similar to: CDR datasets

There seems to be something about SL for queues since when the show queues CLI command is used, it give something like "SL:0.0% within 0s": pbx*CLI> show queues 1 has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242 (dynamic) has taken no calls yet SIP/1251 (dynamic) has taken 4 calls

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

Hi list, I have a client with a strange requirement: putting a noise gate on the Asterisk channel. For those who are not familiar with them, noise gates are used in musical instruments to avoid entering low-level noise into the amp system. What they basically do is, they measure the volume of the channel, and when it's too low they just let the channel close, i.e send perfect silence,

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

Hello Everyone, I checked with zttool that sometimes after the machine boots the order of the boards is changed like this: ??? Alarms Span ??? OK Digium Wildcard TE110P T1/E1 Card 0 ??? OK Digium Wildcard TE110P

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

I'm looking to bock a list of numbers users cant call. Is it possible to pull these from file specified in the dial plan, as apposed to mysql?

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards

Hi, I'm wondering what I need to change to get the "swedish" type ring on a SIP-trunk. When I make an inbound call i still have the "US"-type of ring on my SIP trunks. I need help on changing this. However I've successfully changed this on the Zap interface for all inbound calls. Thanks in advance! Regards, Jan

Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop?

Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application:

hello my friends, when i make a genzaptelconf i get this message ******************** CAS signalling on span 2 conflicts with HDLC with FCS check on channel ******************* Any idea Please? I m installing zaptel 1.4 i checked in "http://bugs.digium.com/view.php?id=7860" that it's a bug but beacause i m a newbie in asterisk i can't undrestand what exactly mean Thank You

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid

Hello list, I have prepared a couple of new tutorials you may find interesting: - Installing an Asterisk 1.4 beta system - at http://astrecipes.net/?n=216 - Installing the Digium's Asterisk GUI for 1.4 - at http://astrecipes.net/?n=217 It's nothing too complex, but you may find them interesting, especially the new Asterisk GUI. Any comment is welcome - the site is a wiki, so feel

Hello list, I am trying to set a custom SIP header on all calls that are made by the app queue because I want to track a certain state at the SIP level. If I use the following code: exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID}) exten => s,n,Queue(myQueue) this works fine for the FIRST call made from the queue to an agent; but if that call does not go through, it's not repeated

Hello there! I've problem with setting up unicall / mfcR2. can't find proper notation for channel, trying unicall/1, unicall/1/1001, unicall/g1, unicall/g1/1000 and still having no luck. klaudia*CLI> !cp call /var/spool/asterisk/outgoing -- Attempting call on Unicall/1001 for application Dial(363) (Retry 1) Mar 25 09:29:34 NOTICE[19920]: channel.c:2429 __ast_request_and_dial:

I have a user who is off my system with IAX. When he calls and goes out my long distance provider my asterisk switch seems to be bridging the two calls. As a result I loose all accounting information. All I get is the call setup time (15 or 20 seconds). How can I either make asterisk not bridge the call, or keep correct tabs on the call accounting for me? Mar 23 21:55:24 VERBOSE[18185]