similar to: Encryption for Windows 2000 beta

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Hello, we are using hardware array and its vendor recommends the following setting in /etc/system: set sd:sd_max_throttle = <value> or set ssd:ssd_max_throttle = <value> Is it possible to monitor *somehow* whether the variable becomes sort of bottleneck ? Or how its value influences io traffic ? Regards przemol

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, Is it at all possible to have the "maxmessage" setting on per user/mailbox value? We have a requirement whereby we want the global maxmessage setting to be 180 seconds per mail box, however, we would like to have certain users to be able to store longer voice mail messages. Is this at all doable in asterisk? Thanks Bruce

Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23

Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or

Hi All, I read rumors of a potential Asterisk 1.4.12 release for last week. I would like to start testing Asterisk 1.4 on Solaris, but, the fix for the segfault in res_features is only in the current development trunk. I would much rather test a release version, and as such, need to wait for the release of 1.4.12, my question is, do we have a guestimate on when it will be released, 1 week, 2

I dual boot WinXP and Fedora Linux and would like to run Agent 3.3 when I boot to Fedora. Has anyone successfully installed WINE (codeweavers?) on Linux and gotten the current Agent to function? What has your experience been? What problems have you encountered? I've seen very little discussion in the newsgroup. The web page that Mark Prince pointed to a few years ago is no longer active:

Hi All, I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial

How's wine support coming with translucent windows? Guild wars 2 has a new launcher that makes extensive use of transparency to mimic the in-game interface, but of course, it turns into a massive black square here. Screenshot of the launcher, all that is black is supposed to be transparent: http://i.imgur.com/BqNo7.jpg I know it's nitpicky but google implied some progress on this and I

I'm trying to configure a Solaris 8 system to authenticate Samba against Windows 2003 ADS. I've compiled the appropriate packages; however, I'm quickly stuck trying to get my kerberos ticket. Here's the error: sumac:/opt/local/kerberos5/bin# ./kinit admin@DCRI.DUKE.NET kinit(v5): Cannot resolve network address for KDC in requested realm while getting initial credentials

Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce

Hello all, Does anybody know if is it possible to install dahdi on solaris 10? I've only found a zaptel modified code for solaris at solarisvoip site. I'd appreciate any comment or experience about asterisk + dahdi/zaptel on solaris.. Best regards, Caio

# Your mailer is set to "none" (default on Windows), # hence we cannot send the bug report directly from R. # Please copy the bug report (after finishing it) to # your favorite email program and send it to # # r-bugs@r-project.org # ###################################################### Today I have installed R 1.02.1 on my MSW-95 laptop; it essentially worked, but thre are two

Can anyone help? I am trying to install R-0.62.2 on a Sun with OS 5.6 (Solaris 2.6), and gcc/f77. Unfortunatley the simple commands ./configure --prefix=/opt/R make did not work, complaining when building help/docs. It claims to have problems perl (which has worked fine so far) but in practice it is not only a documentation problem: if I force "make install" and run R, then it

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps

Here is the structure of my servers : Source : /home/test1/link_dir sub_dir1 file1 file2 ... sub_dir2 ... Destination: /home/test2/real_dir sub_dir1 file1 file2 ... subdir2

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

Hey guys and gals, I have the following object that has acts_as_versioned: class Note < ActiveRecord::Base acts_as_versioned belongs_to :user end The schema for my notes table is as follows: create_table :notes, :force => true do |t| t.column :id, :integer t.column :noteshare_id, :integer t.column :user_id, :integer t.column :title, :string