similar to: Disabling zap echo cancellor from dialplan

what i want to achieve is simple, i want to be able to sitback in my chair and talk over a voip call using speex. as the mic and the speakers are built into the laptop, an echo canceller is required. what i am doing is this : First, each time i finish writing a block to the waveout, i copy it into a buffer like this: Segment *pseg = (Segment *)phead; waveOutUnprepareHeader(waveOut, phead,

I am trying to build 1.2beta1. I downloaded the sources from http://downloads.us.xiph.org/releases/speex/speex-1.2beta1.tar.gz and tried compiling it under visual studio 2005. it gave the following errors: LIB : fatal error LNK1181: cannot open input file '.\debug_rtl_dll\medfilter.obj' Build log was saved at

jean - there is a new header for echo cancellation in 1.1.7. can you give me any pointers about playing with it? i didn't see any sample code that uses it. should the speex manual be updated to reflect this and the preprocessor API? is there anything i can do to help with the documentation? this week, i shall submit the tested binaries for ARM with 1.1.7. - farhan

I'm using a Cologne chip card in my Asterisk box with zapHFC drivers (bristuff-0.0.2). The system works well, but this way I'm not able to run newer version of Asterisk. Do you think it's better to use i4l support and newer version of Asterisk or keep the bristuff with older asterisk ?? Have anyone tried chan_mISDN on a 2.6 box ? How does it run ???

Is it possible to build a dialplan in which shifting from daytime to nightime is not hour based but phone driven ???

I'm not sure that this problem is strictly related to zaphfc, but this is what happens: my asterisk (build on bri-stuff-0.1.0-RC2k) handles a single PCI HFC-S based card. I own a Cisco 7940 Sip phone (fw 7.1) and a pc running X-Lite. Zaptel.conf and zapata.conf are taken directly from zaphfc samples. Extension.conf contains the following lines: [from-ISDN1] exten=>s,1,Wait(1)

I've a problem connecting uniVoice (http://voice.uni.it) from asterisk. Using my account data I can place a call smoothly using xlite or my budgetone phone directly, but I'm not able to use uniVoice as a peer from asterisk. Registration seems to work correctly, but when I try do dial, the sip authentication fails every time. Their tech people told me that they are unable to make

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

Hi, I found this message concerning VAD and was wondering whether VAD has been ported to fixed-point in the latest version? Thanks, SingHui ---------- Forwarded message ---------- From: Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> Date: Jul 22, 2005 1:02 AM Subject: Re: [Speex-dev] Fixed-point To: gue baja <gue_baja@yahoo.com> Cc: speex-dev@xiph.org Hi Baja, Here's a quick

Greetings, Quick question: I am about to acquire some disks for use with ZFS (currently using zfs-fuse v0.7.0). I''m aware of some 4k alignment issues with Western Digital advanced format disks. As far as I can tell, the Hitachi Deskstar 7K3000 (HDS723030ALA640) uses 512B sectors and so I presume does not suffer from such issues (because it doesn''t lie about the physical layout

Hi, Merry Christmas to all. :) I have been trying out new 1.2beta3 here and facing some noise problems, (I can't say this is 1.2beta3 problem because I haven't tried out previous version yet) and was wondering if anyone also has faced and solved similar difficulties. When the speex encoding/decoding (wideband) process starts, the voice is crispy clear with almost no noise in the

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

I am using rails edge. I am using gem "rspec-rails", "= 2.0.0.beta. 19" . I have following code at spec/models/user_spec.rb require ''spec_helper'' describe User do it { should validate_presence_of(:email) } it { should validate_presence_of(:name) } end Here is my gemfile group :development, :test do gem ''factory_girl_rails'',

Hi, I would like to see if during a call a new voicemail was recorded. I want to send a SMS to mobile phones if someone recorded a message on our voicemail system. I can use VMCOUNT to see if there are new messages in the Inbox but this will result in new SMS being sent even if the caller hangs up during the Voicemailpromt, at least if there are still unread/unheard messages in the inbox. Is

Hi We are currently using Asterisk 1.2.4 with IAX and app_meetme for conferencing, but are looking to move to SIP because of issues with an IAX control we're using. The reason we moved from SIP to IAX in the first place was because of the poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older

I have been trying to get a DIALSTATUS output from a call started with originate. I searched a fair bit and have found several references to using local channels to do this. However, I could not find enough of the specifics to get it working myself. What I need to do is dial a zap channel and run various scripts if the channel is answered, busy, no-answer,etc. Here is the dial plan I am

Hi all, is it possible to bridge a call between a Junghanns quadBRI card and a TDM400 in the same server? It should be I think, -- I am trying this and when an incoming call comes in, it hangs both up at the moment the bridge is attempted (and a subsequent 'qozap: dropped audio' error is show in the /var/log/messages) Any thoughts appreciated -- I've seen posts, but no clear

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

I will be on vacation from <22/06/06> to <30/06/06>. I will not be reachable on my mobile. I will have limited access to mails, and please expect a delayed response. In my absence, please contact the following: Ray Richard or Safeer Mohammed Thanks H.Gireesh

On Dec 3, 2007 1:24 PM, Stefan Reuther <streu@gmx.de> wrote: > John Miles wrote: > > What's wrong with a plain old .bat file, or even an NMAKE .mak file? > > Ship two files, debug.bat and release.bat, and call it good. > > > > It is best to leave project-file creation up to individual users, > > in my opinion. > > I second that. When I played around