ASTERWEB Blog

Rilasciato Asterisk 13.7.0-rc2

Il giorno 18 dicembre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.7.0-rc2.

Dal post originale:

Bug

[ASTERISK-25601] - json: Audit reference usage and thread safety
[ASTERISK-25625] - res_sorcery_memory_cache: Add full backend caching

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc2

Rilasciato Asterisk 13.7.0-rc1

Il giorno 15 dicembre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.7.0-rc1.

Dal post originale:

Bug

[ASTERISK-7803] - [patch] Update the maximum packetization values in frame.c
[ASTERISK-24106] - WebSockets Automatically decides what driver it will use
[ASTERISK-24146] - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
[ASTERISK-24543] - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
[ASTERISK-24779] - Passthrough OPUS codec not working with chan_pjsip
[ASTERISK-24958] - Forwarding loop detection inhibits certain desirable scenarios
[ASTERISK-25135] - [patch]RTP Timeout hangup cause code missing
[ASTERISK-25160] - [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
[ASTERISK-25165] - Testsuite - Sorcery memory cache leaks
[ASTERISK-25364] - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
[ASTERISK-25373] - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
[ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
[ASTERISK-25400] - Hints broken when "CustomPresence" doesn't exist in AstDB
[ASTERISK-25404] - segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
[ASTERISK-25434] - Compiler flags not reported in 'core show settings' despite usage during compilation
[ASTERISK-25435] - Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
[ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
[ASTERISK-25441] - Deadlock in res_sorcery_memory_cache.
[ASTERISK-25443] - [patch]IPv6 - Potential issue in via header parsing
[ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
[ASTERISK-25451] - Broken video - erased rtp marker bit
[ASTERISK-25455] - Deadlock of PJSIP realtime over res_config_pgsql
[ASTERISK-25461] - Nested dialplan #includes don't work as expected.
[ASTERISK-25476] - chan_sip loses registrations after a while
[ASTERISK-25484] - [patch] autoframing=yes has no effect
[ASTERISK-25485] - res_pjsip_outbound_registration: registration stops due to 400 response
[ASTERISK-25486] - res_pjsip: Fix deadlock when validating URIs
[ASTERISK-25494] - build: GCC 5.1.x catches some new const, array bounds and missing paren issues
[ASTERISK-25498] - Asterisk crashes when negotiating g729 without that module installed
[ASTERISK-25505] - res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created
[ASTERISK-25513] - Crash: malloc failed with high load of subscriptions.
[ASTERISK-25522] - ARI: Crash when creating channel via ARI originate with requesting channel
[ASTERISK-25527] - Quirky xmldoc description wrapping
[ASTERISK-25533] - [patch] buffer for ast_format_cap_get_names only 64 bytes
[ASTERISK-25535] - [patch] format creation on module load instead of cache
[ASTERISK-25537] - [patch] format-attribute module: RFC or internal defaults?
[ASTERISK-25545] - [patch] translation module gets cached not joint format
[ASTERISK-25546] - threadpool: Race condition between idle timeout and activation
[ASTERISK-25552] - hashtab: Improve NULL tolerance
[ASTERISK-25561] - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!
[ASTERISK-25569] - app_meetme: Audio quality issues
[ASTERISK-25573] - [patch] H.264 format attribute module: resets whole SDP
[ASTERISK-25575] - res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart
[ASTERISK-25582] - Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
[ASTERISK-25583] - [patch] format-attribute module: RFC 7587 (Opus Codec)
[ASTERISK-25584] - [patch] format-attribute module: VP8 missing
[ASTERISK-25585] - [patch]rasterisk never hits most of main(), but it's assumed to
[ASTERISK-25590] - CLI Usage info for 'pjsip send notify' references incorrect config
[ASTERISK-25593] - fastagi: record file closed after sending result
[ASTERISK-25595] - Unescaped : in messge sent to statsd
[ASTERISK-25598] - res_pjsip: Contact status messages are printing a hash instead of the uri
[ASTERISK-25599] - [patch] SLIN Resampling Codec only 80 msec
[ASTERISK-25600] - bridging: Inconsistency in BRIDGEPEER
[ASTERISK-25608] - res_pjsip/contacts/statsd: Lifecycle events aren't consistent
[ASTERISK-25609] - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
[ASTERISK-25610] - Asterisk crash during "sip reload"
[ASTERISK-25615] - res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports
[ASTERISK-25616] - Warning with a Codec Module which supports PLC with FEC
[ASTERISK-25619] - res_chan_stats not sending the correct information to StatsD

