similar to: Asterisk 1.2.0-rc1 Released!

The second 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-beta2' tag). This beta includes a large number of improvements over beta1, including: * Many, many bug fixes * Documentation and sample configuration updates * Vastly improved presence/subscription support in the SIP

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

The second release candidate 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-rc2' tag). This release consists of only bug fixes to the 1.2.0-rc1 version released earlier this week. We ask all interested community members to download and install the release candidate (on a non-production

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Everyone! As I am sure most of you have heard, Asterisk 1.0 was released last week during Astricon. There have already been some bug fixes, so Asterisk 1.0.1 is now available. Now that we have reached this milestone, there will be some changes in the release structure. There will be two branches in CVS: stable and development. There will be

Hello, We have a strange situation (asterisk 1.6.2.14), where we get a result for DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. This is the (relevant) test dialplan: -------------------------------- [incoming-private] exten => _X., n, Dial(SIP/1001,30) exten => _X., n, NoOp(${DIALSTATUS}) exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) [incoming-status] exten

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Pre release versions of CentOS 4 i386/ia64, have now been EOL''ed. Everyone using any of the following is requested to upgrade to the Final Release version. - - CentOS 4 i386 / Beta - - CentOS 4 i386 / Beta2 - - CentOS 4 i386 / Rc1 - - CentOS 4 ia64 / Beta - - CentOS 4 ia64 / Rc1 The repositories for these distributions have now been

The Asterisk development team has released versions 1.6.0-beta2 and and 1.4.18-rc1. The new beta for 1.6 is available for download from http://downloads.digium.com/. The release candidate for 1.4.18 is only available via svn. It is available for anyone that would like to help test 1.4.18 over the next couple of days before it gets officially released. To download the 1.4.18 release

The Asterisk development team has released versions 1.6.0-beta2 and and 1.4.18-rc1. The new beta for 1.6 is available for download from http://downloads.digium.com/. The release candidate for 1.4.18 is only available via svn. It is available for anyone that would like to help test 1.4.18 over the next couple of days before it gets officially released. To download the 1.4.18 release

Hi List, im running a test server with the 1.6.1-rc1-release of * and OpenAIS. Asterisk is configured so far and running stable. Now i set up a second server to test the distributed devstate. In a cluster on the same subnet it's no problem. But we have a customer who wants that feature for checking telephones on two branch offices connected over vpn-tunnels. According to that we have

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() {

We are proud to a announce another beta release of Samba 4.0, beta3 What's new in Samba 4.0 beta3 ============================= Samba 4.0 will be the next version of the Samba suite and incorporates all the technology found in both the Samba4 alpha series and the stable 3.x series. The primary additional features over Samba 3.6 are support for the Active Directory logon protocols used by

We are proud to announce that Asterisk 1.2.0 has been released! This release of Asterisk contains over 3,000 improvements on version 1.0, including hundreds of new features and applications. It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0' tag). We want to extend our thanks to all the community members whose contributions have

I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO

Hi! The Netfilter project proudly presents: conntrack-tools 1.2.0 This release is a major milestone that includes support for expectation synchronization and the new nfct utility that, by now, only supports the new cttimeout infrastructure. See ChangeLog that comes attached to this email for more details. You can download it from:

I have a caller screen queue setup. Basically a caller calls in, goes through a IVR, and before that caller is put into the queue, they get a sub ran on them first asking for them to say there name. That gets saved and they are entered into the queue using Queue(mainqueue,,,,300). In the queues.conf i have a list of members these are local/extension at external-default, there are two

For our internal fax machines, I'm checking if the faxes are going to branch offices. If they are, I want to capture and email them to the branches. I've set up extension 8447 to test this. A fax machines is connected via an SPA 2102 on 173. Any calls from 173 are sent to: [outbound-fax] exten => 8447,1,Answer() exten => 8447,n,GoSub(Capture-Fax,s,1) exten

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>