The Asterisk Development Team has announced the fourth release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently
scheduled to become the full release of Asterisk 1.8.0.

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.

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

With the availability of the Asterisk 1.8.0 release candidates, the binary
add-on modules for Asterisk produced by Digium have been updated with new
versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of
situations.

This release candidate contains fixes since the last release candidate as
reported by the community. A sampling of the changes in this release candidate
include:

* Additional fixups in chan_gtalk that allow outbound calls to both Google
Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
and stunaddr.
(Closes issue #13971. Patched by dvossel)

* Resolve manager crash issue.
(Closes issue #17994. Reported by vrban. Patchd by dvossel)

* Documentation updates for sample configuration files.
(Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)

* Resolve issue where faxdetect would only detect the first fax call in
chan_dahdi.
(Closes issue #18116. Reported by seandarcy. Patched by rmudgett)

* Resolve issue where a channel that is setup and torn down *very* quickly may
not have the right call disposition or ${DIALSTATUS}.
(Closes issue #16946. Reported by davidw. Review
https://reviewboard.asterisk.org/r/740/)

* Set TCLASS field of IPv6 header when SIP QoS options are set.
(Closes issue #18099. Reported by jamesnet. Patched by dvossel)

* Resolve issue where Asterisk could crash on shutdown when using SRTP.
(Closes issue #18085. Reported by st. Patched by twilson)

* Fix issue where peers host port would be lost on a SIP reload.
(Closes issue #18135. Reported, tested by lmadsen. Patched by 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=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-rc4

Thank you for your continued support of Asterisk!

We have a small office installation running over a cable modem. (8M down, 500k up confirmed with numerous speed test sites)

When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We’re using ulaw, and confirmed that traffic with 2 calls is <175kbps in/out. (IAX connection out)

Asterisk doesn’t report any dropped frames, the internet connection looks fine, etc. We have a linux router in place running wondershaper that seems to be running fine (same as our other installations).

Can someone suggest where to look? Could this be the ITSP?

We have a queue that agents log into through the dial plan. Extension
Sip/101 logs in as Agent/101

We have ‘ringinuse = no’ in the queues.conf file.

The issue is that when Ext 101 is on a ‘non queue’ call (they placed a
call, someone called their DID, etc) they still receive queue calls.

Is there a way to stop this from happening?

-Matt

Hi,I cannot get this to work..I have two application maps that call these two macro’s…transfer is done on sip phone and transfer2 is done on the incoming dahdi line….thats all working….but the value stored in dtmf12 is never passed into the second macro so I get ” ” in the NoOp..

So how exactly can I do this…global variable,setGlobalVar,import etc..tried a few combos but they dont seem to work as I think they should?

this is in etensions_custom.. have no Global variable in this config..

[macro-transfer]
;performed on callerexten => s,1,Read(dtmf12,,5,,2,10)exten => s,n,NoOp(${dtmf12} [macro-transfer2]
;performed on callee
exten => s,1,Flash()exten => s,n,NoOp(${dtmf12} exten => s,n,SendDTMF($dtmf12})exten => s,n,Hangup
ThanksJames

Hi!

I’ve configured LDAP to read both users and extensions from LDAP server.
However, I’m experiencing problems with state tracking. Previously when
using static files, I was able to map extension number with channel
state using:

[sip_phones]
….
exten => 100,hint,SIP/user
exten => user,hint,SIP/user
..
rest of the dialplan
…

Thus when someone called the user, hint SIP/user showed channel state as
BUSY and I was able to use call limits etc. Now I’ve added this line to
[sip_phones]:

switch => Realtime/@

My hints, and call limits as well, stopped working. I’ve tried to move
hints to LDAP (which would be ideal situation for me), setting
AstPriority to “hint” but I don’t think they are event fetched. So the
question is… I’m I doing something wrong or it’s just impossible to
use those two solutions (hints + LDAP) together?

PS. I’m using Asterisk 1.6.2 if it helps with anything.

smime.p7s