similar to: Re: how to dial and accept a call with only

Dear Joe Dennick, and all group members, Could you give me more detail instruction to place a call from an outside line (or cell phone) through the * to voicemail or the demos to prove that the system works. Actually, I need to prove if system works or not. Then I can get money to do next step. Thank you a lot. Michael From: "Joe Dennick" <joe@dennick.net> To:

have you set up the db schema? and have you entered any sip data into the db? Sean -----Original Message----- From: Chandra [mailto:chandra@digital.com.np] Sent: Tue 1/6/2004 10:57 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] no results. i have been working with the retrieve_sip_conf_from_mysql.pl file and i have set everything as required. but when i

Hi, I've got a strange problem.. I've installed samba on a Sun machine running Solaris 2.5.1. Everything seems to work well, up until I try to mount my home directory from a Win95 PC. I've gone through the DIAGNOSIS.txt file, and everything up to and including stop 7 (smbclient '\\BIGSERVER\TMP') works great... so from what I can tell, its working. However, when I attempt

The hype and documentation for the last couple of releases of the Flash Operator Panel claim that the Panel can be configured to either change the LED for a phone, or the name of a phone to indicate when that phone is logged into a queue. I've tried on two different versions (0.18 and 0.19) on two different systems to get this feature to work, and have been completely unsuccessful. Any hints

Hello, Does anybody know of a place to get a half-height PCI slot bracket for the Digium T100P card? I have an application to fit them in 1U cases for small, deployable PBX's. The T100P itself is short enough to fit in a half-height slot but I need an appropriate mounting bracket to screw it into the case. Thanks, Mark Hagler

Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redhat Linux 9.0 on pc. Thanks in advance. Best, Michael --------------------------------- Do

I'm having a bit of a problem here: I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a x100p card for testing purposes. As a proof of concept, I wanted to be able to dial into the * using the isdn line, listen to a message, and enter a 3 digit extension number. If this happens, I wanted the * box to dial out using the x100p card, into our PBX (Nortel Meridian). If

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

I concur with Conrad. Cisco phones were retrofitted for SIP, whereas Snom phone are built around, and expressly for, the SIP standard. To be in compliance with Cisco regs, you are also supposed to have a SIP User license and a Smartnet contract for each phone if you abide by their program. I realize it is not difficult to obtain the SIP firmware, but if you were dealing with a Cisco authorized

We have a number of clients who will be needing a server to host Asterisk on. Many of these clients use analog (FXO) lines that will need to be connected to Asterisk via Sangoma cards. Can anyone recommend an industry-standard server (like IBM, Dell, HP, etc.) that has enough open PCI slots to handle up to six of the Sangoma cards? We would like to be able to tell the customer to just go

OK, I've got my script all set up and running, but now Asterisk crashes when the digits are entered with the following error: Ouch ... error while writing audio data: : Broken pipe I just retrieved and compiled the latest CVS this morning, as well as the latest AGI perl module. Why won't the AGI->get_data() function work correctly? Joe Richard Lyman <pchammer@dynx.net>

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com

With the 1U Apple G5 server at a good reliability/storage/pricepoint level... got to thinking about compiling Asterisk on OS X.. at least for SIP phone call switching, voicemail, etc. Has anybody attempted this? Email me off list if this is too dev-heavy for the user list. Thanks, Ted W -----Original Message----- From: asterisk-users-request@lists.digium.com

I've posted this problem a couple of times before with little or no response. Basically I have a T100P in my * box. Incoming calls are working great. However outgoing calls are not working at all. I've copied a previous post into this message which should have all the necessary info. Any ideas or suggestions would be greatly appreciated. Thanks. Mark

Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? Also, can the BRI interface cards participate in conference, etc., since they

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

> -----Original Message----- > From: Mark Charlton [mailto:asterisk@mcwebtree.com] > Plus if you send your users to VoicemailMain(${CALLERIDNUM}) > they don't hear > it at all. > They just get "enter password". Yup. If you do that, the only time they hear it is during the initial setup call (if you have "forcename=yes" or

Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without

I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong.