[asterisk-dev] Asterisk 14.2.0-rc1 Now Available
The Asterisk Development Team has announced the first release candidate of Asterisk 14.2.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.2.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release candidate: Improvements made in this release: --- * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton) Bugs fixed in this release: --- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett) * ASTERISK-26556 - manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes (Reported by Michelle Dupuis) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph) * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason) * ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp) * ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit IPv6 transport configured (Reported by Joshua Colp) * ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli) * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov) * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan) * ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used (Reported by Frankie Chin) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c (Reported by Matt Krokosz) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard) * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav) * ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell) * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour) * ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters) * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp) * ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton) * ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains
[asterisk-dev] Asterisk 13.13.0-rc1 Now Available
The Asterisk Development Team has announced the first release candidate of Asterisk 13.13.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.13.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release candidate: New Features made in this release: --- * ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan) * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan) Bugs fixed in this release: --- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett) * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph) * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason) * ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp) * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov) * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan) * ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard) * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav) * ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell) * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour) * ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters) * ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton) * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp) * ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel) * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud) * ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy) * ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston) * ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not
[asterisk-dev] Asterisk 11.25.0-rc1 Now Available
The Asterisk Development Team has announced the first release candidate of Asterisk 11.25.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.25.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release candidate: Bugs fixed in this release: --- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0-rc1 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
Re: [asterisk-dev] sorcery_memory_cache_delete: Unable to delete object
On Fri, Nov 18, 2016, at 07:54 AM, Ross Beer wrote: > Hi, > > > Should asterisk output the following message if the object isn't in the > cache? > > > [2016-11-18 11:50:25] ERROR[73437]: res_sorcery_memory_cache.c:1558 > sorcery_memory_cache_delete: Unable to delete object '<>' from > sorcery cache > > > In my opinion, this isn't needed and just causes the logs to fill up. > > > If this is actually an error how can this be fixed? The fact that it can't delete the object won't hurt stuff, but it would be nice to know precisely the configuration and scenario that causes it to want to delete something that doesn't exist. An issue would need to be created with the configuration and logs with core debug of level 2. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
[asterisk-dev] sorcery_memory_cache_delete: Unable to delete object
Hi, Should asterisk output the following message if the object isn't in the cache? [2016-11-18 11:50:25] ERROR[73437]: res_sorcery_memory_cache.c:1558 sorcery_memory_cache_delete: Unable to delete object '<>' from sorcery cache In my opinion, this isn't needed and just causes the logs to fill up. If this is actually an error how can this be fixed? Kind regards, Ross -- _ -- 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
