similar to: Meeting notice for the OpenSFS BWG group (new dial-in number)

Coming to Hamburg next week ? The European Open File System (EOFS) and Open Scalable FileSystem (OpenSFS) would love to see you at ISC?12 June 17?-21, 2012, in Hamburg, Germany. Visit our booth (#765) to meet Lustre experts, participate in informative talks, ask hard questions, and join the various events we are hosting. First off, there''s a full scheduled of Lustre talks, each with

Coming to Hamburg next week ? The European Open File System (EOFS) and Open Scalable FileSystem (OpenSFS) would love to see you at ISC?12 June 17?-21, 2012, in Hamburg, Germany. Visit our booth (#765) to meet Lustre experts, participate in informative talks, ask hard questions, and join the various events we are hosting. First off, there''s a full scheduled of Lustre talks, each with

Coming to Hamburg next week ? The European Open File System (EOFS) and Open Scalable FileSystem (OpenSFS) would love to see you at ISC?12 June 17?-21, 2012, in Hamburg, Germany. Visit our booth (#765) to meet Lustre experts, participate in informative talks, ask hard questions, and join the various events we are hosting. First off, there''s a full scheduled of Lustre talks, each with

Hi all. I have come to the conclusion that there just isn't anything out there for allowing SIP and NAT to work together nicely. This is rather amazing considering that as far back as March 2000 there are documents describing how to do it. So I've started a really simple SIP and RTP proxy project, SaRP, on sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. This is the

hi all the help output for DISA ends like below, with the half-sentence 'Note that in the case' what's the rest of that sentence? The file that contains the passcodes (if used) allows specification of either just a passcode (defaulting to the "disa" context, or passcode|context on each line of the file. The file may contain blank lines, or comments starting with

You could accept as the "passcode" the caller punching in their own phone#, then checking that against your whitelist. Lets associates get past the challenge when using someone else's phone, without their remembering some arbitrary passcode. And strangers or barred old associates who abuse it can get an earful about how you're suing them for wire fraud. Preferably after you

> Hello > I complied openssh like this "./configure --with-pam" and I did configure > /etc/pam.conf as follows > # PAM configuration > # > # Authentication management > # > sshd auth required /lib/security/pam_securid.so reserve > sftp auth required /lib/security/pam_securid.so reserve > # > login auth required

Greetings Community! Today we are very excited to announce that Xyratex has purchased Lustre? and its assets from Oracle. We intend for Lustre to remain an open-source, community-driven file system to be promoted by our community organizations. We undertook the acquisition because we realize its importance to the entire community and we want to help ensure that it will continue to deliver for all

Is it possible to implement in the Asterisk dialplan some way to authenticate a user with a dialed passcode which opens session that stays active enabling the user to make and receive calls, until the user logs off with another dialed passcode? I am aware of the Asterisk application 'Authenticate', but as far as I know, with this application the user meeds to dial his pin at each call he

Hello I am compiled openssh3.6.1p2 with PAM and using RSA Security (ACE) token. the command I used to compile ssh as follow: 1. ./configure --with-pam 2. make 3. make install After starting the sshd daemon, I authenticate using the command "ssh xxx.yyy.nih.gov" On the SecurID server I was watching the log monitor and I saw the following errors : "ACCESS DENIED, syntax

My only problem now is this: I have a broadvoice number that I can dial out on, and i ncomming calls word fine. Once the ip phone handset picks up, the audio bi-directionally is perfect. The problem is the audio BEFORE the handset picks up is silent on the broadvoice side. The user calling in over the sip number doesnt hear the menu, or even the ringing. Once the handset picks up then

The OpenSFS Technical Working Group would like to invite you to our next meeting to discuss current developments and our future course. When: Thursday, October 10th, 2013 12:30pm ET, 9:30am PT Dial in: US Toll-free: 866-692-4538 Local/Other US: 517-466-2084 Access code: 1973781 Other numbers may be available for those outside the US; please drop me a line and I''ll see what I can

Are you in Salt Lake City? We''ll be having an ''unofficial'' Birds of a Feather (BOF) session Wednesday evening the 14th, from 5:30pm to 7:30pm, at the Salt Lake Marriott Downtown City Creek 2nd floor Snowbird Rm. We''ve had a really busy year with some great progress around Lustre development in particular! Also some new important participants have joined. Lots

Hi, everyone, A quick reminder: This call will start in approximately four hours. See below for how to join. On our current agenda: Agenda * Full restrict patch * Observation and clarification * Issues encountered * Speed up in intersection code - O(nlog(n)) * Calling Verifier after LoopVectorize pass? -Hal Hal Finkel Lead, Compiler Technology and Programming

How do I make a user dial a passcode if he wants to make an international call? ________________________________ This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient,

Hi, I would like to do some development on the securid patch to get it to display the passcode prompt when users log in with securid on ssh. Can anyone tell me which scripts currently handle the logon/password prompt in openssh please? Thanks, JS. _________________________________________________________________ Join the world?s largest e-mail service with MSN Hotmail.

On 06/08/2015 09:11 PM, John R Pierce wrote: > On 6/8/2015 6:29 PM, Peter wrote: >> You can thank Fedora for making that rather pointless change and >> breaking that capability. > > that 'capability' was a holdover of the 1980s when disks were > measured in megabytes, and memory in kilobytes, so large file > systems were impractical. gee, you sure about that?

I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no

How do I make a user dial a passcode to make calls through asterisk? We would like to place a phone at a client's location for our employee but are afraid it may get abused by the other workers. ________________________________ This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from

I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551212 at sip.com,60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten =>