similar to: newbie question; can * screen calls?

I am just studying Asterisk now and have a question. Is it possible to force anyone who enters a queue into voice mail after they have been in the queue for 30 seconds? /************************************************************** Ken Alker ken@impulse.net ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo,

I just registered ID 0004283 at http://bugs.digium.com for the problem described in subject (found when using a Linksys PAP2-NA). I don't know where the proper forum is to discuss, so I'm hoping anyone interested will read the bug and let me know your thoughts, either at bugs.digium.com, here, or by emailing me directly (or, please suggest another forum that is more appropriate). As

I am considering switching to Asterisk for my ISP. I currently use a NorTel ICS with a NAM II (standard voicemail and Minuet ACD software). I have a technical support center staffed by a max of 5 agents at one time. I am wondering if it is possible to do these things with a queue: 1) When a customer enters a queue, they are only left in the queue for x seconds (say, 30 seconds) while waiting

I have never had to pick out a channel bank before but I'd like to use one with the Digium T-1 card to hook 8 analog CO lines to an * PBX. Is there a reference somewhere describing and comparing channel banks (old and new)? Can modern channel banks handle translating all the "new" analog signaling features into a T-1 format? For example, can it interpret the 1200 baud FSK

I have * working on my Sony Vaio PCG-FX120 laptop. I am trying to get * to recognize my internal PCI Intel modem as an FXO port. I have modified wcfxo.c in order to identify the PCI modem properly. Based on output from dmesg, wcfxo didn't recognize the modem until I inserted the proper vendor and device IDs into wcfxo.c and re-compiled. Note that the error message from "modprobe

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may

I've got a Wildcard T100P along with a Zhone Zplex 10 24S/O which has been working fine for me now for a while. These have been pulled out of a working Asterisk installation (as they were no longer required) to use at home only to find that the fan noise is too loud. As such I'm looking to sell off this hardware and replace it with some combination of fanless hardware that will allow me to

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106.xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the "common" features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put

Martin Pycko <martinp@digium.com> writes: > sure, use the 'n' option of the queue and put voicemail app as the next > priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

Based on several threads I've read on this list, I assume that it would be handy to supply POE (power over ethernet) in an environment without having to purchase POE switches (assumed expensive) and abandon one's existing (familiar/custom/not-yet-expensed/etc.) switches/hubs. Assume I have a non-POE switch with 24 RJ-45 (ethernet) ports. I design a 1U box that can be mounted just

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

I have been testing Gastman and Astman with SIP calls. As I have no Zap phones, so I have a few question on what is normal behavior? When a call comes in and I have created extensions for all phones (example: Channel = "SIP\3846") Should the little lines connect between the pre-made extension or should they pop up temporary icons with no connection to the hand made extensions? The

So I tried to run Impulse on my Ubuntu 11.04 laptop and it installed fine (or at least seemed to). When I tried to run it I got the working cursor for a bit then nothing. I ran it in terminal and got a error message about running the Windows version of Mono so I went online and did a search for the error message and found a file called mono-2.4.2.3-gtksharp-2.12.9-win32-3.exe and ran it. The error

Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a

Dear colleagues/contributors, I'd be pleased if someone could provide insights on how to plot impulse response functions in a format that can easily be copied in a word document just as plotting time-series of variables. I had followed the outline suggested by Benhard Pfaff [see http://127.0.0.1:17693/library/vars/html/irf.html] but I am unable to get the impulse response functions in a

Hi I have a asterisk voip box connected to a classic pbx. The pbx use telecom back impulse (bad translation ?) for billing. To have all my billing done by the pbx I need to send back impulse to pbx from asterisk. Is it possible to simulate telecom back impulse with asterisk ? Thanks for your help. Jerome D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos