ASTERWEB Blog

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 1.8.1-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 1.8.0 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 rilasciate: 1.4.28, 1.6.0.20, 1.6.1.12 e 1.6.2.0

Il 18/12 il Team di Sviluppo di asterisk ha rilascito le versioni 1.4.28, 1.6.0.20, 1.6.1.12 e 1.6.2.0 scaricabili da http://downloads.asterisk.org/pub/telephony/asterisk/.

Questi i post oriniginali:

• 1.4.28
• 1.6.0.20
• 1.6.1.12
• 1.6.2.0

Asterisk 1.4.28-rc1, 1.6.0.20-rc1, 1.6.1.12-rc1, and 1.6.2.0-rc8 released

Il team di sviluppo di Asterisk ha rilasciato le nuove versioni canditade di Asterisk: 1.4.28, 1.6.0.20, 1.6.1.12 e 1.6.2.0.

Per il download: http://downloads.asterisk.org/pub/telephony/asterisk/

Questi i ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.28-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.20-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.12-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.0-rc8

Questo il link del post originale: http://www.asterisk.org/node/49876