ASTERWEB Blog

24Set/19Off

FreeSWITCH v1.10.1 Release

FreeSWITCH 1.10.0 was released earlier this month (one year ahead of schedule) and now we are excited to announce its first update. Thank you to our community for all of your feedback, we have been working hard to incorporate it into our 1.10.1 release.

 

...the link of the post: https://freeswitch.com/index.php/2019/08/20/freeswitch-v1-10-1-release/

20Set/19Off

Asterisk 17.0.0-rc1 Now Available

The Asterisk Development Team would like to announce the first release candidate of Asterisk 17.0.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.0.0-rc1 resolves several issues reported by the
community.

 

...the link of the post: https://www.asterisk.org/downloads/asterisk-news/asterisk-1700-rc1-now-available

 

17Set/19Off

Kamailio v5.2.4 stable Released

Kamailio SIP Server v5.2.4 stable is out – a minor release including fixes in code and documentation since v5.2.3. The configuration file and database schema compatibility is preserved, which means you don’t have to change anything to update.

Kamailio  v5.2.4 is based on the latest source code of GIT branch 5.2 and it represents the latest stable version. We recommend those running previous 5.2.x or older versions to upgrade. There is no change that has to be done to configuration file or database structure comparing with the previous releases of the v5.2 branch.

...the link of the post: https://www.kamailio.org/w/2019/08/kamailio-v5-2-4-released/

13Set/19Off

VitalPBX 2.3.6 announce the release

We are excited to announce the release of VitalPBX 2.3.6. This version presents new add-ons, features, and various bug fixes. Thanks to all our beta testers for reporting issues, suggesting improvements, and help us deliver a very stable version to all the VitalPBX community.

 

Next, we will list all the changes included in this version.

 

.... the link of the post: https://vitalpbx.org/en/vitalpbx-2-3-6/

14Mar/19Off

VitalPBX 2.3.0 – Stable Release

Vital-PBX

Vital-PBX

Finally, the wait is over, today we are officially launching our new stable version, VitalPBX 2.3.0, this version comes to stabilize all the features announced in the release candidate (VitalPBX 2.2.2-1RC), and introduce some other features.

Here’s a list all the improvements, new features, fixes and some notes about this version.

... il link del post: https://vitalpbx.org/en/vitalpbx-2-3-0-stable-release/

13Mar/19Off

Kamailio v5.2.2 Released

kamailio-logo-nuovo

E stata rilasciata la versione 5.2.2 di Kamailio.

Dal post originale:

Kamailio SIP Server v5.2.2 stable is out – a minor release including fixes in code and documentation since v5.2.1. The configuration file and database schema compatibility is preserved, which means you don’t have to change anything to update.

Kamailio® v5.2.2 is based on the latest source code of GIT branch 5.2 and it represents the latest stable version. We recommend those running previous 5.2.x or older versions to upgrade. There is no change that has to be done to configuration file or database structure comparing with the previous releases of the v5.2 branch.

Il link: https://www.kamailio.org/w/2019/03/kamailio-v5-2-2-released/

1Mar/19Off

VitalPBX 2.2.2-RC1

Vital-PBX

Vital-PBX

After a long time without any updates, we are glad to announce the first release candidate of VitalPBX. This version is only available through the ISO, so, current installations will no be able to migrate to this new version until we make sure that everything is fairly stable.

Here is the list of the improvements, fixes, and new features.

... il link del post: https://vitalpbx.org/en/vitalpbx-2-2-2-1rc/

28Feb/19Off

Asterisk 15.7.2 and 16.2.1 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 15 and 16. The available releases are released as versions 15.7.2 and 16.2.1.

These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases 10

The following security vulnerabilities were resolved in these versions:

AST-2019-001: Remote crash vulnerability with SDP protocol violation When Asterisk makes an outgoing call, a very specific SDP protocol violation by the remote party can cause Asterisk to crash.

For a full list of changes in the current releases, please see the ChangeLogs:

ChangeLog-15.7.2 4
ChangeLog-16.2.1 18

The security advisory is available at:

AST-2019-001.pdf 13

Thank you for your continued support of Asterisk!

18Feb/19Off

Kamailio v5.2.1 Released

kamailio-logo-nuovo

E stata rilasciata la versione 5.2.1 di Kamailio.

