similar to: Chameleon Mail

On Fri, May 23, 2008 at 01:25:43PM -0400, Donny Kavanagh wrote: > This is getting downright abusive, and is totally uncalled for, this > is not a list for personal attacks. You thought that Steve suggesting JT step in was abusive? If that's not what you meant, then you need to either a) be clearer, or b) reply to the proper message. And hackers ignoring pleasantries to get right down

Ok, did some experiments. Here's the scoop. You will need a live USB key of CentOS 6 with a persistence layer (overlay) and the EPEL gdisk package installed to make this thing boot. On a system with Chameleon already installed, boot the CentOS 6 install media. Installing Chameleon without OSX is an adventure that I've not done, so a 'testing' OSX install (10.6) with the

Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Hello again, I'm not attempting an install of Samba 3.0.4 from source. I want to specifically compile in ads and winbindd support. So, I already compiled and installed OpenLDAP 2.2.13. Now the configure script is hung on krb5 dependancy: checking for krb5.h... no configure: error: Active Directory cannot be supported without krb5.h So I downloaded and attempted compile of krb5 1.3.4, but

so I want to install c6.2 x86_64 onto a 2.7TB /dev/sda ... its a virgin machine with no software, using pxe boot. disk druid or whatever seems to only want to let me have like 2tb of default stuff, I'm guessing because its not using GPT? do I need to preboot into a shell or something and use parted before I can install ? -- john r pierce N 37, W 122 santa

> On 13 Jul 2017, at 01:55, Johnny Hughes <johnny at centos.org> wrote: > > On 07/12/2017 02:02 PM, Pierre Riteau wrote: >> Hello, >> >> The version of cloud-init packaged in CentOS 7 Extras is rather old (0.7.5, which was released in 2014). It is missing some functionality that we would like to use, as well as important bug fixes. >> Before I tackle

Hey all, I just tried running a 'make rpm' on a fresh install of Fedora Core 4 and ran into an error near the end of the build process. This is the output of the build when the error occurs: done rm -f /tmp/asterisk/var/lib/asterisk/mohmp3/sample-hold.mp3 mkdir -p /tmp/asterisk/var/spool/asterisk/voicemail/default/1234/INBOX :>

Hello listmates, Here's I am, trying to install Centos 6.2 64-bit on a server with 24 TB of disk RAIDed as RAID 6. So I get this 18 TB disk (the RAID is recognized as one disk) where I am trying to put my OS. And I can create a couple of partitions - let us say I defiine a 150 GB swap, a 150 GB "/", a 100 GB "/var". So far so good. Then I try to define the next one, say, a

Hello, The version of cloud-init packaged in CentOS 7 Extras is rather old (0.7.5, which was released in 2014). It is missing some functionality that we would like to use, as well as important bug fixes. Before I tackle rebuilding the package myself, I would like to know if there is a plan to update extras to a newer cloud-init version, or if has anyone has already updated cloud-init and would be

I am using the following macro to dial a ZAP channel. When I dial in, * answers and I go to voicemail. I never hear any ringing, though. It doesn't work with the Ringing command before or after the Dial command. [macro-zapdial] ; ; call a ZAP extension for ${ARG2} seconds, and then voice mail ; ${ARG1} - Extension ; ${ARG2} - Time to ring exten => s,1,Dial(ZAP/${ARG1},${ARG2}) exten

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

I have installed my first Asterisk implementation using the Asterisk@home ISO. I am using the SJPhone software. Using the setup page, I have been able to configure two extensions. Whne I dial from one to the other, the other does not answer even though it is registered. Watching the log in the CLI, I can see that recorded messages are being played;: == No one is available to answer at this time

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

All, Please help me defend Samba. We are having major performance problems when using VisualStudio to build projects. The source lives on the server, and the intermediate files are dumped locally to try to speed things up. One of the engineers downloaded a demo of NetManage/Chameleon NFS and it was significantly faster. Here are some numbers: samba-1.9.18p8 samba-1.9.18p8

Greetings, In reviewing the changelogs it?s unclear in which release FLAC began supporting a sample rate of 192kHz. The reason for my question is that there are many forums and university studies that state that FLAC does not support a sample rate of 192kHz however the current documentation (assumed 1.2.1b) under FORMAT under FRAME_HEADER does note that it is supported. If it was not

Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a

Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing

Dear community, I'm using gnustep runtime -fobjc-runtime=gnustep with gnustep-libobjc2 (https://github.com/gnustep/libobjc2) and Cocotron/Chameleon. For following source file #import <Foundation/Foundation.h> #import <UIKit/UIKit.h> int main(void) {         NSString *str = [NSString stringWithCString:"TEST"];

Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :-

Hi, I am currently trying to benchmark a multithreaded xapian implementation on a chameleon baremetal instance written in C++. My workload is a 3 Gig wikipedia xml dump consisting of ~286 file of different sizes. My results are showing me that indexing on xapian is an order of magnitude faster than my lucene and lucene plusplus implementations. This is a result that I did not expect. Just want to