similar to: I am away on 22 Feb (pm)

I have consolidated the OpenSSH patches for version 3.5p1 to incorporate the following : 1. Include Chroot ability 2. Fix the "change password upon expiry" problems for AIX and Solaris You can get the patches from http://wave.prohosting.com/~hstoh/ Regards, Hwee Siong. ******************************* ** Please support OpenSource ** *******************************

Happy coming Spring, Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Friday, February 22 at 12:00 PM (Eastern US) 9AM PST, 5PM GMT * Call (724) 444-7444 or sip:123 at 66.212.134.192 After the call connects, enter the conf: 22622# and your_PIN# (or 1# if you have no PIN)

Leap year? Election year? Will your GoToIfTime() dialplan function properly on Feb 29th? Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Aastra has some interesting phones so we asked them to come talk about them. Friday, February 29 at 12:00 PM (Eastern US) 9AM PST, 5PM GMT * Call (724)

Hi, Tomorrow's guest is Mike Seto from Polycom who will be "grilled" on application development for their phones. Coincidentally, there is a contest for the best microbrowser application. Anyone can join in on this: http://tr.im/WinPolycom Join us in the usual places: IRC #voip-users-conference Call Info: http://www.VoipUsersConference.org SIP URI: 7463#22622#PIN at

Hi, Few subjects cause as many arguments as "which SIP client works best?" on IRC #asterisk, voip forums, and probably the -users mailing list. I have tried most of the SIP clients available in the last 5 years, both with Asterisk and other platforms such as OnSIP.com, IConnectHere.com, ZipDX.com and the venerable old FWD (in the days when that almost worked). Speaking of ZipDX, we

** Great talks this meeting: (live & on video) ** Ekiga3, Asterisk, GPGPU, GStreamer, Debian Edu, HowTo Present KDE at meetings http://sites.google.com/site/berkeleytip/ Join from anywhere via VOIP conference, with the friendly, educational, productive, BerkeleyTIP people. :) Join the #berkeleytip freenode.net IRC channel for help getting your VOIP working.

You can listen or join the Asterisk Users Conference Fridays at 12:30 PM EDT Today's subject suggestions: FAX capabilities, what's your solution? Multiple asterisk server implimentation: ENUM, DUNDI or even two servers connected Your subjects? Share your ideas, ask your questions! See http://x2z.eu for instructions on how to join or listen

> You can listen or join the Asterisk Users Conference Fridays at 12:30 PM > EDT > > Today's subject suggestions: > > FAX capabilities, what's your solution? > Multiple asterisk server implimentation: ENUM, DUNDI or even two servers > connected > Your subjects? > > Share your ideas, ask your questions! > > See http://x2z.eu for instructions on

Hey folks, Here's your chance to report in about Astricon, ask or answer general asterisk questions, talk about your asterisk-related (or voip-related) projects, sites, work, anything. We interested and listening. We have a great core group on these conferences, even though Indiana is disproportionately represented for some reason :) This conference is NOT limited to developers or gurus,

Has anyone noticed a problem with runaway mpg123 processes for music-on-hold eating up ~100% CPU and driving the load on the machine way up? I've seen this problem consistently with multiple Asterisk installs, 1.2.x and 1.4.x, although admittedly it was more common with 1.2.x as far as I can tell. There is no clearly identifiable sequence of events that causes this to occur, although it

On Wed, Feb 5, 2020 at 3:00 AM Gerd Hoffmann <kraxel at redhat.com> wrote: > > Introduce new virtio_gpu_object_shmem_init() helper function which will > create the virtio_gpu_mem_entry array, containing the backing storage > information for the host. For the most path this just moves code from > virtio_gpu_object_attach(). > > Signed-off-by: Gerd Hoffmann <kraxel at

Hi, I've just observed something a bit odd - I'm wondering if this is the expected behaviour, a bug/feature, or something I'm doing stupid! 1st person gets into MeetMe. Nothing fancy, just: exten => 987,1,MeetMe(400,iM) They enter the passcode and their name, then listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH

Join your colleagues for the 20th highly acclaimed international forum to hear the latest implementations of biometrics... The Winter 2010 BIOMETRICS SUMMIT: -------------------- Practical Implementation Strategies, Market Trends And Best Practices In Government And Business -------------------- February 22-25, 2010 - Miami, FL

I will be out of the office starting 24/12/2002 and will not return until 06/01/2003. You can contact me on 0407 756 568. <p><p><p><p>------------------------------------------------------------------------------------------------------------------ This e-mail and any attachments may be confidential. You must not disclose or use the information contained in this e-mail

Hi All... I set up a conference bridge using MeetMe. It works nicely, except that it seems that certain parameters I give it are ignored or else don't work. Here is the line from my dial plan: exten => 6500,1,absolutetimeout,0 exten => 6500,2,MeetMe,100|ciMpPs|1234 The MOH and * work, but users are not announced when they join or leave and the pin is not requested. Maybe I am

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

Hi, we have an Debian Etch 4.0 amd64 server with 2 B410P cards. Asterisk SVN r99777 is installed. We tried with mISDN shipped with Asterisk/Zaptel (make b410p) as well as with the latest version from mISDN.org 1.1.7.2. zaptel, ztdummy and crt-ccitt modules are loaded. Output of /dev/zap is: root at SrvPhone:~# ls -al /dev/zap total 0 drwxr-xr-x 2 root root 120 2008-01-23 16:42 .

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

In order for suspend/resume reprobing to work, we need to be able to perform sideband communications during suspend/resume, along with runtime PM suspend/resume. In order to do so, we also need to make sure that nouveau doesn't bother grabbing a runtime PM reference to do so, since otherwise we'll start deadlocking runtime PM again. Note that we weren't able to do this before, because

My code is like this: <%= select_time () %> 1) I would like to pass more options so that the a blank value is at the top. I tried ":include_blank => true" but it didn''t work. 2) I would like to limit the number of hours to only upto 12. With the current code the hours has value rangin from ''00'' to ''23'' 3) Would like to have