Dal post originale:

Kamailio SIP Server v5.2.1 stable is out – a minor release including fixes in code and documentation since v5.2.0. The configuration file and database schema compatibility is preserved, which means you don’t have to change anything to update.

Kamailio® v5.2.1 is based on the latest source code of GIT branch 5.2 and it represents the latest stable version. We recommend those running previous 5.2.x or older versions to upgrade. There is no change that has to be done to configuration file or database structure comparing with the previous releases of the v5.2 branch.

Il link: https://www.kamailio.org/w/2019/01/kamailio-v5-2-1-released/

15Feb/19Off

Asterisk 13.25.0-rc3 Now Available

The Asterisk Development Team would like to announce the third release candidate of Asterisk 13.25.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.25.0-rc3 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------

[ASTERISK-28288] -

Resources (udptl fd) leaking for T.38 calls (Reported by Paulo Vicentini)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.25.0-rc3

Thank you for your continued support of Asterisk!

13Feb/19Off

Asterisk 13.25.0-rc2 Now Available

The Asterisk Development Team would like to announce the second release candidate of Asterisk 13.25.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.25.0-rc2 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------

[ASTERISK-28213] -

res_pjsip: Threads pile up needlessly when AOR is blocked
(Reported by Ross Beer)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.25.0-rc2

Thank you for your continued support of Asterisk!

13Feb/19Off

Asterisk 16.2.0-rc2 Now Available

The Asterisk Development Team would like to announce the second release candidate of Asterisk 16.2.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.2.0-rc2 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------

[ASTERISK-28213] -

res_pjsip: Threads pile up needlessly when AOR is blocked
(Reported by Ross Beer)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0-rc2

Thank you for your continued support of Asterisk!

8Feb/19Off

Asterisk 13.25.0-rc1 Now Available

The Asterisk Development Team would like to announce the first release candidate of Asterisk 13.25.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.25.0-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:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will not compile
(Reported by David Wilcox)
* ASTERISK-28104 - AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps
(Reported by George Joseph)
* ASTERISK-28238 - PJSIP realtime. getcontext not working with DUNDI
(Reported by Ray)
* ASTERISK-28173 - Deadlock in chan_sip handling subscribe request during res_parking reload
(Reported by Giuseppe Sucameli)
* ASTERISK-28263 - codec_opus: errors setting max_playback_rate and bitrate to "sdp"
(Reported by Gianluca Merlo)
* ASTERISK-28250 - build: Cross-compilation fails for target arm-linux-gnueabihf
(Reported by Jean Aunis - Prescom)
* ASTERISK-28156 - Race condition involving session->media (res_pjsip_session) leads to crash.
(Reported by Paulo Vicentini)
* ASTERISK-28257 - res_http_websocket: PING / PONG opcodes break data reception
(Reported by Jeremy Lainé)
* ASTERISK-28252 - HangupHandler manager events are never thrown
(Reported by Gerald Schnabel)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28231 - res_http_websocket: Not responding to Connection Close Frame (opcode 8)
(Reported by Jeremy Lainé)
* ASTERISK-28249 - res_monitor: Segfault with Monitor(wav,file,i)
(Reported by Valentin Vidić)
* ASTERISK-28244 - stasis: Filter messages at publishing to AMI/ARI
(Reported by Joshua C. Colp)
* ASTERISK-28197 - stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases
(Reported by Mohit Dhiman)
* ASTERISK-28232 - core: RAII using clang use-after-scope issue
(Reported by Diederik de Groot)
* ASTERISK-28225 - app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent"
(Reported by boatright)
* ASTERISK-28212 - stasis: Statistics broke ABI under developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
* ASTERISK-28221 - Bug in ast_coredumper
(Reported by Andrew Nagy)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation
(Reported by Alexei Gradinari)
* ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs
(Reported by George Joseph)
* ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
(Reported by David Kuehling)
* ASTERISK-28117 - stasis: Add statistics for usage when in developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28201 - [patch] confbridge: no announce to the marked users when they join an empty conference
(Reported by Alexei Gradinari)
* ASTERISK-28194 - chan_sip: Leak using contact ACL
(Reported by Giuseppe Sucameli)
* ASTERISK-28186 - stasis: Filter messages at publishing based on to_* presence
(Reported by Joshua C. Colp)
* ASTERISK-27095 - chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE
(Reported by George Joseph)
* ASTERISK-28182 - chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE
(Reported by nappsoft)

Improvements made in this release:
-----------------------------------
* ASTERISK-28246 - Support skipping on the g726 format

(Reported by Eyal Hasson)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.25.0-rc1

Thank you for your continued support of Asterisk!

