From asteriskteam at digium.com Thu Dec 1 16:01:27 2022 From: asteriskteam at digium.com (Asterisk Development Team) Date: Thu, 1 Dec 2022 16:01:27 -0600 Subject: [asterisk-dev] Asterisk 16.29.1, 18.15.1, 19.7.1, 20.0.1 Now Available Message-ID: The Asterisk Development Team would like to announce the release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues reported by the community and would have not been possible without your participation.Thank you! The following issue is resolved in this release: Bugs fixed in this release: ———————————– [ASTERISK-30103 ] chan_ooh323 vulnerability in calling/called party IE (Reported By: Michael Bradeen) [ASTERISK-30176 ] GetConfig can read files outside of Asterisk (Reported By: shawty) [ASTERISK-30244 ] Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported By: nappsoft) [ASTERISK-30338 ] Backport 2.13 security fixes from pjproject For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1 https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1 https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1 https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment was scrubbed... URL: From asteriskteam at digium.com Thu Dec 15 07:38:22 2022 From: asteriskteam at digium.com (Asterisk Development Team) Date: Thu, 15 Dec 2022 13:38:22 +0000 Subject: [asterisk-dev] Asterisk 16.30.0-rc1 Now Available Message-ID: The Asterisk Development Team would like to announce the first release candidate of Asterisk 16.30.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.30.0-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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-30338 - pjproject: Backport security fixes from 2.13 (Reported by Benjamin Keith Ford) * ASTERISK-30176 - manager: GetConfig can read files outside of Asterisk (Reported by shawty) * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called party IE (Reported by Michael Bradeen) Improvements made in this release: ----------------------------------- * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level (Reported by N A) * ASTERISK-30223 - features: add no-answer option to Bridge application (Reported by N A) * ASTERISK-30158 - PJSIP: Add new 100rel option "peer_supported" (Reported by Maximilian Fridrich) Bugs fixed in this release: ----------------------------------- * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported by nappsoft) * ASTERISK-28689 - res_pjsip: Crash when locking group lock when sending stateful response (Reported by Jesse Ross) * ASTERISK-30217 - Registration do not allow multiple proxies (Reported by Igor Goncharovsky) * ASTERISK-30193 - chan_pjsip should return all codecs on a re-INVITE without SDP (Reported by Henning Westerholt) * ASTERISK-30213 - Make crypto_load() reentrant and handle symlinks correctly (Reported by Philip Prindeville) * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings (Reported by N A) * ASTERISK-30245 - db: ListItems is incorrect (Reported by N A) * ASTERISK-30243 - func_logic: IF function complains if both branches are empty (Reported by N A) * ASTERISK-30232 - Initialize stack-based ast_test_capture structures correctly (Reported by Philip Prindeville) * ASTERISK-30220 - func_scramble: Fix segfault due to null pointer deref (Reported by N A) * ASTERISK-30235 - res_crypto and tests: Memory issues and and uninitialized variable error (Reported by George Joseph) * ASTERISK-30234 - res_geolocation: ...may be used uninitialized error in geoloc_config.c (Reported by George Joseph) * ASTERISK-30226 - REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys (Reported by George Joseph) New Features made in this release: ----------------------------------- * ASTERISK-30091 - cdr: Allow CDRs to ignore call state changes (Reported by N A) * ASTERISK-30254 - res_tonedetect: Add audible ringback detection to TONE_DETECT (Reported by N A) * ASTERISK-30032 - Support of mediasec SIP headers and SDP attributes (Reported by Maximilian Fridrich) * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to not answer (Reported by N A) * ASTERISK-30179 - app_amd: Allow audio to be played while AMD is running (Reported by N A) * ASTERISK-29432 - New function to allow access to any channel (Reported by N A) * ASTERISK-30222 - func_strings: Add trim functions (Reported by N A) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0-rc1 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment was scrubbed... URL: From asteriskteam at digium.com Thu Dec 15 07:39:57 2022 From: asteriskteam at digium.com (Asterisk Development Team) Date: Thu, 15 Dec 2022 13:39:57 +0000 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available Message-ID: The Asterisk Development Team would like to announce the first release candidate of Asterisk 18.16.