Improvement

[ASTERISK-24718] - [patch]Add inital support of "sanitize" to configure
[ASTERISK-25477] - pjsip show "command" like [criteria]
[ASTERISK-25518] - taskprocessor: Add high water mark
[ASTERISK-25571] - PJSIP: Add StatsD stats for some common PJSIP objects
[ASTERISK-25572] - Endpoints: Add StatsD stats for Asterisk endpoints
[ASTERISK-25618] - res_pjsip: Check for readability of TLS files at startup

New Feature

[ASTERISK-24922] - ARI: Add the ability to intercept hold and raise an event
[ASTERISK-25419] - Dialplan Application for Integration of StatsD
[ASTERISK-25549] - Confbridge: Add participant timeout option

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc1

Rilasciato Asterisk 11.21.0-rc1

Il giorno 15 dicembre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.21.0-rc1.

Dal post originale:

Bug

[ASTERISK-7803] - [patch] Update the maximum packetization values in frame.c
[ASTERISK-24146] - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
[ASTERISK-25135] - [patch]RTP Timeout hangup cause code missing
[ASTERISK-25364] - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
[ASTERISK-25373] - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
[ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
[ASTERISK-25400] - Hints broken when "CustomPresence" doesn't exist in AstDB
[ASTERISK-25434] - Compiler flags not reported in 'core show settings' despite usage during compilation
[ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
[ASTERISK-25443] - [patch]IPv6 - Potential issue in via header parsing
[ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
[ASTERISK-25455] - Deadlock of PJSIP realtime over res_config_pgsql
[ASTERISK-25461] - Nested dialplan #includes don't work as expected.
[ASTERISK-25476] - chan_sip loses registrations after a while
[ASTERISK-25494] - build: GCC 5.1.x catches some new const, array bounds and missing paren issues
[ASTERISK-25498] - Asterisk crashes when negotiating g729 without that module installed
[ASTERISK-25527] - Quirky xmldoc description wrapping
[ASTERISK-25537] - [patch] format-attribute module: RFC or internal defaults?
[ASTERISK-25552] - hashtab: Improve NULL tolerance
[ASTERISK-25569] - app_meetme: Audio quality issues
[ASTERISK-25585] - [patch]rasterisk never hits most of main(), but it's assumed to
[ASTERISK-25593] - fastagi: record file closed after sending result
[ASTERISK-25599] - [patch] SLIN Resampling Codec only 80 msec
[ASTERISK-25609] - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
[ASTERISK-25610] - Asterisk crash during "sip reload"
[ASTERISK-25616] - Warning with a Codec Module which supports PLC with FEC

Improvement

[ASTERISK-24718] - [patch]Add inital support of "sanitize" to configure

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0-rc1

Bridges, T.38, and other good times

Matt Jordan, il 06 dicembre 2915, ha pubblicato un "lunghissimo" post con una accurata "discussione" circa il (non)funzionamento di fax/pjsip/directmedia_reinvite_t38

Questo il link del post:
https://www.mail-archive.com/asterisk-dev@lists.digium.com/msg42520.html

Feedback sui ​​miglioramenti ODBC in Asterisk by Mark Michelson

Mark Michelson, il 30 novembre 2915, ha pubblicato una richiesta di un feedback sui ​​miglioramenti ODBC in Asterisk ed ha affrontato interessanti tematiche relative all'utilizzo del pjsip rispetto alle commessioni tramite ODBX.

Questo il link del post:
https://www.mail-archive.com/asterisk-dev@lists.digium.com/msg42512.html