ASTERWEB Blog

15Dic/110

Rilasciato Asterisk 10.0.0-rc3

Il giorno 09 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 10.0.0-rc3

Dal post originale:
The Asterisk Development Team has announced the third release candidate of
Asterisk 10.0.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.0.0-rc3 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Add ASTSBINDIR to the list of configurable paths
This patch also makes astdb2sqlite3 and astcanary use the configured
directory instead of relying on $PATH.

Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible
to crash Asterisk by sending an INFO request if no channel had been
created yet.

Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages.

Fix a change in behavior in 'database show' from 1.8.
In 1.8 and previous versions, one could use any fullword portion of
the key name, including the full key, to obtain the record. Until this
patch, this did not work for the full key.

Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and
user/peer nat settings differ in whether to respond to the port a request is
sent from or the port listed for responses in the Via header. In 1.4 and
1.6.2, this would mean if one setting was nat=yes or nat=route and the other
was either nat=no or nat=never. In 1.8 and 10, this would mean when one
was nat=force_rport and the other was nat=no.

In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all
possible.

Fixed SendMessage stripping extension from To: header in SIP MESSAGE
When using the MessageSend application to send a SIP MESSAGE to a
non-peer, chan_sip stripped off the extension and failed to add it back
to the sip_pvt structure before transmitting. This patch adds the full
URI passed in from the message core to the sip_pvt structure.

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3

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15Dic/110

Rilasciato Asterisk 1.8.8.0-rc5

Il giorno 09 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.8.0-rc5

Dal post originale:
he release of Asterisk 1.8.8.0-rc5 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible
to crash Asterisk by sending an INFO request if no channel had been
created yet.

Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages.

Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and
user/peer nat settings differ in whether to respond to the port a request is
sent from or the port listed for responses in the Via header. In 1.4 and
1.6.2, this would mean if one setting was nat=yes or nat=route and the other
was either nat=no or nat=never. In 1.8 and 10, this would mean when one
was nat=force_rport and the other was nat=no.

In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all
possible.

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc5

Inserito in: Asterisk Nessun commento
15Dic/110

Rilasciati Asterisk (Security Release) 1.4.43, 1.6.2.21 e 1.8.7.2

Il giorno 12 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio delle versioni Asterisk  (Security Release) 1.4.43, 1.6.2.21 e 1.8.7.2

Dal post originale:
hese releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk versions 1.4.43, 1.6.2.21, and 1.8.7.2 resolves an issue
with possible remote enumeration of SIP endpoints with differing NAT settings.

The release of Asterisk versions 1.6.2.21 and 1.8.7.2 resolves a remote crash
possibility with SIP when the "automon" feature is enabled.

The issues and resolutions are described in the AST-2011-013 and AST-2011-014
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-013 and AST-2011-014, which were released at the
same time as this announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

Security advisory AST-2011-013 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-013.pdf
Security advisory AST-2011-014 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-014.pdf

29Nov/110

Mercoledì 6 dicembre FREE WEBINAR parte II: “Monitoriamo i nostri Asterisk (e non solo) con Nagios”

Mercoledì 6 dicembre, dalle ore 14:00 alle ore 15:00, si terrà il FREE WEBINAR parte II: "Monitoriamo i nostri Asterisk (e non solo) con Nagios" organizzato in collaborazione con Sigmaware Srl ed aperto a tutti.

Questo incontro sarà di carattere esclusimamente tecnico. Vedremo:

  • struttura dei file di configurazione di Nagios
  • gestione contatti e alert
  • host e servizi da monitorare
  • ... altro ...

Attenzione: potete richiederci il pdf del precedente Free Webinar di Nagios, scrivendoci a webinar@sigmaware.it

Per l'adesione: www.asterweb.org nella home page, troverai il form da compilare.

Per qualsiasi info, puoi contattarci:

- CHAT: dal sito www.asterweb.org

- SKYPE: asterweb

- MSN: asterweb@tiscali.it

- TELEFONO: 02-45077711

 

Con l'auspicio di incontrarti al webinar, ti salutiamo cordialemte

ASTERWEB

Lo Staff

 

Il nostro software per tutte le distro.

Scopri le tante funzioni che trasformeranno il tuo centralino e miglioreranno l'organizzazione della tua Azienda. Scarica la DEMO gratuita. Clicca QUI

20Nov/110

FREE WEBINAR: “Monitoriamo i nostri Asterisk (e non solo) con Nagios”

Mercoledì 23 novembre, dalle ore 14:00 alle ore 15:00, si terrà il FREE WEBINAR: "Monitoriamo i nostri Asterisk (e non solo) con Nagios" organizzato in collaborazione con Sigmaware Srl ed aperto a tutti.

