CryptoPhone IP 19: da Snom e GSMK un telefono a prova di intercettazioni
Lo snom 870 Cryptophone nasce dalla collaborazione tra Snom technology AG e la GSMK ed è in grado di soddissfare le esigenza di sicurezza, di che ne farà uso, grazie all'utilizzo di algoritmi cifrati integrati nel telefono (AES 256 e Twofish) che sono tra i più sicuri e complessi.
Il telefono Snom 870 Cryptophone edition è facile da installare ed usare; ha un ampio display touch screen a colori ed è integrabile in qualsiasi impianto telefonico basato sullo standard aperto SIP.
Per quanto concerne il prezzo, non ho trovato riferimenti nelle fonti ufficiali ma ho trovato altri post "in giro", che parlano di un prezzo raccomandato tra i 2.300 e i 2.900 € (bisogna proprio tenerci alla sicurezza!).
Asterisk 1.8.1 Now Available
Il giorno 8 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.8.1
Dal post originale:
The release of Asterisk 1.8.1 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 issue when using directmedia. Asterisk needs to limit the codecs offered
to just the ones that both sides recognize, otherwise they may end up sending
audio that the other side doesn't understand.
(Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix playback failure when using IAX with the timerfd module.
(Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Fix issue where it is possible to crash Asterisk by feeding the curl engine
invalid data.
(Closes issue #18161. Reported by wdoekes. Patched by tilghman)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
Asterisk 1.4.38 Now Available
Il giorno 8 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.4.38.
Dal post originale:
The release of Asterisk 1.4.38 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:
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Fix a crash in res_jabber by ensuring that we don't alter memory after it's
freed.
(Closes issue #17387. Reported, tested by jmls. Patched by tilghman)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.38
Asterisk 1.6.2.15 Now Available
Il giorno 8 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.6.2.15.
Dal post originale:
he release of Asterisk 1.6.2.15 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:
* When using chan_skinny, don't crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15
Asterisk 1.4.38-rc1 Now Available
Il giorno 23 novembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.4.35-rc1.
Dal post originale:
The release of Asterisk 1.4.38-rc1 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:
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Fix a crash in res_jabber by ensuring that we don't alter memory after it's
freed.
(Closes issue #17387. Reported, tested by jmls. Patched by tilghman)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.38-rc1
Thank you for your continued support of Asterisk!
libpri 1.4.11.5 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione 14.11.5 di LIBPRI.
Questo il contenuto del post originale:
The release of libpri 1.4.11.5 resolves several issues reported by the
community and would not have been possible without your participation.
Thank you!
The following are some of the issues resolved in this release:
* Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the
wrong state.
(issue #17360. Reported by: shawkris. Patched by rmudgett)
* Made Q.921 delay events to Q.931 if the event could immediately
generate response frames.
(closes issue #17360. Reported by: shawkris. Patched by rmudgett)
* BRI PTMP: Active channels not cleared when the interface goes down.
(closes issue #17865. Reported by: wimpy. Patched by rmudgett)
* Segfault in pri_schedule_del() - ctrl value is invalid.
(closes issue #17522)
(closes issue #18032. Reported by: schmoozecom. Patched by rmudgett)
* Crash when receiving an unknown/unsupported message type.
(closes issue #17968. Reported by: gelo. Patched by rmudgett)
* B410P gets incoming call packets on ISDN but Asterisk doesn't see the
call.
(closes issue #18232. Reported by: lelio. Patched by rmudgett)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.11.5
Thank you for your continued support of Asterisk!
Asterisk 1.8.1-rc1 Now Available
Il giorno 23 novembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.8.1-rc1.
The release of Asterisk 1.8.1-rc1 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 issue when using directmedia. Asterisk needs to limit the codecs offered
to just the ones that both sides recognize, otherwise they may end up sending
audio that the other side doesn't understand.
(Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix playback failure when using IAX with the timerfd module.
(Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Fix issue where it is possible to crash Asterisk by feeding the curl engine
invalid data.
(Closes issue #18161. Reported by wdoekes. Patched by tilghman)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.15-rc1 Now Available
Il giorno 23 novembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.6.2.15-rc1.
Dal post originale:
The release of Asterisk 1.6.2.15-rc1 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:
* When using chan_skinny, don't crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.14 Released
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.14.
The Asterisk Development Team has announced the release of Asterisk 1.6.2.14. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.14 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 issue where session timers would be advertised as supported even when session-timers=refuse was set in sip.conf. Also fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)
* Parse all "Accept" headers for SIP SUBSCRIBE requests.
(Closes issue #17758. Reported by ibc. Patched by dvossel)
* Fix issue where queue stats would be reset on reload.
(Closes issue #17535. Reported by raarts. Patched by tilghman)
* Fix issue where MoH files were no longer rescanned on during a reload.
(Closes issue #16744. Reported by pj. Patched by Qwell)
* Fix issue with dialplan pattern matching where the specificity for pattern ranges and pattern characters was inconsistent.
(Closes issue #16903. Reported, patched by Nick_Lewis)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14
Thank you for your continued support of Asterisk!
Asterisk 1.4.37 Released
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.37.
The release of Asterisk 1.4.37 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 issue with decoding ^-escaped characters in realtime (res_pgsql)
(Closes issue #17790. Reported denzs. Patched by Qwell)
* Don't send a devstate change on poke_noanswer if the state did not change.