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.16.0-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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-30338 - pjproject: Backport security fixes from 2.13 (Reported by Benjamin Keith Ford) * ASTERISK-30176 - manager: GetConfig can read files outside of Asterisk (Reported by shawty) * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called party IE (Reported by Michael Bradeen) Improvements made in this release: ----------------------------------- * ASTERISK-30328 - Typo in from_domain description on res_pjsip configuration documentation (Reported by Marcel Wagner) * ASTERISK-30316 - res_pjsip: Documentation should point out default if contact_user is not being set for outbound registrations (Reported by Marcel Wagner) * ASTERISK-30289 - xmldoc: Allow XML docs to be reloaded (Reported by N A) * ASTERISK-30327 - rtp_engine.h: Remove obsolete example usage (Reported by N A) * ASTERISK-30286 - app_mixmonitor: Add option to use real Caller ID for Caller ID (Reported by N A) * ASTERISK-30308 - pbx_builtins: Allow Answer to return immediately (Reported by N A) * ASTERISK-30295 - test_json: Remove duplicated static function (Reported by N A) * ASTERISK-30290 - file.c: Don't emit warnings on winks. (Reported by N A) * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level (Reported by N A) * ASTERISK-30223 - features: add no-answer option to Bridge application (Reported by N A) * ASTERISK-30158 - PJSIP: Add new 100rel option "peer_supported" (Reported by Maximilian Fridrich) Bugs fixed in this release: ----------------------------------- * ASTERISK-30349 - app_if: Format truncation error (Reported by George Joseph) * ASTERISK-30265 - res_pjsip_session: Fix missing PLAR support on INVITEs (Reported by N A) * ASTERISK-30283 - app_voicemail: Fix msg_create_from_file not sending email to user (Reported by N A) * ASTERISK-29793 - adsi: CAS is malformed (Reported by N A) * ASTERISK-30344 - ari: Memory leak in create when specifying JSON (Reported by Saken) * ASTERISK-30311 - func_presencestate: Fix invalid memory access. (Reported by N A) * ASTERISK-30336 - sig_analog: Fix no timeout duration (Reported by N A) * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported by nappsoft) * ASTERISK-30184 - res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg (Reported by Maximilian Fridrich) * ASTERISK-29998 - sla: deadlock when calling SLAStation application (Reported by N A) * ASTERISK-30321 - Build: Embedded blobs have executable stacks (Reported by George Joseph) * ASTERISK-30293 - Memory leak in JSON_DECODE (Reported by David Uczen) * ASTERISK-30314 - res_agi: RECORD FILE doesn't respect "transmit_silence" asterisk.conf option (Reported by Joshua C. Colp) * ASTERISK-30285 - manager.c: Remove outdated documentation (Reported by N A) * ASTERISK-30076 - app_stack: Incorrect exit location in predial handlers logged (Reported by N A) * ASTERISK-30282 - CI: Coredump output isn't saved when running unittests (Reported by George Joseph) * ASTERISK-30281 - chan_rtp: Local address being used before being set (Reported by George Joseph) * ASTERISK-28689 - res_pjsip: Crash when locking group lock when sending stateful response (Reported by Jesse Ross) * ASTERISK-30217 - Registration do not allow multiple proxies (Reported by Igor Goncharovsky) * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled (Reported by N A) * ASTERISK-30273 - test_mwi: compilation fails on 32-bit Debian (Reported by N A) * ASTERISK-30193 - chan_pjsip should return all codecs on a re-INVITE without SDP (Reported by Henning Westerholt) * ASTERISK-30258 - Dialing API: Cancel a running async thread, does not always cancel all calls (Reported by Frederic LE FOLL) * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY (Reported by N A) * ASTERISK-30248 - ast_get_digit_str adds bogus initial delimiter if first character not to be spoken (Reported by David Woolley) * ASTERISK-30264 - res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash (Reported by N A) * ASTERISK-30213 - Make crypto_load() reentrant and handle symlinks correctly (Reported by Philip Prindeville) * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings (Reported by N A) * ASTERISK-30239 - Prometheus plugin crashes Asterisk when using local channel (Reported by Joeran Vinzens) * ASTERISK-30237 - res_prometheus: Crash when scraping bridges (Reported by Igor Yeroshev) * ASTERISK-30245 - db: ListItems is incorrect (Reported by N A) * ASTERISK-30243 - func_logic: IF function complains if both branches are empty (Reported by N A) * ASTERISK-30232 - Initialize stack-based ast_test_capture structures correctly (Reported by Philip Prindeville) * ASTERISK-30220 - func_scramble: Fix segfault due to null pointer deref (Reported by N A) * ASTERISK-30235 - res_crypto and tests: Memory issues and and uninitialized variable error (Reported by George Joseph) * ASTERISK-30234 - res_geolocation: ...may be used uninitialized error in geoloc_config.c (Reported by George Joseph) * ASTERISK-30226 - REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys (Reported by George Joseph) New Features made in this release: ----------------------------------- * ASTERISK-21502 - New SIP Channel Driver - add Advice of Charge support (Reported by Matt Jordan) * ASTERISK-30150 - res_pjsip_session: Add support for custom parameters (Reported by N A) * ASTERISK-30322 - res_hep: Add capture agent name support (Reported by N A) * ASTERISK-29497 - Add conditional branch applications (Reported by N A) * ASTERISK-30305 - chan_dahdi: Allow FXO channels to start immediately (Reported by N A) * ASTERISK-30284 - app_mixmonitor: Add option to delete recording file