Vedremo come: 

  • configurare i servizi lato server
  • utilizzare i plugins
  • definire i servizi
  • ... altro ...

Per l'adesione: www.asterweb.org nella home page, troverai il form da compilare.

Per qualsiasi info, puoi contattarci:

- CHAT: dal sito www.asterweb.org

- SKYPE: asterweb

- MSN: asterweb@tiscali.it

- TELEFONO: 02-45077711

 

Con l'auspicio di incontrarti al webinar, ti salutiamo cordialemte

ASTERWEB

Lo Staff

 

Il nostro software per tutte le distro.

Scopri le tante funzioni che trasformeranno il tuo centralino e miglioreranno l'organizzazione della tua Azienda. Scarica la DEMO gratuita. Clicca QUI

18Nov/110

Rilasciato Asterisk 1.8.8.0-rc4

Il giorno 17 novembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.8.0-rc4

Dal post originale:

The release of Asterisk 1.8.8.0-rc4 resolves a particular issue with BLF
subscriptions. A change in Asterisk 1.8.8.0-rc3 had the potential to cause a
segfault, and this release candidate was created to resolve that.

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc4

Thank you for your continued support of Asterisk!

Inserito in: Asterisk Nessun commento
6Nov/110

FREE WEBINAR: “Asterisk: integrare Google Calendar”

Mercoledì 9 novembre, dalle ore 14:00 alle ore 15:00, si terrà il FREE WEBINAR: "Asterisk: integrare Google Calendar" organizzato in collaborazione con Sigmaware Srl ed aperto a tutti.

Vedremo come: 

  • leggere i dati del Calendar
  • effettuare operazioni lato Asterisk sul DialPlan
  • scrivere sul Calendar
  • ... altro ...

Per l'adesione: www.asterweb.org nella home page, troverai il form da compilare.

Per qualsiasi info, puoi contattarci:

- CHAT: dal sito www.asterweb.org

- SKYPE: asterweb

- MSN: asterweb@tiscali.it

- TELEFONO: 02-45077711

 

Con l'auspicio di incontrarti al webinar, ti salutiamo cordialemte

ASTERWEB

Lo Staff

 

Il nostro software per tutte le distro.

Scopri le tante funzioni che trasformeranno il tuo centralino e miglioreranno l'organizzazione della tua Azienda. Scarica la DEMO gratuita. Clicca QUI

2Nov/110

Rilasciato Asterisk 1.8.7.1 (Security Release)

Il giorno 17 ottobre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.7.1 (Security Release)

Dal post originale:
The release of Asterisk 1.8.7.1 resolves an issue with SIP URI parsing which can
lead to a remotely exploitable crash:

Remote Crash Vulnerability in SIP channel driver (AST-2011-012)

The issue and resolution is described in the AST-2011-012 security
advisory.

For more information about the details of this vulnerability, please read the
security advisory AST-2011-012, which was released at the same time as this
announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.7.1

Security advisory AST-2011-012 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-012.pdf

14Ott/110

Free Webinar: Asterisk e Sicurezza

Mercoledì 19 ottobre, dalle ore 14:00 alle ore 15:00 abbiamo organizzato il webinar: "Asterisk e Sicurezza".

L'argomento riteniamo sia interessante ed attuale e per questo attendiamo numerose le iscrizioni.

Per l'iscrizione, compilare il form alla pagina WEBINAR ASTERWEB o inviare una mail a: incontri@sigmaware.it specificando il nome dell'azienda, i nomi dei partecipanti e le rispettive e-mail a cui inviare la conferma di partecipazione.

Buon lavoro e buon business a tutti

7Ott/110

Webinar per Partner Sigmaware: Asterisk e Sicurezza

In collaborazione con Sigmaware Srl, abbiamo organizzato per martedì 11 ottobre il WEBINAR: "Asterisk e Sicurezza" riservato ai nostri Partners.

Il Webinar avrà la durata di 1 ora, dalle ore 14:00 alle ore 15:00

Buon lavoro e buon business a tutti

1Ott/110

Rilasciato Asterisk 10.0.0-beta2

Il giorno 27 settembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 10.0.0-beta2

Dal post originale:
The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a...

