK-27012>]
- app_confbridge: ConfBridge sometimes does not play user name recording
while leaving
(Reported by Robert Mordec)
- [ASTERISK-26979 <https://issues.asterisk.org/jira/browse/ASTERISK-26979>]
- res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or
110
(Reported by Javier Riveros )
- [ASTERISK-26982 <https://issues.asterisk.org/jira/browse/ASTERISK-26982>]
- chan_sip: rtcp_mux setting may cause ice completion failure/delay if
client offers rtcp-mux as negotiable
(Reported by Stefan Engström)
- [ASTERISK-26964 <https://issues.asterisk.org/jira/browse/ASTERISK-26964>]
- res_pjsip_session: Wrong From on reinvite when request and To URI differ
(Reported by Yasin CANER)
- [ASTERISK-26789 <https://issues.asterisk.org/jira/browse/ASTERISK-26789>]
- Audit manipulation of channel flags without locks
(Reported by Joshua Colp)
- [ASTERISK-26333 <https://issues.asterisk.org/jira/browse/ASTERISK-26333>]
- Problems with Blind Transfer, PJSIP (Aastra 6869i)
(Reported by Matthias Binder)

*Information Requests made in this release:*
-----------------------------------
- [ASTERISK-26976 <https://issues.asterisk.org/jira/browse/ASTERISK-26976>]
- libsrtp-2.x.x support
(Reported by Alex)

*Improvements made in this release:*
-----------------------------------
- [ASTERISK-26230 <https://issues.asterisk.org/jira/browse/ASTERISK-26230>]
- [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on
startup
(Reported by Alexei Gradinari)
- [ASTERISK-27043 <https://issues.asterisk.org/jira/browse/ASTERISK-27043>]
- Core/BuildSystem: Add defines to fix build with LibreSSL
(Reported by Guido Falsi)
- [ASTERISK-27042 <https://issues.asterisk.org/jira/browse/ASTERISK-27042>]
- Unpatched asterisk sources fail to build on FreeBSD due to missing
crypt.h file
(Reported by Guido Falsi)
- [ASTERISK-26419 <https://issues.asterisk.org/jira/browse/ASTERISK-26419>]
- audiohooks: Remove redundant codec translations when using audiohooks
(Reported by Michael Walton)
- [ASTERISK-26124 <https://issues.asterisk.org/jira/browse/ASTERISK-26124>]
- res_agi: Set audio format for EAGI audio stream
(Reported by John Fawcett)

For a full list of changes in this release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0-rc1

*Thank you for your continued support of Asterisk!*


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