-dev: RPID Support – Please test

http://bugs.digium.com/view.php?id=2471

This works asterisk to asterisk now. The RPID is built and sent. You must
have trustrpid=yes and generaterpid=yes

This should allow you to properly send callerid over SIP between asterisk
servers. Please test and comment.

/b

[Asterisk-Dev] SIP presence notification updated (#3644)

After months in the bug tracker, we've finally committed a lot of
changes to the SIP Subscribe subsystem in Asterisk cvs head.

+ It now works even if you reload the dial plan
+ It does not accept subscriptions to extensions without hints
+ It will terminate subscriptions if the hint does not exist after a dialplan reload

To get this to work properly, you
- Configure incominglimit for the device
- Add a hint to the dialplan for the extension

Now, you will see if the device is online, if it is
occupied in a call or just available for a call. This is confirmed
to work with Polycom phones, SNOM phones and Eye-Beam.

A big thank you to everyone that have been assisting me working with
this patch and to Kevin that solved the reload problem.

/O

*UPDATED EXAMPLE*

Dialplan (extensions.conf)
* Add a hint that points to the phone or phones that you want to
connect to an extension

exten => 3000,hint,SIP/olle
exten => 3000,1,dial(SIP/olle,30)
exten => 3010,hint,SIP/ssokol&IAX2/ssokol
exten => 3010,1,dial(SIP/ssokol&IAX2/ssokol,30)

SIP, IAX2 and agents support this kind of notification, so you can
use them as hints.

In sip.conf or iax.conf, add an call limit to the peer entry

[olle]
type=friend
incominglimit=2
secret=1234
host=dynamic

(This config option will soon be renamed call-limit)

Then in your phone, add a subscription for the extension. How you do
that depends on the phone. In Eye-Beam you have a buddy list. On the
Polycom, you configure an entry in the directory to be a buddy.
We support several formats for notification, and they all have different
properties and abilities. For Eye-Beam, you will get an icon and text
saying "Not online" if the phone is not registred or "On the phone" if
the phone is in a call. Polycom simply says "away" and the Snom phones
happily turns on and off LED's.

A lot of the changes done internally to the PBX core extension state
subsystem and in chan_sip also improves the status notification in the
manager for programs that connect to Asterisk over AMI and visualizes
the status of extensions, like FOP, The Flash operator's panel.

I hope this helps.

Regards,
/Olle

New astGUIclient version released 1.1.6

Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.6

http://astguiclient.sf.net/

The client suite runs on Windows, UNIX and Mac, includes the
astGUIclient client-side web app which extends your phone's
functionality and the VICIDIAL client-side web app auto-dialer. This
package is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP,IAX or
Zap phones and Zaptel, IAX or SIP trunks.

For this revision, we have finished the VICIDIAL web-client, added
compatibility with the Asterisk 1.2 release tree, streamlined several
server-side apps and added cpu percentages to our stats logging
scripts.

As of this release, all client apps and daily administration functions
can be access through a web browser and we have tested our new
AJAX-enabled(PHP, Javascript and XMLHTTPRequest) VICIDIAL client in
production with great results.

Let me know what you think.

Thanks,

MATT---
http://astguiclient.blogspot.com

[Announce] Web-MeetMe v1.3.3

Work intrudes again and I will not be able to get to modifying the db
and gui to support per-conference flags as soon as I expected. So I have
released an update with what I do have available.

[Location]
http://www.fitawi.com/Asterisk

[Features]
1.  Schedule new conferences
	a. Control start and end times
	b. Set conference pin #
		i. Generate one if the requester leaves it blank
		ii. Identify pin # conflicts (another conference with
			the same pin is scheduled at the same time)
	c. Set Admin and User passwords
		i. Generate a user password if an Admin pw is set
			but the User pw is blank
2. Email the details for a successfully scheduled conference
3. Separate views for Current, Past and Future conferences
4. Ability to modify the end time of a running conference
	a. Can also reschedule a past or future conference.
5. Monitor realtime conference activity
	a. Mute/Kick participants
6. Optional authentication
	a. Currently Active Directory or LDAP based
	b. Authentication is abstracted so unix/PAM/DB/RADIUS
		support could be easily added (but outside of my
		interest to do so (patches welcome))
7. Users can only monitor, update or delete their conferences
8. Verified administrators can monitor, update or delete any conferences.
9. Updated to CVS-Head (a couple weeks ago, will target 1.2 soon)
	a. Changes to the Manager interface may have caused
		support for 1.0.X to slip, I cannot test that)

There is one functional issue to be addressed, and that is that
MeetMe tracks conference participants by channel. From a
conference management perspective it makes more sense to track
the participant by caller-id. I have a patch for 1.0.X on my
site, but have not polished one for CVS-Head or the 1.2.0beta
release.

