similar to: Configuring A@H with Analog Phones

All, I am trying to install puppet master version 2.7.13 on Red Hat Enterprise Linux 6, and utilize stored configs. I followed the guide here: http://projects.puppetlabs.com/projects/1/wiki/Using_Stored_Configuration When I run puppet --noop on one of the clients, I get the following error: err: Could not retrieve catalog from remote server: Error 400 on SERVER: Could not autoload

Hi guys I am currently getting the following in my log asterisk1*CLI> Parsing '/etc/asterisk/manager.conf': Found asterisk1*CLI> == Parsing '/etc/asterisk/manager_custom.conf': Found asterisk1*CLI> == Connect attempt from '127.0.0.1' unable to authenticate Can anyone tell me why asterisk would not be able to authenticate it's self?

Hi All, Hope you all are fine and good, Today i have found that Mine all PRI Channels are restating after every interval of one hour, and i have search and psot on fourms and everyone said that this is a normal behaviour. If this is a normal behaviour is there is any way to stop it { i still don't know what is the reson to restart ever hour } . Because this is listed everywhere that

I'm having a problem getting 3 way calling to work correctly using two outside lines and one extension. The two outside lines are connected to the X100P's and a standard model 2500 phone is connected to the TDM10. When I dial the first outside destination 9xxxxxxx, the call completes correctly. When I flash the hook switch and dial the second location 9yyyyyyy. The call doesn't

For some reason, I can't get call pickup to work between Sip phones or between Sip and Zap phones. All phones are in the same call group and pickup group (1). The source code was downloaded and built as of today 11/15/03. Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=aliens ; ; SIP Entry for sipura line 1 ; This

>rom: "Jose Hernandez" <jfhernandez_01@verizon.net> >Subject: RE: [Asterisk-Users] Unable to open master device > '/dev/zap/ctl' > >If you are using udev you will get a message during compile suggesting to >read README.udev, I did not pay attention to this message. > >Anyways.. I did not get the error after creating a udev rules file in

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

I''ve seen a lot of issue regarding the stability of Rails apps. I''m charged with investigation of Rails for my company and I''ve looked at numerous fourms, groups, etc. (Textdrive, here, etc.) and it *seems* like there is a stability problem with Rails (ie: crashes, etc.) Is this as common as it looks, or is this tied to things like Lighttpd (web server) or Typo

When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507)

Hello, I am experiencing a number of 'Time moved backwards errors' such as: Mar 27 11:38:20 host-78-129-239-60 dovecot: imap-login: Time just moved backwards by 729 seconds. This might cause a lot of problems, so I'll just kill myself now. http://wiki.dovecot.org/TimeMovedBackwards Mar 27 15:20:10 host-78-129-239-60 dovecot: Time just moved backwards by 4214 seconds. This might cause

Hello, I noticed all my DC's have port 5353 closed. I'm using the internal DNS and wasn't sure if multicast DNS must be enabled? I do not appear to be having any DNS issues. My only concern is with the wiki on Multicast DNS. "By default, mDNS only and exclusively resolves host names ending with the |.local| top-level domain (TLD). This can cause problems if that domain

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Dear List, Apologies for this off-topic post but it is R-related in the sense that I am trying to understand what R is telling me with the data to hand. ROC curves have recently been used to determine a dissimilarity threshold for identifying whether two samples are from the same "type" or not. Given the bashing that ROC curves get whenever anyone asks about them on this list (and

I'm trying to use the randomTF function from package palaeoSig to test the significance of a MAT reconstruction with nine analogs and a WA-PLS reconstruction with four components. I'm probably missing something obvious here but how do I make sure that randomTF is testing the reconstruction based on the desired number of analogs / components? In: fitmap.wapls = WAPLS( lumapspc,

I've got Samba 2.2.2 and winbind up and running fine between a TurboLinux server and an NT 4.0 PDC. I can "chown" files on the linux box fine using "$chown DOMAIN+username filename" but when i try to change an objects group using "$chgrp DOMAIN+groupname filename" i get errors stating that it is an invalid group name. What's the deal here? None of my group

Ok, I have a People table, a Shows table, a Techs table, and a Castmembers table. The people table has contact info about each person, the techs and castmembers tables have information about the persons specific role in that sshow. i want to call @show.techs (or somethign equivelent, I don''t want to start that code yet) and get all the techs associated with the show, and the

I have a relationship where a Show has_many Techs through People. From what I understand that requires me to put belongs_to Show in the People model. My problem is that a Tech can belong to many Shows. Is there any way of solving this besides putting a has_many Shows relationship in People -- Posted via http://www.ruby-forum.com/.

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the

Hi, I am tring to us ip route to load balance between two interfaces. ip route add equalize 10.200.1.0/24 nexthop via 10.200.0.2 dev neta nexthop via 10.200.0.2 dev neta2 Where neta and neta2 are gre tunnels. Testing show that packets travel in a single sided manner. Do I need to use the multipath (IP_ROUTE_MULTIPATH_CACHED) module? thx jason