similar to: Problems with XEN-modified Kernel an SDLT 320 / System freeze

This update to bind fixes the problem where bind is stopped following an update. https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=143786 refers. Changed files are : updates/ia64/RPMS/bind-9.2.4-7_EL3.ia64.rpm updates/ia64/RPMS/bind-chroot-9.2.4-7_EL3.ia64.rpm updates/ia64/RPMS/bind-devel-9.2.4-7_EL3.ia64.rpm updates/ia64/RPMS/bind-libs-9.2.4-7_EL3.ia64.rpm

I'm not sure if this has been brought up before, but there is error with the XFree86-devel package on CentOS3. Both gccmakedep and xft-config are broken symlinks to archexec in the folder /usr/X11R6/bin. Is there another RPM that provides archexec? This is not a problem on CentOS4. It does look like RHEL suffers from the same problem. -- Regards, John Quaglieri InterServer, Inc

On Wed, 2006-12-06 at 23:51 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Wed, 06 Dec 2006 22:37:01 -0500 > From: Steve Prior <sprior@geekster.com> > Subject: Re: [asterisk-users] Is there any Asterisk controllable > thermostat? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >

Hello, Your email has been received. During the dates of 2/13/02, 2/14/02 2/15/02 and 2/16/02 we will be going through a major internal upgrade/server transition. This will not affect our hosting serversl. If you have an issue that needs solved immediately you can call 412.654.6500, or between the hours of 9 and 5 Monday thru Friday, you can contact Kenneth Munz @ 412.381.3410. Also, you can

Hi All, We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing. Problem Statement: How do we integrate the 2 systems such that Audio SIP calls are seamlessly passed between the two. Sorry we're just starting up so a bit of general advice, or a link to any document would be great! If anybody has done this -

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs to pass THROUGH Asterisk, and are NOT nated" What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call

hi all, i want to use a compaq sdlt 320 under centos. the tape drive was already installed and after rebooting i was not able to see it(dmesg | grep tape or dmesg | grep scsi). i add the correct module (modprobe st) but i'm not able to use with mt. the st0 file is missing, can someone help to resolve this issue. thanks ?

Hello, I have a weird problem. If I use winamp to stream to my icecast server, I can give it any password (even the incorrect one) and it logs in fine and sends the stream to the server. If I use Ices, if I use the password supplied in the icecast.conf file and give it to the ices configuration file, it kicks this: Failed connecting to server 127.0.0.1, error: Could not login on server.

We have here a TandBerg "videoconferencing system" connected in asterisk with a Beronet card BN4S0 (4 BRI ports). I`m trying to make a videoconferece (video + audio) with this Tandberg to another Tandberg using the ISDN channels through the BN4S0 BRI Card. But, i'm only obtaining audio calls, the video not appears. When i try to make a 64K calls (data calls), the call does not

Hi All, I have a CentOS5.2 server using a Qlogic QLE220 HBA, connected to Quantum DX30 and M1800 libraries via a Qlogic SANbox 5600 switch. I've installed the latest (v8.02.14) drivers from Qlogic for the HBA The issue I have is I am missing 2 of the 4 SDLT drives provided by the M1800 Library. The DX30 provides 6 drives, which are listed in dmesg and assigned /dev/st0->5. I would

Hi, For some time now (starting with the 3.10 or 3.11 kernels, but I'm not sure) I have the following problem: I have a machine with 6 SATA slots and two SAS controllers, one onboard HBA and one RAID controller in a PCIe slot. The problem is that the order of the SAS controllers changes randomly after a reboot, so the SCSI addresses of the devices change. One time lsscsi prints (empty

Friends, SIPit is the main interoperability event for all things SIP. It's organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants - videocaps, Marc Blanchet's IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all. I have 15 or so CentOS servers (now v4.4, I will upgrade them all to 5 when it comes out) and a few windows machines. I am looking to implement a network backup solution using Amanda. I will be backing up 200 gig or so to start, but that will grow. Does anyone have any recommendations on tape drives that will work "out of the box" with CentOS? Thanks, Joe

I have been trying to use Freelance Graphics under Wine for a long time (since Oct 2000 :), but I cannot seem to get the display stuff working correctly. I always get text splattered over top of other text when trying to view existing .PRZ files. This happens in both Freelance and in Lotus Notes (when Notes is trying to display a file attachment of the .prz persuasion). A quick Google search

I received this email link from Macromedia today http://www.macromedia.com/newsletters/edge/february2005/index.html?secti onIndex=3&trackingid=AWQX They are now selling Breeze video conferencing server by the minute as an asp, this service is powered by the macromedia communications server which is just a sip server like asterisk but for video. I'm still offering a $2,000 bounty

Hi, I'm trying to configure Asterisk as SIP-H.323 Gateway via ooh323, but I have an error relatively to the GK Confirmation message. >From the log: "H323 RAS channel creation - succesful Sent GRQ message Gatekeeper Confirmed (GCF) message received ERROR:No Gatekeeper ID present in received GKconfirmed message Ignoring message and will retransmit GRQ after timeout Error: Failed to

For the last 10 years we have had some Tandberg video conferencing units spread across our WAN. To make a long story short we had to work with another entity to allow access into those units with H.323. Calls from public networks have never worked to those devices with private IP's. Over the summer, my coworker and I built an * VOIP system for one of our high schools. The need came up for