All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of features includes:

T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

Inserito in: Asterisk Nessun commento
26Set/110

Rilasciato Astrisk 1.8.7.0

Il giorno 23 settembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.7

Dal post originale:
he release of Asterisk 1.8.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

Please note that a significant numbers of changes and fixes have gone into
features.c in this release (call parking, built-in transfers, call pickup,
etc.).

NOTE:

Recently, we were notified that the mechanism included in our Asterisk source
code releases to download and build support for the iLBC codec had stopped
working correctly; a little investigation revealed that this occurred because of
some changes on the ilbcfreeware.org website. These changes occurred as a result
of Google's acquisition of GIPS, who produced (and provided licenses for) the
iLBC codec.

If you are a user of Asterisk and iLBC together, and you've already executed a
license agreement with GIPS, we believe you can continue using iLBC with
Asterisk. If you are a user of Asterisk and iLBC together, but you had not
executed a license agreement with GIPS, we encourage you to research the
situation and consult with your own legal representatives to determine what
actions you may want to take (or avoid taking).

More information is available on the Asterisk blog:

http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-goog...

The following is a sample of the issues resolved in this release:

Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set
HASH(SIP_CAUSE,) on the channel carries a significant performance
penalty because of the usage of the MASTER_CHANNEL() dialplan function.

We've decided to disable this feature by default in future 1.8 versions. This
would be an unexpected behavior change for anyone depending on that SIP_CAUSE
update in their dialplan. Please refer to the asterisk-dev mailing list more
information:

http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
Significant fixes and improvements to parking lots.
(Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452,
ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada,
Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)
Numerous issues have been reported for deadlocks that are caused by a blocking
read in res_timing_timerfd on a file descriptor that will never be written to.

A change to Asterisk adds some checks to make sure that the timerfd is both
valid and armed before calling read(). Should fix: ASTERISK-18142,
ASTERISK-18197, ASTERISK-18166 and possibly others.
(In essence, this change should make res_timing_timerfd usable.)
Resolve segfault when publishing device states via XMPP and not connected.
(Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested
by Jonathan Rose)
Refresh peer address if DNS unavailable at peer creation.
(Closes issue ASTERISK-18000)
Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections
for channel configuration.
(Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard
Mudgett)
Remove unnecessary libpri dependency checks in the configure script.
(Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard
Mudgett)
Update get_ilbc_source.sh script to work again.
(Closes issue ASTERISK-18412)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0

Inserito in: Asterisk Nessun commento
6Set/110

Rilasciato Asterisk 1.8.6.0

Il giorno 31 agosto, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.6

Dal post originale:
The release of Asterisk 1.8.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Fix an issue with Music on Hold classes losing files in playlist when realtime
is used.
(Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor
Goncharovsky)
Resolve a potential crash in chan_sip when utilizing auth= and performing a
'sip reload' from the console.
(Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett)
Address some improper sql statements in res_odbc that would cause an update
to fail on realtime peers due to trying to set as "(NULL)" rather than an
actual NULL.
(Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by Tilghman
Lesher)
Resolve issue where 403 Forbidden would always be sent maximum number of times
regardless to receipt of ACK.
(Patched by Richard Mudgett)
Resolve issue where if a call to MeetMe includes both the dynamic(D) and
always request PIN(P) options, MeetMe will ask for the PIN two times: once for
creating the conference and once for entering the conference.
(Patched by Kinsey Moore)
Fix New Zealand indications profile based on
http://www.telepermit.co.nz/TNA102.pdf
(Closes issue ASTERISK-16263. Reported, Patched by richardf)
Segfault in shell_helper in func_shell.c
(Closes issue ASTERISK-18109. Reported by Michael Myles, patched by Richard
Mudgett)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0

Inserito in: Asterisk Nessun commento
10Ago/110

Rilasciato Asterisk 1.6.20.0

Il giorno 08 agosto, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.6.20.0

Dal post originale:
The Asterisk Development Team announces the release of Asterisk 1.6.2.20. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.20 resolves a regression that was introduced just
prior to the release of Asterisk 1.6.2.19.

Fix reload crash caused by destroying default parking lot.
(Closes issue ASTERISK-18103. Reported by 808blogger. Patched by jrose.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

Inserito in: Asterisk Nessun commento
2Ago/110

Rilasciato Asterisk 10.0.0 Beta 1

Il giorno 22 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 10.0.0 Beta 1

Dal post originale:
The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 10.0.0-beta1. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a...

All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.
Additionally users can make use of the RPM and DEB packages now being built for
all Asterisk releases. More information available at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of included features includes:

T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

Inserito in: Asterisk Nessun commento