similar to: picking a channel bank

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

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

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 * 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

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

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

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

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

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

Just starting my search for a SIP/MGCP/H323 channel bank. I need analog ports in a building that only have network connections back to my * server. I could install another * server and use a normal CB with a PRI but I would like to investigate any good CB's with network trunk abilities Thanks Dave Packham

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

Can someone recommend a good quality 24 or greater port channel bank? Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070506/e5896021/attachment.htm

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

Hi, Here's the second attempt of a patch to get the impulse response from the echo canceller :) -------------- next part -------------- diff -ubBwr clean/include/speex/speex_echo.h get_impulse/include/speex/speex_echo.h --- clean/include/speex/speex_echo.h 2007-10-09 13:08:15.000000000 +0200 +++ get_impulse/include/speex/speex_echo.h 2008-02-12 23:58:11.000000000 +0100 @@ -51,6 +51,14 @@

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hello, I don't have any experience with channel banks and would appreciate any feedback on my theory outlined below: We have a single T1 entering the building with channels 1-12 being voice lines and 13-24 being a 768k internet connection. This T1 terminates to an Adit 600 (T1-1). Here's what I know. Channels 11-12 go out the Adit 600's 25-pair connector to a wiring block (and

> > <with Prof. Farnsworth voice> "Good News, everyone". > > I've just kicked off project "Sphinx". Which is supposed to > sound like "Speex" merged with "INT". ;) Meaning I am working > on an integer encoder and decoder. > Great. I looked into converting speex to fixed point a while ago, but my job has gotten much busier

Jon, I actually had some more discussions with Tim on this issue, and it seems that the channel bank would still be a good option to choose for internal purposes. I would not see any other solution than a channel bank to connect many 2wire phones into one asterisk box. I had a talk today with Carrier Access, and it seems that the adit would do us fine. The fxs cards of the adit 600 are