similar to: Strange Problem

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

I need to redirect the sound from a remote Centos 5.2 computer to my local Centos 5.2 computer. Both are i386 OS. Searching the web and Centos web site has indicated that it is possible but I have not found any information about how to do it. I am currently using ssh and/or vnc to display the remote computer locally. At this point, the sound is being played on the remote computer only.

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

Not sure yet what or where the problems are but having just done a jum update on my HP laptop (nw8240) and my IBM desktop from 2.6.9.42.0.2.EL the laptop is locking up during boot sequence and the desktop when running VMware Workstation seems to take all CPU and makes strange noises from the loudspeaker. Using grub to fall back to last kernel all is ok again !!. Ian -------------- next part

Hi all. I have two Rails applications. Each is on its own VPS hosted by Pipespring (excellent service btw). My app runs lightning quick - AFTER the first load. If I visit my site after a period of inactivity (i.e. no visitors to my site), it takes up to 10 seconds to load that first time. After that I can hop around with no problems. Has anyone run into this before? Ideas? - Rabbit

This is probably more a Ruby question, but somebody here probably knows ;). Is there some handy Ruby or Rails library that can get data from other sites (like Amazon and Flickr) via REST, then return it in nice arrays, hashes, objects, etc.? I''ve searched quite a bit and haven''t been able to really find anything. There is a ruby-amazon library, but it''s for an older

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):

>From the docs: "Note that any fields in the join table will override matching field names in the two joined tables. As a consequence, having an "id" field in the join table usually has the undesirable result of clobbering the "id" fields in either of the other two tables." I do have an ID field in the join table, and as a result it''s clobbering the ID

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I

I''m getting this error in Rails: undefined method `include'' for #<MenusController:0x408efd1c> The offending code is: require "rexml/document" include REXML I see there may be a problem with GCC 4 on OSX, but I''m using Redhat with GCC 3.2. http://weblog.rubyonrails.org/articles/2005/12/22/is-gcc-4-0-incompatible-with-ruby-on-os-x-and-elsewhere

* Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > I use kmix as mixer, but there is nothing in it about a > > "capture channel". How could I find where it is defined ? > then try alsamixer or amixer and watch out for capture and unmute and apply > this for the according channel (mic, line-in, pcm or master) Hum... the problem is that RH 9 uses

Hello out there! Nooby question: What''s the "Rails" way to work with tables of relatively static lookup data? Objectively (I''m thinking); I don''t want to hit the DB each time, so for each lookup table, initialize a globally available hash with lable/value pairs... Am I on the right track? How would you do this kind of thing in Rails? Thanks! --

Hi all, I have just purchased the DevKit from Digium and received a X100P and a TDM400P (it has one FXS module). The problem is that I can't get the kernel module (wcfxs) to load and run. I have searched the archives and can't find anything about this. Do the messages below ring any bells with anyone ? There is no interrupt clash (see pci list below). I am running on a AMD 1.2G

Hallo, I'd like to get all channels of an ac3 5.1 stream coded into vorbis. With sox and oggenc it doesn't seem to work. Does anyone know a tool that can do this under linux? The ac3 stream is also avialabe as wav6 file converted by the cvs version of a52dec. This wav6 file contains one stereo stream and 3 mono streams. Is there a convention which stream number in an ogg file means

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

>> Actually, computing the "power spectrum" for each frame of W shows >> how large an ammount of the original signal at time offset j the >> echo canceller thinks should be removed from the current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

Hello Mark, >> What is known about Nvidia plans - do they plan to develop a dom0-only, >> or maybe even a domU drivers as well? > dom0 3D drivers are certainly easier (these should also work in a domU that > has access to PCI graphics card, once PCI-passthrough is working again - > someone will have to figure out how to make X behave correctly, tho). Ooh, that sounds