similar to: RSync for OpenVMS

I'm looking at need to duplicate some drives data from one OpenVMS site to another OpenVMS site. Clustering is not an option, so was wondering if Rsync could be used between two OpenVMS sites and if there would be any issues with the files. James Horn SHSU 2449 Computer Services Sam Houston State University Huntsville TX 77340 Phone: (936) 294-1042 -------------- next part -------------- An

Hi. I'm seeing some issues where Entourage clients can see new mail, download it (or partially), but won't display the mail in the mailbox list. Mail shows up as having arrived, but never displays. These clients have worked in the past with versions of 0.99, but not in 1.0-test-56. I have dumped the connection traffic and it looks like Entourage aborts the IMAP session before it

In article <Pine.LNX.4.64.0603211635320.7043@ab1-1-246.shsu.edu>, amdtech@shsu.edu says... > You have to set up a dialplan.xml file in your tftpboot directory for the > phone to pull: > > <DIALTEMPLATE> > <TEMPLATE MATCH="9,59....." Timeout="0"/> > <TEMPLATE MATCH="9,29....." Timeout="0"/> >

I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* > Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) > From: Aaron Daniel <amdtech@shsu.edu> > Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! > To: Asterisk Users Mailing List -

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

Has anyone successfully gotten two separate servers to handle voicemail on an NFS shared mount? The main thing I'm concerned about at this point is keeping both systems from writing the voicemail file to the same filename... any thoughts? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I know I compiled AEL2 into 1.2 before, considering I just copied my source from one server to another, yet I can't seem to figure out why I'm getting this error. Anyone have any ideas? make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' flex argdesc.l "argdesc.l", line 19: unrecognized %option: reentrant "argdesc.l", line 20: unrecognized

Of the people in here that have hinting working with the polycom 601's (or any phone for that matter)... do you have it working so that the shared line appearance shows that there's someone on the phone? If so, any hints on how to do it? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Just doing some test installs of asterisk running on branch (noticed first on branch), and noticed if you move to virtual terminal 9 (may be different on everyone else's), the CLI is running. Anyone have any idea how to turn this off? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We

I know you can set up monitoring of queued calls, and I'm pretty sure my question's been answered before, but has anyone devised of a way to actually barge into a queue channel so you can do in place monitoring of calls? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Darren, > -----Original Message----- > From: Darren Tucker [mailto:dtucker at zip.com.au] > Sent: Thursday, June 17, 2004 11:08 PM > To: Scott Rankin > Subject: Re: SSH_MSG_USERAUTH_PASSWD_CHANGEREQ and 3.1.0 > F-SECURE SSH - Pr oces s Software SSH for OpenVMS > > > Scott Rankin wrote: > >>That will depend on which versions exhibit the problems. Is it >

How does the hinting work on the polycoms? I've got a polycom set up with hinting, I can see when the shared line rings, but I can't tell if someone's on the line. Any suggestions? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Does anyone have the pickup application working? I'm attempting to get it so that a particular extension programmed into a phone can be picked up by another phone with that extension programmed with a speed dial with a 'p' in front... basically, if you dial p100 and extension 100 is ringing, it'll pick up that extension, otherwise it dials the number. The problem I'm having is

Hello, I searched through the mailing list archives here, at securityfocus.com, at HP and google and have come up dry. Sorry in advance if my search was not complete enough or I missed something but I sure as heck didn't see anything. ENV: Slackware 10 w/ ssh (SSH-2.0-OpenSSH_3.8.1p1) connecting to an alpha with OpenVMS 7.3-2 w/ sshd (Remote protocol version 2.0, remote software version

On Thu, 2020-06-11 at 17:02 -0600, Swift Griggs via samba wrote: > Refer to this: > https://legacyos.org/vsi-introduces-new-samba-release/ > > I noticed that VSI, the new owners of VMS, have created a package > for > Samba4 under OpenVMS. I also notice the samba-vms list appears to be > abandoned (for years). So, I apologize for posting here on the > general > list,

Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it

On Fri, 12 Jun 2020, Andrew Bartlett via samba wrote: > I can't find any evidence in the source code for new patches to have > Samba build on VMS Okay, well, at least I didn't just overlook something. > it certainly wasn't something we actively targeted when we moved from > autoconf to WAF with Samba 4.1. I'd be shocked if you did. Python 3.x isn't available on

Has anyone successfully gotten an SCCP cisco phone to register to two different asterisk servers at the same time for redundancy? I can't seem to get the phone to recognize the second server in the callmanagergroup... Aaron -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198