when done (Reported by N A) * ASTERISK-30146 - res_pjsip_logger: Add method-based log filtering (Reported by N A) * ASTERISK-30263 - res_pjsip_notify: Allow using pjsip_notify.conf from AMI (Reported by N A) * ASTERISK-30254 - res_tonedetect: Add audible ringback detection to TONE_DETECT (Reported by N A) * ASTERISK-30091 - cdr: Allow CDRs to ignore call state changes (Reported by N A) * ASTERISK-30032 - Support of mediasec SIP headers and SDP attributes (Reported by Maximilian Fridrich) * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to not answer (Reported by N A) * ASTERISK-30179 - app_amd: Allow audio to be played while AMD is running (Reported by N A) * ASTERISK-29432 - New function to allow access to any channel (Reported by N A) * ASTERISK-30222 - func_strings: Add trim functions (Reported by N A) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0-rc1 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment was scrubbed... URL: From asteriskteam at digium.com Thu Dec 15 07:41:35 2022 From: asteriskteam at digium.com (Asterisk Development Team) Date: Thu, 15 Dec 2022 13:41:35 +0000 Subject: [asterisk-dev] Asterisk 19.8.0-rc1 Now Available Message-ID: The Asterisk Development Team would like to announce the first release candidate of Asterisk 19.8.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 19.8.0-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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-30338 - pjproject: Backport security fixes from 2.13 (Reported by Benjamin Keith Ford) * ASTERISK-30176 - manager: GetConfig can read files outside of Asterisk (Reported by shawty) * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called party IE (Reported by Michael Bradeen) Improvements made in this release: ----------------------------------- * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level (Reported by N A) * ASTERISK-30223 - features: add no-answer option to Bridge application (Reported by N A) * ASTERISK-30158 - PJSIP: Add new 100rel option "peer_supported" (Reported by Maximilian Fridrich) Bugs fixed in this release: ----------------------------------- * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported by nappsoft) * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled (Reported by N A) * ASTERISK-28689 - res_pjsip: Crash when locking group lock when sending stateful response (Reported by Jesse Ross) * ASTERISK-30217 - Registration do not allow multiple proxies (Reported by Igor Goncharovsky) * ASTERISK-30273 - test_mwi: compilation fails on 32-bit Debian (Reported by N A) * ASTERISK-30193 - chan_pjsip should return all codecs on a re-INVITE without SDP (Reported by Henning Westerholt) * ASTERISK-30258 - Dialing API: Cancel a running async thread, does not always cancel all calls (Reported by Frederic LE FOLL) * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY (Reported by N A) * ASTERISK-30264 - res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash (Reported by N A) * ASTERISK-30248 - ast_get_digit_str adds bogus initial delimiter if first character not to be spoken (Reported by David Woolley) * ASTERISK-30213 - Make crypto_load() reentrant and handle symlinks correctly (Reported by Philip Prindeville) * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings (Reported by N A) * ASTERISK-30239 - Prometheus plugin crashes Asterisk when using local channel (Reported by Joeran Vinzens) * ASTERISK-30237 - res_prometheus: Crash when scraping bridges (Reported by Igor Yeroshev) * ASTERISK-30245 - db: ListItems is incorrect (Reported by N A) * ASTERISK-30243 - func_logic: IF function complains if both branches are empty (Reported by N A) * ASTERISK-30232 - Initialize stack-based ast_test_capture structures correctly (Reported by Philip Prindeville) * ASTERISK-30220 - func_scramble: Fix segfault due to null pointer deref (Reported by N A) * ASTERISK-30235 - res_crypto and tests: Memory issues and and uninitialized variable error (Reported by George Joseph) * ASTERISK-30234 - res_geolocation: ...may be used uninitialized error in geoloc_config.c (Reported by George Joseph) * ASTERISK-30226 - REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys (Reported by George Joseph) New Features made in this release: ----------------------------------- * ASTERISK-30146 - res_pjsip_logger: Add method-based log filtering (Reported by N A) * ASTERISK-30263 - res_pjsip_notify: Allow using pjsip_notify.conf from AMI (Reported by N A) * ASTERISK-30091 - cdr: Allow CDRs to ignore call state changes (Reported by N A) * ASTERISK-30254 - res_tonedetect: Add audible ringback detection to TONE_DETECT (Reported by N A) * ASTERISK-30032 - Support of mediasec SIP headers and SDP attributes (Reported by Maximilian Fridrich) * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to not answer (Reported by N A) * ASTERISK-30179 - app_amd: Allow audio to be played while AMD is running (Reported by N A) * ASTERISK-29432 - New function to allow access to any channel (Reported by N A) * ASTERISK-30222 - func_strings: Add trim functions (Reported by N A) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0-rc1 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment was scrubbed... URL: From asteriskteam at digium.com Thu Dec 15 07:44:50 2022 From: asteriskteam at digium.com (Asterisk Development Team) Date: Thu, 15 Dec 2022 13:44:50 +0000 Subject: [asterisk-dev] Asterisk 20.1.0-rc1 Now Available Message-ID: The Asterisk Development Team would like to announce the first release candidate of Asterisk 20.1.