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Door , , 1 reactie
Bron: Asterisk

Asterisk is een uitgebreide pbx voor BSD, Linux en Mac OS X. Het programma biedt alle functies die je van een telefooncentrale mag verwachten. Zo beschikt het onder andere over mogelijkheden voor voicemail, conferencing en call queueing. Daarnaast is ondersteuning voor caller-id-services, adsi, sip en h323 aanwezig. Voor een compleet overzicht van alle mogelijkheden verwijzen we naar deze pagina. De ontwikkelaars hebben versies 14.2.0, 13.13.0, en 11.25.0 uitgebracht, voorzien van de volgende aankondigingen:

Asterisk 14.2.0 Now Available

Improvements made in this release:
  • ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue
  • ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
  • ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample
  • ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
  • ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
Bugs fixed in this release:
  • ASTERISK-26608 - Compile and link failures on OpenBSD
  • ASTERISK-26520 - codec_opus: Generated fmtp line has no content
  • ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
  • ASTERISK-26516 - pjsip: Memory corruption with possible memory leak.
  • ASTERISK-26556 - manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes
  • ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi
  • ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
  • ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
  • ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
  • ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit IPv6 transport configuredASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls
  • ASTERISK-24400 - ooh323 sends wrong hangup code
  • ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions
  • ASTERISK-26412 - build: Prepare for gcc 6.2
  • ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10
  • ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
  • ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state
  • ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
  • ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types.
  • ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
  • ASTERISK-25070 - Fix FTBFS on Hurd
  • ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
  • ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum
  • ASTERISK-26537 - AMI: NewConnectedLine event is not documented
  • ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
  • ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
  • ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash
  • ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls.
  • ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
  • ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf
  • ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
  • ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak
  • ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
  • ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
  • ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
  • ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
  • ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session
  • ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients
  • ASTERISK-26444 - 'features show' command in CLI does not return prompt.
  • ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
  • ASTERISK-26356 - menuselect: invalid test for GTK2
  • ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
  • ASTERISK-26439 - chan_rtp: Crash when originating
  • ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered.
  • ASTERISK-26618 - build: Backport addition of librt check to configure.ac
New Features made in this release:
  • ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge
  • ASTERISK-26492 - ARI: Add ability to specify channel variables on websocket events
  • ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0

Asterisk 13.13.0 Now Available

New Features made in this release:
  • ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge
  • ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events
Bugs fixed in this release:
  • ASTERISK-26608 - Compile and link failures on OpenBSD
  • ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi
  • ASTERISK-26520 - codec_opus: Generated fmtp line has no content
  • ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
  • ASTERISK-26516 - pjsip: Memory corruption with possible memory leak.
  • ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
  • ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
  • ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
  • ASTERISK-24400 - ooh323 sends wrong hangup code
  • ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions
  • ASTERISK-26412 - build: Prepare for gcc 6.2
  • ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10
  • ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
  • ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls
  • ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types.
  • ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state
  • ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum
  • ASTERISK-25070 - Fix FTBFS on Hurd
  • ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
  • ASTERISK-26537 - AMI: NewConnectedLine event is not documented
  • ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
  • ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
  • ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash
  • ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls.
  • ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf
  • ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
  • ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
  • ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak
  • ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
  • ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
  • ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
  • ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
  • ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients
  • ASTERISK-26444 - 'features show' command in CLI does not return prompt.
  • ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session
  • ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
  • ASTERISK-26356 - menuselect: invalid test for GTK2
  • ASTERISK-26439 - chan_rtp: Crash when originating
  • ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
  • ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered.
  • ASTERISK-26618 - build: Backport addition of librt check to configure.ac
Improvements made in this release:
  • ASTERISK-25063 - [patch]add X.509 subject alternative name support to Asterisk TLS support
  • ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue
  • ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
  • ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample
  • ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
  • ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0

Asterisk 11.25.0 Now Available

Bugs fixed in this release:
  • ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
  • ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
  • ASTERISK-26356 - menuselect: invalid test for GTK2
  • ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
  • ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
Thank you for your continued support of Asterisk!
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Reacties (1)

Asterisk heeft sinds kort een gebundelde pjproject, inclusief speciale patches voor Asterisk. Die dien je wl aan te roepen in de configure regel. Op die manier krijg je maximale functionaliteit en performance. Zo doe ik het:

./configure \
CFLAGS="-DPJ_HAS_IPV6=1 -fPIC -O3 -march=native -flto" \
CXXFLAGS="-fPIC -O3 -march=native -flto" \
--with-pjproject-bundled

Mijn volledig quick & dirty bash scriptje staat hier, voor wie ik er een plezier mee kan doen: http://pastebin.com/qS0CnbFN


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