search for: cliffed

I''m trying to get Puppet to run on ESX v3.0.1. Being that ESX doesn''t come with ruby, I installed v1.8.6 under /opt/ruby/ruby-1.8.6 and linked the bin to /usr/bin. facter installed and runs without issue. However, when I try to install puppet, I get: - Could not load openssl; cannot install Is this due to the way I installed ruby or something else? Thanks, Clif

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

Hi I have noticed that some versions of syslinux later than 1.66 fail to install a working boot block on a CF card in Soekris net4521 board. The Bios is: comBIOS ver. 1.05 20020419 Copyright (C) 2000-2001 Soekris Engineering. The card is a 64M Sandisk and I am useing a version of LEAF called WISP-dist but I also experianced the same problem with the Bering distro. Here is my syslinux.cfg:

Hi, Thought I have been using Linux for a few years, use wine almost every day to run Corel apps on my laptop, and can get small win apps to load, I have not had any sucess running Filemaker. If it is possable- I would very much love to demo the new version of ebase running with wine on Linux at the August Filemaker Confrence. Is there anyone who monitors this group who has sucessfuly run

I''m getting the following error on the client: /usr/lib64/site_ruby/1.8/puppet/util.rb:300: warning: fork terminates thread at /usr/lib64/site_ruby/1.8/puppet/external/event-loop/event-loop.rb :123 when using the following code: case $operatingsystem { CentOS: { service { "nrpe": ensure => true, enable => true,

I have a friend with a PRI coming into a modem bank that is receiving 56K modem calls and some ISDN data calls. He wants to dump his analog office phone lines and use some of the capacity on the PRI. I have been digging through the mail archives and Wiki site on this subject but the information I found doesn't give me a high enough confidence to go buy a few T1 cards and try it out.

Hello all, I have 1.0 Alpha 4. Is it better to shut down sendmail, uninstall dovecot 1.0a4 then install 1.0a5, or just install 1.0a5 over the top of 1.0a4? Thanks, Cliff

I have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

I'm running 3.4.6 (was running 3.0.28a but upgraded in hopes to fix this issue). Clients running Windows 7 that are NOT joined to the AD domain (samba authenticates against it via "security = server") cannot authenticate to access the server. Clients running Windows 7 that are on the domain as well as Windows XP, Windows 2003 on and off the domain work as expected. Any help would

I am currently helping a friend build an Asterisk PBX that spans several cities using anything from T1s to DSL connections to link remote SIP phones, IAX gateways, etc. to a central Asterisk PBX server that serves up voicemail, features, etc. The biggest problem that I have had with this system appears to be the leading problem that my day job company finds with their VOIP deployments: Most

What is the difference between the version 1 and 2 voicemail and IAX? I see different examples using both so I want to know why would one use a particular version. Thanks!

I have finally crashed Asterisk for the first time and I'm wondering if anyone has seen this. This is a configuration with SIP endpoints and an IAX2 channel to another Asterisk PBX. The main PBX dropped a core file after a SEGV (signal 11 ) with the following trace: #0 0x42079133 in strchr () from /lib/tls/libc.so.6 #1 0x41bb0f9c in _fini () from /usr/lib/asterisk/modules/chan_sip.so #2

I have had several cases where the message waiting indicator was stuck in the on state with Cisco 7960 SIP phones. Here are the two cases: 1. Single extension that mapped to a single voice mailbox. Restarting Asterisk or getting a new voicemail then clearing it fixed the problem. 2. Three SIP extensions that mapped to a single voice mailbox. Getting a new voicemail and then clearing it

I have tried to get my TDM400P card to automatically dial a number or run an application when I pick up the phone without much luck. After reviewing the email archives, config files and source to chan_zap.c it appeared that all I had to do was set "immediate=yes" in the zapata.conf file and have a default number in the TDM400P's context (ex. s,1,Directory(default)). So far I have

Has anyone experienced any date/time problems on Asterisk? I have noticed it on earlier builds and I believe on this build: CVS-02/16/04-23:57:06 Specifically, I have seen the following problems: 1. Change system time using the "date" command while Asterisk is running. Asterisk does not synchronize with new system time. 2. Daylight savings time goes on/off at the designated

Good People, I found a very helpful document in your archives and decided to return the favor by returning it, with my notes included. It's in StarOffice7 format but I don't expect you guys to have trouble opening it :~) Please feel free to edit it, where you think appropriate, before publishing to the internet. As noted in the first page, I have now successfully integrated Solaris 10_x86

Should I be concerned of the following error? - Value of ''preferred_serialization_format'' (pson) is invalid for report, using default (b64_zlib_yaml) I''m running 0.25.4 on both server and clients and all are running Red Hat 5.4. Thanks, Clif -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this

I am wanting to perform toll bypass using multiple gateways for outgoing calls. For example, if I call from location A to location B and I have a gateway in location B I obviously want to use location B's gateway to make it a local call. I understand how to get the local prefixes from NANPA but my question is: If there are additional numbers that are local (non-toll) calls from a particular

Howdy, I've just successfully configured a win2k system to print on a raw queue, but apparenty not through samba. I'm courious as to why that didn't work, and would like to help update the documentation to cover this in more detail, if only I understood it better. I started by following the directions here under "Raw Print Serving Vendor Drivers on Windows Clients":

I am trying to music on hold but I am having all sorts of problems with it. I am running RH9 and the latest version of Asterisk as of yesterday. Here is what I did to test it: 1. I manually deleted the mpg123 softlink to mpg321. 2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and installed the the archive and loaded ztdummy.o module. 3. I threw a couple extra files in the mohmp3 directory