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 20.1.0-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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-30338 - pjproject: Backport security fixes from 2.13 (Reported by Benjamin Keith Ford) * ASTERISK-30176 - manager: GetConfig can read files outside of Asterisk (Reported by shawty) * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called party IE (Reported by Michael Bradeen) Improvements made in this release: ----------------------------------- * ASTERISK-30328 - Typo in from_domain description on res_pjsip configuration documentation (Reported by Marcel Wagner) * ASTERISK-30316 - res_pjsip: Documentation should point out default if contact_user is not being set for outbound registrations (Reported by Marcel Wagner) * ASTERISK-30289 - xmldoc: Allow XML docs to be reloaded (Reported by N A) * ASTERISK-30327 - rtp_engine.h: Remove obsolete example usage (Reported by N A) * ASTERISK-30286 - app_mixmonitor: Add option to use real Caller ID for Caller ID (Reported by N A) * ASTERISK-30308 - pbx_builtins: Allow Answer to return immediately (Reported by N A) * ASTERISK-30295 - test_json: Remove duplicated static function (Reported by N A) * ASTERISK-30290 - file.c: Don't emit warnings on winks. (Reported by N A) * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level (Reported by N A) * ASTERISK-30223 - features: add no-answer option to Bridge application (Reported by N A) * ASTERISK-30158 - PJSIP: Add new 100rel option "peer_supported" (Reported by Maximilian Fridrich) Bugs fixed in this release: ----------------------------------- * ASTERISK-30349 - app_if: Format truncation error (Reported by George Joseph) * ASTERISK-30344 - ari: Memory leak in create when specifying JSON (Reported by Saken) * ASTERISK-30283 - app_voicemail: Fix msg_create_from_file not sending email to user (Reported by N A) * ASTERISK-30265 - res_pjsip_session: Fix missing PLAR support on INVITEs (Reported by N A) * ASTERISK-29793 - adsi: CAS is malformed (Reported by N A) * ASTERISK-30311 - func_presencestate: Fix invalid memory access. (Reported by N A) * ASTERISK-30336 - sig_analog: Fix no timeout duration (Reported by N A) * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported by nappsoft) * ASTERISK-30184 - res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg (Reported by Maximilian Fridrich) * ASTERISK-29998 - sla: deadlock when calling SLAStation application (Reported by N A) * ASTERISK-30321 - Build: Embedded blobs have executable stacks (Reported by George Joseph) * ASTERISK-30293 - Memory leak in JSON_DECODE (Reported by David Uczen) * ASTERISK-30314 - res_agi: RECORD FILE doesn't respect "transmit_silence" asterisk.conf option (Reported by Joshua C. Colp) * ASTERISK-30285 - manager.c: Remove outdated documentation (Reported by N A) * ASTERISK-30282 - CI: Coredump output isn't saved when running unittests (Reported by George Joseph) * ASTERISK-30076 - app_stack: Incorrect exit location in predial handlers logged (Reported by N A) * ASTERISK-30281 - chan_rtp: Local address being used before being set (Reported by George Joseph) * ASTERISK-28689 - res_pjsip: Crash when locking group lock when sending stateful response (Reported by Jesse Ross) * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled (Reported by N A) * ASTERISK-30217 - Registration do not allow multiple proxies (Reported by Igor Goncharovsky) * ASTERISK-30273 - test_mwi: compilation fails on 32-bit Debian (Reported by N A) * ASTERISK-30193 - chan_pjsip should return all codecs on a re-INVITE without SDP (Reported by Henning Westerholt) * ASTERISK-30258 - Dialing API: Cancel a running async thread, does not always cancel all calls (Reported by Frederic LE FOLL) * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY (Reported by N A) * ASTERISK-30264 - res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash (Reported by N A) * ASTERISK-30248 - ast_get_digit_str adds bogus initial delimiter if first character not to be spoken (Reported by David Woolley) * ASTERISK-30213 - Make crypto_load() reentrant and handle symlinks correctly (Reported by Philip Prindeville) * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings (Reported by N A) * ASTERISK-30239 - Prometheus plugin crashes Asterisk when using local channel (Reported by Joeran Vinzens) * ASTERISK-30237 - res_prometheus: Crash when scraping bridges (Reported by Igor Yeroshev) * ASTERISK-30245 - db: ListItems is incorrect (Reported by N A) * ASTERISK-30243 - func_logic: IF function complains if both branches are empty (Reported by N A) * ASTERISK-30232 - Initialize stack-based ast_test_capture structures correctly (Reported by Philip Prindeville) * ASTERISK-30220 - func_scramble: Fix segfault due to null pointer deref (Reported by N A) * ASTERISK-30235 - res_crypto and tests: Memory issues and and uninitialized variable error (Reported by George Joseph) * ASTERISK-30234 - res_geolocation: ...may be used uninitialized error in geoloc_config.c (Reported by George Joseph) * ASTERISK-30226 - REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys (Reported by George Joseph) New Features made in this release: ----------------------------------- * ASTERISK-21502 - New SIP Channel Driver - add Advice of Charge support (Reported by Matt Jordan) * ASTERISK-30322 - res_hep: Add capture agent name support (Reported by N A) * ASTERISK-29497 - Add conditional branch applications (Reported by N A) * ASTERISK-30150 - res_pjsip_session: Add support for custom parameters (Reported by N A) * ASTERISK-30305 - chan_dahdi: Allow FXO channels to start immediately (Reported by N A) * ASTERISK-30284 - app_mixmonitor: Add option to delete recording file when done (Reported by N A) * ASTERISK-30263 - res_pjsip_notify: Allow using pjsip_notify.conf from AMI (Reported by N A) * ASTERISK-30146 - res_pjsip_logger: Add method-based log filtering (Reported by N A) * ASTERISK-30091 - cdr: Allow CDRs to ignore call state changes (Reported by N A) * ASTERISK-30254 - res_tonedetect: Add audible ringback detection to TONE_DETECT (Reported by N A) * ASTERISK-30032 - Support of mediasec SIP headers and SDP attributes (Reported by Maximilian Fridrich) * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to not answer (Reported by N A) * ASTERISK-30179 - app_amd: Allow audio to be played while AMD is running (Reported by N A) * ASTERISK-29432 - New function to allow access to any channel (Reported by N A) * ASTERISK-30222 - func_strings: Add trim functions (Reported by N A) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0-rc1 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment was scrubbed... URL: From m1278468 at mailbox.org Sat Dec 17 11:21:24 2022 From: m1278468 at mailbox.org (Michael Maier) Date: Sat, 17 Dec 2022 18:21:24 +0100 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: Message-ID: On 15.12.22 at 14:39 Asterisk Development Team wrote: > # [ASTERISK-30032 ] - > Support of mediasec SIP headers and SDP attributes > (Reported by Maximilian Fridrich) Hi Maximilian, thanks for adding Mediasec support to asterisk! Unfortunately somewhat late - at least regarding Deutsche Telekom consumer infrastructure. Telekom started quite some time ago dropping the Mediasec requirement for their SIP servers. The new SIP servers like [city]-l01-mav-pc-rt-001.edns.t-ipnet.de don't need any Mediasec headers any more. Those servers are meanwhile often the primary ones provided by the SRV lookup (if they already exist for the location you are requesting for). But the old ones, which are still provided currently with lower priority, still need it. Therefore it is necessary to enable Mediasec nevertheless to be able to change to them if necessary. Does your patch work, too, if a server doesn't answer the Mediasec request? Another question: Do you maybe plan to get asterisk ready to cope with 3 completely independent SIP servers provided by the SRV lookup? Currently, asterisk can't handle it (it must be ensured that all subsequent requests after the register must use the same server previously registered to and not any other server of the list. Before switching to another server, it's necessary to unregister the current active server and afterwards register to the new one). Thanks Michael From m1278468 at mailbox.org Sun Dec 18 01:46:03 2022 From: m1278468 at mailbox.org (Michael Maier) Date: Sun, 18 Dec 2022 08:46:03 +0100 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: Message-ID: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> On 17.12.22 at 18:21 Michael Maier wrote: > On 15.12.22 at 14:39 Asterisk Development Team wrote: > >> # [ASTERISK-30032 ] - >>         Support of mediasec SIP headers and SDP attributes >> (Reported by Maximilian Fridrich) Hi Maximilian, today I tried your implementation with Deutsche Telekom (MagentaZuhause). My configuration: security_negotiation=mediasec security_mechanisms=sdes-srtp(,dtls-srtp,msrp-tls) for endpoint and registration. I wasn't able to get it working. The headers you are setting unfortunately doesn't meet the Deutsche Telekom requirements - besides one additional bug. If you compare it to my patch, you most probably will see the differences / the bug. Please be careful as Deutsche Telekom not always responds with 494 during registration and even authorization during registration isn't always necessary (only sometimes)! You should have a close look over ~1 day and each reRegistration to be sure to get each possible procedure! Try to force reRegistrations with pjsip send register ... and carefully check the headers. Does it still work 1 hour later (in- and outbound calls)? The registration implementation is not necessarily ok if the registration is fine (no error is thrown)! You must try to place an outgoing call e.g. - this call must work! At the moment, the registration proceeds, but outgoing calls are rejected (403 Forbidden). The cause is, that the registration wasn't correct and therefore srtp isn't possible - but Deutsche Telekom requires / enforces srtp if tls is active! Here you can see how it should look like: https://www.ip-phone-forum.de/threads/telekom-all-ip-privat-endlich-erfolgreiche-registierung-ssl-tls-m%C3%B6glich.300166/page-3 (Meester Proper is a very good skilled employee of Deutsche Telekom) Btw: - Are Options packages working? Did you test this, too? - I can provide my mediasec patch for 18.16.0-rc1 if you want to have it. Thanks Michael From M.Fridrich at commend.com Tue Dec 20 04:29:12 2022 From: M.Fridrich at commend.com (Fridrich Maximilian) Date: Tue, 20 Dec 2022 10:29:12 +0000 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> Message-ID: Mr. Maier, thank you very much for your feedback! We provided this specifically for Telekom's "CompanyFlex" trunks which still require mediasec headers according to their website [1]. Specifically, we have to adhere to their technical specification 1TR119 [2]. > Does your patch work, too, if a server doesn't answer the Mediasec > request? We set the Require: mediasec header, so if a server does not understand this, it MUST respond with 420 Bad Extension. Nonetheless, if you have configured mediasec a server could ignore the mediasec headers and still send 2XX replies to our requests. Since the mediasec headers are static and no real security mechanism is negotiated anyways (all we need to do is satisfy Telekom's requirements), we still allow further transactions to take place (which is not how RFC 3329 intends it). However, it does affect the SDP by setting the 3ge2ae attribute, even if the server never sent us Security-Server headers. > Do you maybe plan to get asterisk ready to cope with 3 completely > independent SIP servers provided by the SRV lookup? Unfortunately, I will probably not have time to look into that in the near future. As stated above, we provided this patch to work with SIP trunks (e.g. CompanyFlex) that explicitly require mediasec. > I wasn't able to get it working. The headers you are setting > unfortunately doesn't meet the Deutsche Telekom requirements - besides > one additional bug. Thank you for testing it! We have identified similar issues (see ASTERISK-30276) and I just uploaded a patch fixing those [3]. I believe this patch fixes the issues you are seeing. In our setup, it seems to be working fine - including outgoing calls, re-registrations, and OPTIONS. Please let me know, if you are still experiencing issues with the new patch. Thanks, Max [1] https://hilfe.companyflex.de/de/grundlagen/systemvoraussetzungen [2] https://www.telekom.de/hilfe/downloads/1tr119.pdf [3] https://gerrit.asterisk.org/c/asterisk/+/19740 From m1278468 at mailbox.org Tue Dec 20 10:10:35 2022 From: m1278468 at mailbox.org (Michael Maier) Date: Tue, 20 Dec 2022 17:10:35 +0100 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> Message-ID: Hello Max, On 20.12.22 at 11:29 Fridrich Maximilian wrote: > Mr. Maier, Michael :-) > > thank you very much for your feedback! We provided this specifically > for Telekom's "CompanyFlex" trunks which still require mediasec headers > according to their website [1]. Specifically, we have to adhere to > their technical specification 1TR119 [2]. Well, for Telekom MagentaZuhause, the headers must look like this to work (there seems to be a difference to the CompanyFlex servers): Security-Client: sdes-srtp;mediasec ^^^^^^^^^ The ",mediasec" is missing. Further more, if you configure (id est: a list) security_mechanisms=sdes-srtp,... always the first entry of the list configured above is dropped in the following register request. Is this fixed by your mentioned patch below? Example: Request Response 401 Request (now without first entry of the list) Security-Client: ... > >> Does your patch work, too, if a server doesn't answer the Mediasec >> request? > > We set the Require: mediasec header, so if a server does not understand > this, it MUST respond with 420 Bad Extension. The new consumer VoIP server just ignores it ... > Nonetheless, if you have > configured mediasec a server could ignore the mediasec headers and still > send 2XX replies to our requests. Since the mediasec headers are static > and no real security mechanism is negotiated anyways (all we need to do > is satisfy Telekom's requirements), we still allow further transactions > to take place (which is not how RFC 3329 intends it). Same as I do :-) > However, it does > affect the SDP by setting the 3ge2ae attribute, even if the server never > sent us Security-Server headers. [...] >> I wasn't able to get it working. The headers you are setting >> unfortunately doesn't meet the Deutsche Telekom requirements - besides >> one additional bug. > > Thank you for testing it! We have identified similar issues (see > ASTERISK-30276) and I just uploaded a patch fixing those [3]. I believe > this patch fixes the issues you are seeing. In our setup, it seems to > be working fine - including outgoing calls, re-registrations, and > OPTIONS. > > Please let me know, if you are still experiencing issues with the new > patch. I think the different headers are not addressed? Thanks Michael From m1278468 at mailbox.org Tue Dec 20 10:58:10 2022 From: m1278468 at mailbox.org (Michael Maier) Date: Tue, 20 Dec 2022 17:58:10 +0100 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> Message-ID: <402c0462-5682-2e4b-0127-70278a131e4e@mailbox.org> On 20.12.22 at 17:10 Michael Maier wrote: > Hello Max, > > On 20.12.22 at 11:29 