[asterisk-users] Asterisk 18.3.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.3.0 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: Security bugs fixed in this release: --- * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash (Reported by Gregory Massel) * ASTERISK-29260 - sRTP Replay Protection ignored; even tears down long calls (Reported by Alexander Traud) * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash (Reported by Ivan Poddubny) Bugs fixed in this release: --- * ASTERISK-29215 - res_pjsip_session: NULL active_media_state topology caused asterisk crash (Reported by sungtae kim) * ASTERISK-29035 - chan_local: Multistream support breaks T.38 faxing (Reported by Matthias Hensler) * ASTERISK-29071 - app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs (Reported by Stefan Ruf) * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events (Reported by N A) * ASTERISK-24434 - Fix differing usage of assignment operators in modules.conf (Reported by Rusty Newton) * ASTERISK-29306 - strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition (Reported by Vitezslav Novy) * ASTERISK-29300 - res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent (Reported by Sebastian Damm) * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address (Reported by Brian Paboojian) * ASTERISK-29266 - ICE Role conflict with an unauthorized session (Reported by Salah Ahmed) * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed into progress (Reported by Sebastian Damm) * ASTERISK-29297 - say: Y2021 problem â Asterisk cannot say year 2021 in Dutch (Reported by Jacek Konieczny) * ASTERISK-29315 - res_pjsip: re-registration gets stuck if setting initial auth credentials fails (Reported by Nick French) * ASTERISK-29312 - res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters (Reported by Alexei Gradinari) * ASTERISK-16799 - Callee declined when 'beep' audio file does not exist (Reported by IAMJames_) * ASTERISK-29313 - res_pjsip_refer: Segfault in progress notify (Reported by George Joseph) * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to return one (no more) record (Reported by Boris P. Korzun) * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't (Reported by Benjamin Keith Ford) * ASTERISK-29311 - res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit (Reported by Jaco Kroon) * ASTERISK-28452 - pjsip: of SDP is not incremented though SDP may be changed on reinvite without SDP offer (Reported by Michael Maier) * ASTERISK-29287 - app.h: C++ compatibility broken (Reported by Jean Aunis - Prescom) * ASTERISK-28369 - app_queue: Member device state "invalid" when second call is ringing and hint is used (Reported by Boolah ) * ASTERISK-29203 - res_pjsip_t38: Crash when changing state (Reported by Gregory Massel) * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client (Reported by Edvin Vidmar) * ASTERISK-29196 - res_pjsip: Segmentation fault (Reported by Mauri de Souza Meneguzzo (3CPlus)) * ASTERISK-29280 - chan_sip: Allow peers without audio (text+video). (Reported by Alexander Traud) * ASTERISK-29265 - chan_sip: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29261 - res_pjsip: user=phone validation fail for isup numbers containing *# (Reported by Mark Petersen) * ASTERISK-29259 - channel: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29258 - chan_sip: Audio stream rejected, Other stream present: Invalid SDP. (Reported by Alexander Traud) * ASTERISK-29220 - After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used (Reported by Robert Cripps) * ASTERISK-29248 - res_pjsip_session: res sometimes uninitialized reported by compi
[asterisk-users] Asterisk 16.17.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.17.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.17.0 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: Security bugs fixed in this release: --- * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash (Reported by Gregory Massel) * ASTERISK-29260 - sRTP Replay Protection ignored; even tears down long calls (Reported by Alexander Traud) * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash (Reported by Ivan Poddubny) Bugs fixed in this release: --- * ASTERISK-29215 - res_pjsip_session: NULL active_media_state topology caused asterisk crash (Reported by sungtae kim) * ASTERISK-29035 - chan_local: Multistream support breaks T.38 faxing (Reported by Matthias Hensler) * ASTERISK-29071 - app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs (Reported by Stefan Ruf) * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events (Reported by N A) * ASTERISK-24434 - Fix differing usage of assignment operators in modules.conf (Reported by Rusty Newton) * ASTERISK-29306 - strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition (Reported by Vitezslav Novy) * ASTERISK-29300 - res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent (Reported by Sebastian