similar to: Star Wars Echo Sound

Hey list, arriving home from SambaXP, it was really nice. That darth vader cake was super sweet :-) Hearing some people talking about how much they'd like bash completion for `samba-tool`, I've put together an ugly script that does that. It's a very dumb script (I'm boldly calling it script), but it might make someones day a little bit easier.

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

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone

I've searched both the archives and dejanews, to no avail. I'm trying to get a hp 4mp (PostScript) print out sane output. I've got a old machine running Red Hat Linux, hooked up to the 4MP. The printcap prints up postscript just fine. I make the printer a share, tell it that the printer is postscript, then go to the win98 box, choose the HP 4MP driver and try printing and get

I have looked a bit through some of the archives for this list, and I have caught different ideas on this matter.. I've also read through the HOWTOs on how to do this stuff, but I haven't received the same results that I was expecting, so I need someone to help me out a bit, give me a definite YES or NO and if YES, how? I have the latest release candidate of samba 3 installed on redhat-9.

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys

Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660]

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

I've been tooling with this for a while, and I need some help... please!! :) Here's what I'm trying to do. I want a samba server to work with winbind, so that when I type 'getent passwd' it shows a list of local users, as well as my domain user list. I want a user to be able to ssh into the machine using their NT 4 domain username, like domain\username. That's pretty

Could anyone give me some hint on how to get winbindd working ? I have added the following to smb.conf , called smbpasswd to join the domain and got security=domain working. security = domain encrypt passwords = yes workgroup = MYDOMAIN password server = x.x.x.x But when I try to use winbindd, even if I followed the man page instruction and did the following: added the following to smb.conf:

I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Hello All, Hoping all of you might have an additional option for me to try at this point. :) My Goal: To have a paging option that does the following.... When I press **_XXXX it will send a ring-answer page to that person. The person on the other end should be muted, so if they are in a conference, you can't hear what is going on in the meeting. If that person hears me and decides they want

Hoping someone out there has run into this or has some ideas for us. We currently have asterisk set up with Realtime (using mysql) for its extensions,sip and voicemail files. The problem we are trying to solve, is one of a failover mechanism. What if our mysql server went down. Can Realtime be set up with a secondary mysql server to get its data from. We can set up mysql to sync with its fellow

After installation of Star Wars Empire at War and it's Expansion with the latest version of Wine, the game will not start up. I click on the icon and a blank text box shows up instead of the start of the game. I have the latest versions of Wine, Ubuntu and I just got winetricks. I have tried changing the Library with all of the items there. I have tried adding the .exe to Applications,

Hi, Everyone... I love WINE but am getting a bit light headed... While attempting to install SW: EAW and Supreme Commander (Gold) both installs stop to request a second disk (though SC is a single DVD) and both ask for the same "data3.cab" file. SW:EAW is a CD and requests that the second disk be inserted. I get that. What I don't understand is the path to the disk. WINE

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060