8Feb/19Off

Asterisk 16.2.0-rc1 Now Available

The Asterisk Development Team would like to announce the first release candidate of Asterisk 16.2.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.2.0-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:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-28173 - Deadlock in chan_sip handling subscribe request during res_parking reload
(Reported by Giuseppe Sucameli)
* ASTERISK-28104 - AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps
(Reported by George Joseph)
* ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will not compile
(Reported by David Wilcox)
* ASTERISK-28238 - PJSIP realtime. getcontext not working with DUNDI
(Reported by Ray)
* ASTERISK-28263 - codec_opus: errors setting max_playback_rate and bitrate to "sdp"
(Reported by Gianluca Merlo)
* ASTERISK-28250 - build: Cross-compilation fails for target arm-linux-gnueabihf
(Reported by Jean Aunis - Prescom)
* ASTERISK-28257 - res_http_websocket: PING / PONG opcodes break data reception
(Reported by Jeremy Lainé)
* ASTERISK-28252 - HangupHandler manager events are never thrown
(Reported by Gerald Schnabel)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28249 - res_monitor: Segfault with Monitor(wav,file,i)
(Reported by Valentin Vidić)
* ASTERISK-28244 - stasis: Filter messages at publishing to AMI/ARI
(Reported by Joshua C. Colp)
* ASTERISK-28231 - res_http_websocket: Not responding to Connection Close Frame (opcode 8)
(Reported by Jeremy Lainé)
* ASTERISK-28197 - stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases
(Reported by Mohit Dhiman)
* ASTERISK-28232 - core: RAII using clang use-after-scope issue
(Reported by Diederik de Groot)
* ASTERISK-28230 - res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony
(Reported by David Kuehling)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation
(Reported by Alexei Gradinari)
* ASTERISK-28225 - app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent"
(Reported by boatright)
* ASTERISK-28218 - app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b)
(Reported by Mark)
* ASTERISK-28212 - stasis: Statistics broke ABI under developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
* ASTERISK-28221 - Bug in ast_coredumper
(Reported by Andrew Nagy)
* ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs
(Reported by George Joseph)
* ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
(Reported by David Kuehling)
* ASTERISK-28201 - [patch] confbridge: no announce to the marked users when they join an empty conference
(Reported by Alexei Gradinari)
* ASTERISK-28117 - stasis: Add statistics for usage when in developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28186 - stasis: Filter messages at publishing based on to_* presence
(Reported by Joshua C. Colp)
* ASTERISK-28194 - chan_sip: Leak using contact ACL
(Reported by Giuseppe Sucameli)
* ASTERISK-27095 - chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE
(Reported by George Joseph)
* ASTERISK-28182 - chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE
(Reported by nappsoft)
* ASTERISK-28157 - Asterisk crashes when the res_pjsip_* modules unload
(Reported by sungtae kim)

Improvements made in this release:
-----------------------------------
* ASTERISK-28246 - Support skipping on the g726 format
(Reported by Eyal Hasson)
* ASTERISK-28196 - bridge_softmix: Does not support WebRTC source with multi video tracks.
(Reported by Xiemin Chen)
* ASTERISK-28198 - res_ari: Add new hangup causes for ARI Channel DELETE command
(Reported by Sebastian Damm)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0-rc1

Thank you for your continued support of Asterisk!

6Feb/19Off

L’assistente telefonico “educato” per tutti i centralini Asterisk (lo vorrai anche tu)

Molte sono le applicazioni che inglobano un sistema di riconoscimento vocale sempre finalizzato a rendere l'esperienza d'uso più smart. Poco è stato fatto però nell'ambito PBX contestualizzato alle aziende, almeno fino ad ora. Infatti, l' azienda campana BeeVoip ha messo a punto una sorta di segretaria virtuale con un minimo di intelligenza artificiale in grado di riconoscere il cliente chiamante e di instradarlo verso la persona o il settore di competenza desiderati. L'idea è molto interessante, visto che tra l'altro il sistema che hanno sviluppato si aggancia al PBX Asterisk senza l'ausilio di API quali quelle di GOOGLE, Lumenvox o altri. Allego il video di una dimostrazione che certamente promette molto.