I see it all the time, but now when I am trying to find it of course I can't. What I am finding is not answering the question thoroughly!
Many times in the forum people(read: eeman) have stated that Asterisk 1.6 is ok if you can live with the fact that things don't work well or not at all.
Is this still true?
If so what does not work or what only works and is it worth the loss of one function but I gain more function?
Is 1.6 viable for production boxes now?
Should I look at upgrading my servers to 1.6 or stay with 1.4.26.3?
Please weigh in and let me know what you think?
In conjunction with the above question is the thirdlane gui something I should be concerned about staying at a specific level or should I just do the check for update and update when it finds one?
I am curious what features
I am curious what features are enhanced with 1.6.2.X.
Is there something killer?
-Matt
some highlights 1.6.1 to
some highlights
1.6.1 to 1.6.2 changes
-
* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
the sound will be played to the target of an attended transfer
* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
finer control over how many peers Asterisk will qualify and the gap between them
when all peers need to be qualified at the same time.
* Added a function to remove SIP headers added in the dialplan before the
first INVITE is generated - SIPRemoveHeader()
* Channel variables set with setvar= in a device configuration is now
set both for inbound and outbound calls.
* Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
* Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
is used in conjunction with the 'faxdetect' configuration option. When
'faxbuffers' is used and fax tones are detected, the channel will dynamically
switch to the configured faxbuffers policy. For example, to use 6 buffers
and a 'full' buffer policy for a fax transmission, add:
faxbuffers=>6,full
The faxbuffers configuration will be in affect until the call is torn down.
* Voicemail now permits setting the emailsubject and emailbody per mailbox,
in addition to the setting in the "general" context.
* Added ConfBridge dialplan application which does conference bridges without
DAHDI. For information on its use, please see the output of
"core show application ConfBridge" from the CLI.
* ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
instead of the /var/run/asterisk.pid where it used to be. This will make
installs as non-root easier to manage.
-
1.6.0 to 1.6.1 changes
* multiple parking lots with BLF support
* Directory now permits both first and last names to be matched at the same
time. In addition, the number of digits to enter of the name can be set in
the arguments to Directory; previously, you could enter only 3, regardless
of how many names are in your company. For large companies, this should be
quite helpful.
* Added DNS manager support to registrations for peers referencing peer entries.
DNS manager runs in the background which allows DNS lookups to be run asynchronously
as well as periodically updating the IP address. These properties allow for
better performance as well as recovery in the event of an IP change.
* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
load/reload of large numbers of peers/users by ~40x (for large lists of peers).
These changes also provide performance improvements for call setup and tear down.
* Added ability to specify registration expiry time on a per registration basis in
the register line.
* 'sip show peers' and 'sip show users' display their entries sorted in
alphabetical order, as opposed to the order they were in, in the config
file or database.
* Videosupport now supports an additional option, "always", which always sets
up video RTP ports, even on clients that don't support it. This helps with
callfiles and certain transfers to ensure that if two video phones are
connected, they will always share video feeds.
* New CLI command, "config reload " which reloads any module that
references that particular configuration file. Also added "config list"
which shows which configuration files are in use.
* New CLI commands, "pri show version" and "ss7 show version" that will
display which version of libpri and libss7 are being used, respectively.
A new API call was added so trunk will now have to be compiled against
a versions of libpri and libss7 that have them or it will not know that
these libraries exist.
* Addresses managed by DNS manager now can check to see if there is a DNS
SRV record for a given domain and will use that hostname/port if present.
obviously the changes are quite alluring, it was some pretty serious stuff holding up its viability like breaking some deal-killers such as transfers, dtmf, and other problems along the way.
as of right now, 1.6.2.8 seems to have the most features working again.. however I cannot testify as to how other behaviors work on an inter-op (ie are there still DTMF problems with different carriers etc?)
I only have 1.6.x running in a test environment and its peering with a 1.4.x version of asterisk (its gateway) so I cannot say if you will experience problems talking to X brand of switch. The biggest hurdles are getting solved though.
1 exception..
FXS station ports still get 1way audio if call-waiting is enabled and the 2nd call tries to ring the phone.