similar to: New sounds also now in CVS

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,

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

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'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

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

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

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

Does * have the capability to screen calls? IOW, if someone calls in from outside (ie. not a local extension), can * ask the calling party to state their name, record it, ring the recipient, play the caller's name for the recipient, then give the recipient the choice of answering or forcing the call to voice mail? /************************************************************** Ken

--On Monday, January 19, 2004 11:01 AM -0500 Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: <SNIP'd from the "ADSI phone vs. IP phone" thread> > I'm looking at ADSI phones simply because I don't have to re-tool my > entire building; I can use the existing phone network and (I think) get > all the functionality I need with the (far) cheaper

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

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

>> Generate a test signal (10+x sine waves per frame), where x increases by >> one for each iteration, and wraps around at 100. > > Testing with sine waves is usually not a good idea. If you intend on > cancelling speech, then test with speech. Ok, I tested more extensively with both music and two-way speech. More on this below. >> However, when peeking at the

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Gary Franczyk > Sent: Thursday, January 15, 2004 10:37 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question > > > Whaaa?? So, to allow 24+ lines of dial in access, how

Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten => TestTrakt,1,Dial(ZAP/1-2/517255333) exten => TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [TestTrakt at

Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier.

Hello, I'm new on this list. First of all, thank you to all the developers of this great project! At the moment there is only an application that keeps me on both macos and windows, its name is PWGL a free environment for computer assisted composition in openGL. (http://www2.siba.fi/PWGL/) I'm running Ubuntu 8.04 and wine 0.9.59. I have to say that I also installed vcrun2005 and

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

I'm a bit displeased at the way this happened. I received an email from VoicePulse. Here's some excerpts: ------------------ >We're sending you this important update so you can take advantage of improvements we've >been making to your VoicePulse Connect! service. >We've been working hard on improving the audio quality and reliability of your Connect! >service,