Fridrich Maximilian wrote: >> Please let me know, if you are still experiencing issues with the new >> patch. > > I think the different headers are not addressed? Just tested it - doesn't work. It is fine for me if you don't want to address the requirements of VoIP MagentaZuhause. But it would be good, I think, if you would add the information to the documents page, that this implementation is not a generic mediasec implementation, but a specific Telekom CompanyFlex one :-), which doesn't work with MagentaZuhause (or some others, too). If I remember correctly, somebody told me my patch would have been working, too, with CompanyFlex - but I'm not sure any more. This could be wrong, too. It has been quite some time ago .... > Thanks > Michael > From M.Fridrich at commend.com Wed Dec 21 01:21:19 2022 From: M.Fridrich at commend.com (Fridrich Maximilian) Date: Wed, 21 Dec 2022 07:21:19 +0000 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: <402c0462-5682-2e4b-0127-70278a131e4e@mailbox.org> References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> <402c0462-5682-2e4b-0127-70278a131e4e@mailbox.org> Message-ID: > Security-Client: sdes-srtp;mediasec > ^^^^^^^^^ > The ",mediasec" is missing. Yes, the security_mechanisms option is a comma separated list of the literal security_mechanisms that should be used. I.e. you have to specify security_mechanisms=sdes-srtp\;mediasec,dtls-srtp\;mediasec (don't forget to escape the semicolon). > always the first entry of the list configured above is dropped in > the following register request. Is this fixed by your mentioned patch > below? I could not reproduce this behavior. I just tested it with the current patch and no list entries were dropped. > I think the different headers are not addressed? Do you mean the missing ";mediasec" values? That is due to the configuration, as stated above. Besides that, I'm quite confident it behaves as intended, we have been running test systems for quite a while now. I hope we can resolve your issues, it would certainly be desirable if this patch worked for more than just very specific Telekom servers. Best, Max From m1278468 at mailbox.org Wed Dec 21 02:17:14 2022 From: m1278468 at mailbox.org (Michael Maier) Date: Wed, 21 Dec 2022 09:17:14 +0100 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> <402c0462-5682-2e4b-0127-70278a131e4e@mailbox.org> Message-ID: On 21.12.22 at 08:21 Fridrich Maximilian wrote: >> Security-Client: sdes-srtp;mediasec >> ^^^^^^^^^ >> The ",mediasec" is missing. > > Yes, the security_mechanisms option is a comma separated list of the > literal security_mechanisms that should be used. I.e. you have to > specify security_mechanisms=sdes-srtp\;mediasec,dtls-srtp\;mediasec > (don't forget to escape the semicolon). I got it working now with (order has been important if I remember correctly) security_mechanisms=msrp-tls\;mediasec,sdes-srtp\;mediasec,dtls-srtp\;mediasec I think this should be part of the documentation. But: The Invite has to many headers - those are not needed (or is it Telekom specific?): Security-Client: msrp-tls;mediasec Security-Client: sdes-srtp;mediasec Security-Client: dtls-srtp;mediasec Maybe remove them? > >> always the first entry of the list configured above is dropped in >> the following register request. Is this fixed by your mentioned patch >> below? > > I could not reproduce this behavior. I just tested it with the current > patch and no list entries were dropped. Yes - the current version including your additional patch doesn't show this behavior any more. > >> I think the different headers are not addressed? > > Do you mean the missing ";mediasec" values? That is due to the > configuration, as stated above. Besides that, I'm quite confident it > behaves as intended, we have been running test systems for quite a > while now. > > I hope we can resolve your issues, it would certainly be desirable if > this patch worked for more than just very specific Telekom servers. Yes - that would be very good! Please think about documentation using some practical examples to get a working connection! I tested inbound and outbound calls. I did not test options and reInvite. Did you test reInvites? Thanks Michael From m1278468 at mailbox.org Wed Dec 21 04:20:53 2022 From: m1278468 at mailbox.org (Michael Maier) Date: Wed, 21 Dec 2022 11:20:53 +0100 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> <402c0462-5682-2e4b-0127-70278a131e4e@mailbox.org> Message-ID: On 21.12.22 at 09:17 Michael Maier wrote: > On 21.12.22 at 08:21 Fridrich Maximilian wrote: >>> Security-Client: sdes-srtp;mediasec >>>                          ^^^^^^^^^ >>> The ",mediasec" is missing. >> >> Yes, the security_mechanisms option is a comma separated list of the >> literal security_mechanisms that should be used. I.e. you have to >> specify security_mechanisms=sdes-srtp\;mediasec,dtls-srtp\;mediasec >> (don't forget to escape the semicolon). > > I got it working now with (order has been important if I remember correctly) > > security_mechanisms=msrp-tls\;mediasec,sdes-srtp\;mediasec,dtls-srtp\;mediasec > > I think this should be part of the documentation. > > > But: > The Invite has to many headers - those are not needed (or is it Telekom specific?): > > Security-Client: msrp-tls;mediasec > Security-Client: sdes-srtp;mediasec > Security-Client: dtls-srtp;mediasec > > Maybe remove them? Just detected some more eventually unneeded headers in Invite: Require: mediasec Proxy-Require: mediasec Those three headers in Invite seem to be enough (besides the a=3ge2ae:requested in SDP) - couldn't see any problem during ~ 3 years until now. Security-Verify: msrp-tls;mediasec Security-Verify: sdes-srtp;mediasec Security-Verify: dtls-srtp;mediasec Thanks Michael From M.Fridrich at commend.com Wed Dec 21 08:52:49 2022 From: M.Fridrich at commend.com (Fridrich Maximilian) Date: Wed, 21 Dec 2022 14:52:49 +0000 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> <402c0462-5682-2e4b-0127-70278a131e4e@mailbox.org> Message-ID: > I got it working now with [...] That is excellent news! > I think this should be part of the documentation. I'm not a maintainer but I think usually the documentation is kept quite general without describing specific use cases. The docs for the security_mechanisms parameter say "This is a comma-delimited list of security mechanisms to use. Each security mechanism must be in the form defined by RFC 3329 section 2.2." [1]. I think this should suffice. > The Invite has to many headers Thank you, I will look into it. > I did not test options and reInvite. Did you test reInvites? I have tested OPTIONS and in my current setup I have only tested re-INVITES on outgoing calls from the caller. Best, Max [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip#Asterisk18Configuration_res_pjsip-endpoint_security_mechanisms From m1278468 at mailbox.org Wed Dec 21 09:53:39 2022 From: m1278468 at mailbox.org (Michael Maier) Date: Wed, 21 Dec 2022 16:53:39 +0100 Subject: [asterisk-dev] Asterisk 18.16.0-rc1 Now Available In-Reply-To: References: <6f173021-7cec-3656-0f38-c31c7be5c49d@mailbox.org> <402c0462-5682-2e4b-0127-70278a131e4e@mailbox.org> Message-ID: <192c4b36-e6cc-6ebf-c3fa-2282cc47e2f1@mailbox.org> On 21.12.22 at 15:52 Fridrich Maximilian wrote: >> I got it working now with [...] > > That is excellent news! But the remaining useless headers in Invite should be removed before the final release. >> I think this should be part of the documentation. > > I'm not a maintainer but I think usually the documentation is kept > quite general without describing specific use cases. The docs for the > security_mechanisms parameter say "This is a comma-delimited list of > security mechanisms to use. Each security mechanism must be in the form > defined by RFC 3329 section 2.2." [1]. I think this should suffice. Sorry - I wasn't able to derive the options string needed *for your implementation* based on RFC 3329 section 2.2. For me it wasn't obvious, that the ';mediasec' has to be added (I would have expected, that this is done automatically if the first parameter is set to mediasec (security_negotiation=mediasec)) security_mechanisms=msrp-tls\;mediasec,sdes-srtp\;mediasec,dtls-srtp\;mediasec Therefore I still think this must be part of the documentation. What's wrong to provide an example for a (specific) use case? Why should it be secret? This way, users know, how the config option has to be used without long try and error to guess the correct syntax. Thanks Michael From luke at primevox.net Fri Dec 23 10:47:56 2022 From: luke at primevox.net (=?iso-8859-1?Q?Luke_Escud=E9?=) Date: Fri, 23 Dec 2022 16:47:56 -0000 Subject: [asterisk-dev] Logging and multi-tenancy Message-ID: So, it's possible to achieve multi-tenancy in Asterisk with well-designed dial plan. By multi-tenancy, in terms of Asterisk, I mean each "customer" has their own separate dial plan, and each customer cannot "see" another. However, when it comes to looking at the asterisk CLI/console, you see all the calls flowing across all dial plans, which can be a lot to look at. What is the possibility of adding some kind of "tenancy filtering" to CLI logging? Example: At the beginning of a dial plan call, let's say exten => s, we set LoggingTenancy(customer_id), then all of the subsequent log messages are visible in the CLI only if you run the "logging view tenant (x)" or something like this. That would make it easier to diagnose call issues on high-volume systems. I, personally, do not have the ability to implement this, but I wanted to put it out there. Luke Escudé 972.600.1150 support at primevox.net Schedule a meeting! View the PrimeVOX R&D Roadmap here View the PrimeVOX Status Page here -------------- next part -------------- An HTML attachment was scrubbed... URL: From wojtek at puchar.net Sat Dec 31 09:03:00 2022 From: wojtek at puchar.net (Wojciech Puchar) Date: Sat, 31 Dec 2022 15:03:00 -0000 Subject: [asterisk-dev] SMS with VoIP Message-ID: <4cb1fd5-1276-5831-57c7-8e8a4f8921f8@puchar.net> I have many SIP phones - like gigaset wireless, siemens wired etc. They have "SMS" capability, and there must be "SMS center" set up. That's what i know and nothing else for now. So two questions: 1) Can asterisk act as "SMS center" and send and receive SMS to/from SMS capable voip phones? If yes - please just point me to proper URLs to read. 2) Can asterisk cooperate with SIP operators and send/receive SMS with the rest of world? As well - please point me to proper URLs.