similar to: Agent Status on FOP

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

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:

sounds like you need to do some reading at the many fine resources available. start at http://www.voip-info.org. Here's a hint for you though.... exten => s,1,Answer exten => s,2,VoicemailMain Barring that, just run 'make samples' which will create a wonderful set of sample config files which will allow you to test the system out pretty thoroughly.... Sean -----Original

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?

> -----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 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

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.

Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they don't have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have just upgraded my Cisco 7960 phone to SIP firmware today and I have to say it's working great with Asterisk. At work (which uses Cisco Call Manager), when a voicemail is recieved the read light remains lit until the voicemail is retrieved. Is there any way to achieve the same effect with Asterisk ? Thanks, Paul. -------------- next part -------------- An HTML attachment was

Hi All, Whenever i try to start FOP using script ./op_panel_redhat.sh start given in directory /usr/local/op_panel-snapshot/init I got the following error: Starting Flash Operator Panel: execvp: No such file or directory [FAILED] Please let me know the reason for this. Thanks in Advance With Regards, newbie

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

I am trying to figure out which one of our agents is answering the calls. According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff & logon. There is the connect & completeagent message, but it doesn't show which channel (agent) answered the phone. I can't even figure it our cross referencing the

I've been continuing to experiment with MySQL. I'm having absolutely no luck getting asterisk to read voicemail configuration data and mailbox configuration data from mysql tables instead of from voicemail.conf. The default Asterisk setup that reads from voicemail.conf and extensions.conf works fine. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox Enterprise Linux box.

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'm fairly new to Asterisk, but I've been searching the archives extensively and haven't seen much information on using IVR for more than menus. I'd like to prompt a caller to enter his ID (employee, customer, account, etc). The business use I have in mind requires a five-digit ID. Then I need to be able to capture the ID entered, validate it (probably by playing it back with

Hi, Has anyone managed to get X-Lite to work with Asterisk using the iLBC codec.. I have just tried updating the the latest version 1079 (BTW this new version supports up to 10 proxy configurations, Not that I can see a reason to have 10 proxies setup, I would rather have the ability to transfer calls).. I can make a call using iLBC but the sound that I hear is just a lot of pop's and

I've tried to do it myself, but my head is now bleeding from hitting it on the wall so much. We need someone who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline

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