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I'm running RH7.1 and Samba 2.2.2 . I would like to offer a VFAT partition on the server hard drive for universal read-write access. I mount the partition in fstab with /dev.hda5 /sys vfat defaults,rw,uid=1001,gid=1001 0 0 It mounts fine and shows up on the Windows 98 machine and can be opened. I can't write to it however. User and group 1001 are smbuser account, and they

...under SysV conventions, with the section numbers of course also being changed in the body of the manual pages themselves. As I'm not a registered member of the list, please respond directly to me as well as to the list -- thanks in advance for any enlightenment! Thanks and Cheerio, Rick Rodgers (rodgers at nlm.nih.gov) -- -------------------------------------------------------------------------------- R. P. C. Rodgers, M.D. * rodgers at nlm.nih.gov * (301)496-9305 (voice, fax) CSB, LHNCBC, U.S. National Library of Medicine, NIH Bldg 38A, Rm. 9S-916, 8600 Rockville Pike, Bethesda MD 2089...

-------------- next part -------------- An embedded message was scrubbed... From: Steve Rodgers <hwstar at rodgers.sdcoxmail.com> Subject: Via Mini-itx anaconda install failure Date: Sat, 1 May 2004 14:31:43 -0700 Size: 849 URL: <http://lists.centos.org/pipermail/centos/attachments/20040501/e69b2948/attachment-0005.mht>

...Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing What have you set the PSTN Dialing Delay: on the PSTN Line tab (logged in as admin advanced) ? Mine is set to 1 and it works well. Chris ----- Original Message ----- From: "Anthony Rodgers" <Anthony_Rodgers@dnv.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, February 08, 2006 9:50 PM Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delaybetweenanswering and ringing >...

...connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion." Does anyone know if * can provide the "network signaling" required? If so, how? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp

...e-Teod Bercea <Gheorghe-Teod.Bercea@ibm.com></font><br><font size=1 color=#5f5f5f face="sans-serif">Cc: </font><font size=1 face="sans-serif">Alexey Bataev <a.bataev@outlook.com>, Guray Ozen <gozen@nvidia.com>, "Gregory.Rodgers@amd.com" <Gregory.Rodgers@amd.com>, "Finkel, Hal J." <hfinkel@anl.gov>, "kli@ca.ibm.com" <kli@ca.ibm.com>, "openmp-dev@lists.llvm.org" <openmp-dev@lists.llvm.org>, LLVM-Dev <llvm-dev@lists.llvm.org>, "cfe-dev@lists.llvm.org"...

...and amd; not contain “gpu” because offloading may extend to non-gpu archs; avoid “arch” by itself so as not to be confused with first field of a triple; and not collide with existing flags to allow both options. "--target-subarchs" satisfies all the above. Comments? Greg Rodgers

...our phones with more than one registration, the MWI goes out periodically, and then relights a short time later with the characteristic Polycom warble tone. Any ideas what could be wrong here? I have a hunch that it has to do with registration expiry, but I'm not sure. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp

...h of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

...banner. Is there a mechanism in open ssh, when using tcp wrappers, to support a banner? Thanks in advance for any helpful insights. We would be happy to share our installation instructions for both systems and welcome comments about the most efficient way in which we might do so. Cheerio, Rick Rodgers

I figured out what was going on with the lack of/stuck on stuttered dial tone. Apparently, there are two voicemail directories being referenced: /var/spool/asterisk/voicemail/default, and /var/spool/asterisk/voicemail/local. The sip phones were using /var/spool/asterisk/voicemail/local to dump VM messages into, yet the MWI looks at /var/spool/asterisk/voicemail/default. Does anyone know why

What Linux distribution is best for use with Asterisk? (easiest compile, least problems, etc) Thanks, Sean Rodger

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

...r limit on directory entries is? The size of the XML file itself is only 60K - you'd think that would all fit into the phone with no problems..... I would appreciate a cc: on any replies to the list, as I don't always get to read it in a timely fashion..... thanks! Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp

...ing: checking configure summary... ERROR: No locking available. Running Samba would be unsafe configure: error: summary failure. Aborting config What exactly does "No locking available" mean, and is there a workaround? I'm currently running version 1.9.15p8 on the same OS. TIA! Ed Rodgers edr@dtmr.com

Hi, I'm trying to track down why I'm getting calls dropped on an infrequent basis between two asterisk servers which are at the same physical location and connected to each other with UTP ethernet. Here is the connection diagram Asterisk Server 1 ===UTPENET== Switch ====UTPENET==== Asterisk Server 2 I see sporadic RX and TX frame retries when I enable iax2 debugging on either box.

I am running ntpd on the same machine as asterisk in order for the grandstream phones to display the time. After a while the time display fails until the phone is re-booted. Has anyone run into this problem before? Is it simply a bug in the GS firmware? Sean

Trying to figure out if a bug report should be submitted. Can anyone on 1.2.x verify of this has been corrected? I am on CVS 8/2005 If a call comes in to an extension that dials more than one channel (rings at more than one phone) both calls in the CDR show a status of answered when only one is answered, the source channel is bridged to only one of the two destination channels, but both CDRs

Hello, I'm using the latest SAMBA-Version I have problems in a SAMBA - WINDOWS NT Configuration. Maybe you can help me : I used the diagnosis.txt to test my configuration. Everything works until Point 8,9,10. I get from my windows-NT machine : system-error 5 system-error 1240 Can't login from this konto ?! Yours sincerely R. Brand