Thanks and enjoy,
Dan

[Asterisk-Users] Asterisk + AstLinux testing images now available

Hello everyone,

A few days ago on *-dev I proposed the idea of making AstLinux images on a routine basis as a test platform for Asterisk. The ultimate goal is to have a web driven interface (accessible to the public) where users can download the latest and greatest versions of Asterisk (HEAD, STABLE, and recently 1.2.0-beta*), apply optional patches from Mantis (or elsewhere), compile third party apps, BRIStuff, etc. Possibly if only to see if it compiles, with the goal of creating AstLinux disk images and an ISO that can also be used as a live cd for testing or even (oh no!) production systems (don't do that)...

The ultimate goal here is to reduce the problems commonly associated with testing, and to get more people to test Asterisk. Just in time for 1.2!!! While I don't have the web interface, I have scripted my build system enough that it will (on a daily basis, or whenever):

- download and compile (hopefully) Asterisk, libpri, and zaptel
- create appropriate AstLinux images (net4801, geni586, live cd)
- upload them to my webserver for all to test and enjoy

For now, it seems like it would be quite simple to even interface to the asterisk-cvs mailing list and automatically checkout and build asterisk, etc. as soon as patches go in. (There would be a ~20 min. delay, but it would still be beneficial).

And now, onto the images:

http://build.astlinux.org

Here is the naming convention of the file names, with a sample:

AstLinux-08282005-CVS-HEAD-05-47-27-sc1100a.img.gz

This is AstLinux, with CVS HEAD checked out on August 08, 2005 at 5:47:27 (that's with seconds, in UTC), and it has been compiled for the SC1100 based single board family of computers (Soekris Net4801, PC Engines WRAP, RouterBoard, etc).

AstLinux-08282005-CVS-HEAD-06-40-57-geni586a.img.gz

This is practically the same thing, checked out almost an hour later, and it has been compiled for i586 and higher machines (generic PC hardware).

As I mentioned there is also an ISO with a very long filename that includes the above images, and can be used as a live cd for testing. Once I get the booting from USB in the 2.6 kernel working again, the i586 image will also boot quite nicely from a USB key/thumb drive.

P.S. - I probably won't turn on the "automatic build" until sometime next week, please test these images and make sure they work before I start cranking out images every 15 minutes when Kevin submits a patch! :)

--
Kristian Kielhofner

PA-168X IAX2 1.44 Firmware finally released

go here to download for your chipset: http://www.aredfox.com/edownloadsiax2.htm

The first beta of Asterisk 1.2.0 has been released!

The first beta of Asterisk 1.2.0 has been released! It is available from
the ftp.digium.com FTP servers, as well as the Digium CVS servers (under
the 'v1-2-0-beta1' tag).

This version of Asterisk represents a significant improvement in
features, stability and compatibility over the 1.0.x releases. Some of
the major new (or upgraded) features include:

- Asterisk Realtime Architecture
- Asterisk Manager Interface
- Asterisk Extension Language
- Dialplan functions
- More powerful dialplan expression parser
- Portability enhancements for FreeBSD, OpenBSD, Solaris and Mac OS X
- ... and many more!

We ask all interested community members to download and install the beta
release (on a non-production server) and report their findings via our
bug tracker. Please be sure to read the UPGRADE.txt file in the
distribution before upgrading your server, as there are a large number
of changes that you will need to be aware of (some of them are not
backwards compatible with the 1.0.x releases).

We want to extend our thanks to all the community members whose
contributions have made this release possible; without their support,
testing and other involvement we would not have reached this milestone
so soon!

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