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
Thank you for your continued support of Asterisk!
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