(Closes issue #17741. Reported, patched by schmidts)
* Transmit silence when reading DTMF in ast_readstring. Otherwise you could get issues with DTMF timeouts causing hangups.
(Closes issue #17370. Reported, patched by makoto)
* Fix to SIP extension state update (deadlock issues)
(Closes issue #17888. Reported by zerohalo. Patched by dvossel)
* Fix issue with MoH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MoH class.
(Closes issue #17807. Reported by kshumard. Patched by bbryant)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.37
Thank you for your continued support of Asterisk!
Elastix: 1.000.000 di donwloads
Elastix
Nato nel 2006, Elastix, il progetto della PaoloSanto Solutions, ha avuto sempre più riscontro nel mercato delle distro basate su Asterisk e come annunciato dalla stessa comunità, ha raggiunto il milione di download.
Questo il link del progetto: ELASTIX
Asterisk 1.8.0 Released
Il giorno 26 ottobre, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.8.0.
Dal post originale:
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to
this release.
You can find a summary of the work involved with the 1.8.0 release in the sumary:
http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
A short list of available features includes:
- Secure RTP
- IPv6 Support in the SIP channel driver
- Connected Party Identification Support
- Calendaring Integration
- A new call logging system, Channel Event Logging (CEL)
- Distributed Device State using Jabber/XMPP PubSub
- Call Completion Supplementary Services support
- Advice of Charge support
- Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0
Thank you for your continued support of Asterisk!
Asterisk 1.8.0 Release Candidate 3 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio della terza beta della 1.8.0.
Dal post originale:
This release candidate contains fixes since the release candidate as reported by
the community. A sampling of the changes in this release candidate include:
* Still build chan_sip even if res_crypto cannot be built (use, but not depend)
(Reported by a user on the mailing list. Patched by tilghman)
* Get notifications for call files only when a file is closed, not when created
(Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
* Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk
expects the DTMF to arrive on the RTP stream and not via jingle DTMF
signalling.
(Patched by dvossel. Tested by malcolmd)
* Fixes to allow chan_gtalk to communicate with the Gmail web client.
(Patched by phsultan and dvossel)
* Fix to GET DATA to allow audio to be streamed via an AGI.
(Closes issue #18001. Reported by jamicque. Patched by tilghman)
* Resolve dnsmgr memory corruption in chan_iax2.
(Closes issue #17902. Reported by afried. Patched by russell, dvossel)
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3
Asterisk 1.8.0-rc2 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio della seconda beta della 1.8.0.
Dal post originale:
Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.
* Make AMI honor enabled=no
(Closes issue #18040. Reported by: twilson
Review: https://reviewboard.asterisk.org/r/938/)
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.
This release candidate contains fixes since the last beta release as reported by
the community. A sampling of the changes in this release candidate include:
* Add slin16 support for format_wav (new wav16 file extension)
(Closes issue #15029. Reported, patched by andrew. Tested by Qwell)
* Fixes a bug in manager.c where the default configuration values weren't reset
when the manager configuration was reloaded.
(Closes issue #17917. Reported by lmadsen. Patched by bbryant)
* Various fixes for the calendar modules.
(Patched by Jan Kalab.
Reviewboard: https://reviewboard.asterisk.org/r/880/
Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)
* Add CHANNEL(checkhangup) to check whether a channel is in the process of
being hung up.
(Closes issue #17652. Reported, patched by kobaz)
* Fix a bug with MeetMe where after announcing the amount of time left in a
conference, if music on hold was playing, it doesn't restart.
(Closes issue #17408, Reported, patched by sysreq)
* Fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)
* Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
determined to be one of the most significant bottlenecks in SIP registration
processing. This patch improved the speed of an astdb load test by 50000%
(yes, Fifty-Thousand Percent). On this particular load test setup, this
doubled the number of SIP registrations the server could handle.
(Review: https://reviewboard.asterisk.org/r/825/)
* Don't clear the username from a realtime database when a registration
expires. Non-realtime chan_sip does not clear the username from memory when a
registration expiries so realtime probably shouldn't either.
(Closes issue #17551. Reported, patched by: ricardolandim. Patched by
mnicholson)
* Don't hang up a call on an SRTP unprotect failure. Also make it more obvious
when there is an issue en/decrypting.
(Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
twilson)
* Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2
Asterisk 1.6.2.14-rc1 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.14-rc1
Dal post originale:
The release of Asterisk 1.6.2.14-rc1 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:
* Fix issue where session timers would be advertised as supported even when
session-timers=refuse was set in sip.conf. Also fix interoperability
problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)
* Fix issue with decoding ^-escaped characters in realtime (res_pgsql).
(Closes issue #17790. Reported by denzs. Patched by Qwell)
* Parse all "Accept" headers for SIP SUBSCRIBE requests.
(Closes issue #17758. Reported by ibc. Patched by dvossel)
* Fix issue where queue stats would be reset on reload.
(Closes issue #17535. Reported by raarts. Patched by tilghman)
* Fix issue where MoH files were no longer rescanned on during a reload.
(Closes issue #16744. Reported by pj. Patched by Qwell)
* Fix issue with dialplan pattern matching where the specificity for pattern
ranges and pattern characters was inconsistent.
(Closes issue #16903. Reported, patched by Nick_Lewis)
Questo il ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14-rc1