similar to: Asterisk on OS X

Hello, I am looking for a VOIP Service Provider to work with getting started with Asterisk. Does anyone have any brief recommendations? Ease of use and support are the key criteria. ----------------------------------------------------------------- Ed Mansouri Ucompass - http://www.ucompass.com Make sure we stay connected to you Add yourself to the Ucompass Address Book

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

I have a couple of questions I'm hoping folks can help me out with. When I search the mailing list, I see folks doing what I'm interested in so here's hoping ! - How are people making out with interfacing to the 7960? I'm considering buying a number of these as they look quite feature rich. But, are they easy to interface to? - Will I be able to interface with softkeys on

A customer contacted us today concerning getting a VoIP to PSTN system with a few IP Phones setup. Asterisk should fit his needs. It is not a big job, but I think that this customer is going to need onsite work. Please contact me off list if you are an interested reseller in the Washington, DC area. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389

Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L

I'm having problems with a new install of Asterisk (I had to reinstall because hard drive failed). I've used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for

Is there is an open source ACD component for Asterisk? Likewise, is there an open source CTI component that will work with Asterisk? Regards, Jim O'Brien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/e6ee98da/attachment.htm

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Here are a few things I would like to see .. 1. In addition to time/date stamps, store/read the caller id info with the voicemail messages. 2. Have the ability to configure the system to ignore and delete messages left by a caller that are 3 seconds or less (maybe make this configurable) Not sure but that would cut alot of hangup calls out of your voicemail box. I can't think of much more

thank you Martin , but the problem now is that when I run " R CMD INSTALL -c SJava.tar .gz" I have the message " perl n'est pas reconnue en tant que command interne ou externe" [[alternative HTML version deleted]]

I understand that softphone are the answer in fact I deploy a ton of the Ip comm version every week. I am under contract with the phones so I can't sell them and there no easy way out of the contract. As for 79XX's I have several office that have them working over a VPN backed in to our main office where the CCM's and GW's are with managable problem and for the most part they

That is not working... I tried like you mentioned it and even a few different ways and will not create the file at all.... Am I doing something wrong... -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Todd Lieberman Posted At: Sunday, May 09, 2004 4:00 PM Posted To: Asterisk User Group Conversation:

Bonjour, Je developpe une application java et je veux integrer du code R dans mon programme . Le programme de test que j’ai fait est : import org.omegahat.R.Java.*; public class REvalSample { public static void main(String [] args) { String [] rargs = {"--slave", "--vanilla"}; System.out.println("Sample program to call R engine from

Hi, When I use the analog phone connected to Zap/1 how do I transfer & hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? Cheers Jay

How do I stop asterisk when it is in a bad mood? It keeps dialing extensions and won't listen! I tried kill <PID>. No go. I don't want to have to reboot again. Thanks. Jim Friedeck P.S. I love it when my boss looks over my shoulder and I don't have an answer when he says: 'So, what are you doing?'

Hello I m tryng to execute the code below and I have a problem with the R-Java connection.: import org.omegahat.R.Java.*; public class REvalSample { public static void main(String [] args) { String [] rargs = {"--slave", "--vanilla"}; System.out.println("Sample program to call R engine from Java"); ROmegahatInterpreter interp = new

Hello Everybody, Is there a function available for calculating the density of a non-central t distribution? (dt does not accept the ncp option). Thanks in advance! -- Wolfgang Viechtbauer "Are you not thinking what I am not thinking?" -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read

Hi, Is it possible to stop users from having the option to change their voicemail greetings ? I'm running asterisk on an old machine with a small hard drive and I don't want users to be able to take up room on it by recording their own messages. Also, once someone has recorded their own messages, cant it be set back to the default one ? Once final question, is it possible to see how

Anyone successfully implemented a solution for allowing ZapBarge call monitoring only for a specific group of agents calls? The issue I see is that the feature only works on zap channels, and all of the agents (in many cases) are IP phones. Allowing ZapBarge and ZapScan on the TDM PSTN (t100p) interface has privacy issues for senior managers, but would allow all outbound zap calls to be