Damm) * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address (Reported by Brian Paboojian) * ASTERISK-29266 - ICE Role conflict with an unauthorized session (Reported by Salah Ahmed) * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed into progress (Reported by Sebastian Damm) * ASTERISK-29297 - say: Y2021 problem â Asterisk cannot say year 2021 in Dutch (Reported by Jacek Konieczny) * ASTERISK-29312 - res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters (Reported by Alexei Gradinari) * ASTERISK-16799 - Callee declined when 'beep' audio file does not exist (Reported by IAMJames_) * ASTERISK-29313 - res_pjsip_refer: Segfault in progress notify (Reported by George Joseph) * ASTERISK-28452 - pjsip: of SDP is not incremented though SDP may be changed on reinvite without SDP offer (Reported by Michael Maier) * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't (Reported by Benjamin Keith Ford) * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to return one (no more) record (Reported by Boris P. Korzun) * ASTERISK-29311 - res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit (Reported by Jaco Kroon) * ASTERISK-28369 - app_queue: Member device state "invalid" when second call is ringing and hint is used (Reported by Boolah ) * ASTERISK-29287 - app.h: C++ compatibility broken (Reported by Jean Aunis - Prescom) * ASTERISK-29203 - res_pjsip_t38: Crash when changing state (Reported by Gregory Massel) * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client (Reported by Edvin Vidmar) * ASTERISK-29196 - res_pjsip: Segmentation fault (Reported by Mauri de Souza Meneguzzo (3CPlus)) * ASTERISK-29280 - chan_sip: Allow peers without audio (text+video). (Reported by Alexander Traud) * ASTERISK-29265 - chan_sip: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29259 - channel: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29261 - res_pjsip: user=phone validation fail for isup numbers containing *# (Reported by Mark Petersen) * ASTERISK-29258 - chan_sip: Audio stream rejected, Other stream present: Invalid SDP. (Reported by Alexander Traud) * ASTERISK-29220 - After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used (Reported by Robert Cripps) * ASTERISK-29248 - res_pjsip_session: res sometimes uninitialized reported by compiler Clang. (Reported by Alexander Traud) Improvements made in this release: --- * ASTERISK-29321
Re: [asterisk-users] STIR/SHAKEN
Hey All. I spoke to the guys at TILTX and they agreed to host a 30 minute webinar for STIR/SHAKEN and Asterisk. They will coordinate internally and they will send me an invite. I will share this invite in the event anybody would like to join. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Realtime peers
Hi. Has nobody else tried to do this, or worked out how to (or, possibly, reported it as a bug)? On Saturday 13 March 2021 at 16:13:04, Antony Stone wrote: > On Thursday 11 March 2021 at 14:03:23, Antony Stone wrote: > > Hi. > > > > I'm using MariaDB (via ODBC) to store realtime SIP peers with Asterisk 13 > > and 16. > > > > In general it's all working fine, however there's something that puzzles > > me: > > > > If I connect to the Asterisk console and use the command "sip show peers" > > I get a list of the peers including the last qualify time in the > > "Status" column. The "Realtime" column tells me "Cached RT". > > > > Some of my peers are in the same data centre as the server I'm querying, > > and generally show a qualify time of 1ms; some are in another data centre > > and have qualify times around 15ms. > > > > However, if I go to my database server and ask "select * from sippeers" I > > get the same list of peers but the "lastms" field is always zero. I had > > expected this to show me the last qualify time in milliseconds for each > > peer. > > > > Am I just totally misinterpreting what "lastms" means in the table, or do > > I need to do something else to get this value to reflect what Asterisk > > itself will tell me? > > I've done a little more investigating, and from both documentation and the > source code I've established that this field should be getting updated > provided that "rtupdate" is set to "yes" in sip.conf, however as that is > the default and I have not set it to "no", I had expected the value indeed > to be "yes". > > Setting it explicitly to "yes" makes no difference. > > The totally unexpected part, though, is that if I modify the value of > "lastms" in the database table, and then just wait about 30 seconds, > Asterisk changes all the values back to zero. > > So, Asterisk definitely *is* updating this field in the database table; > it's just not using the correct values. > > > Can anyone please comment on whether this is the bug it looks like to me, > or whether I'm just missing some configuration setting which would make > this work as expected? > > > Thanks, > > Antony. -- Atheism is a non-prophet-making organisation. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
