similar to: Multiple simultaneous calls

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hi! I want to use speex for echo cancellation in my program, but I have bad results. I will explain what my program does. it is a client-server application. I run a server in room A and a client in room B. the client sends some voice to the server and the server plays it on loudspeakers. I run another server in room B and connects to it from room A using the same application that runs

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

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

Hi, Indeed I too have troubles implementing echo removal, I like ask kindly to Jean-Marc (or any other) if him can put a source code demo to show us how to use effectively echo removal API and parameters in real case scenario. A big thank you! Roberto -----Messaggio originale----- Da: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] Per conto di hs Inviato: marted? 11 ottobre

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call

Having the same trouble that Mark ( http://www.zytor.com/pipermail/syslinux/2002-March/000244.html) had over a test platform I have here. Having a redhat 7.3 with dhcp V3.0rc12 and atftp (0.6.2) on a local small hub and a W2k Server with RIS installed. If I stop binlsvc on nt server my network boot run ok to pxelinux, but as soon as binlsvc is started, RIS server always respond over my local

Hi everyone! I'm a Linux newbie (running Ubuntu Edgy), and even newer to Wine, but anyhow ... My problem is the following: I've been trying to start up an old Win 95 game -- Curse of Monkey Island 3, if you remember it -- and have the two CDs on my hard drive as .iso image files. So, I mount the first CD to make it accessible, and then add it as drive d: in the dosdevices folder. All

Hi! there are some problems with echo cancellation filter from speex. problem 1: some noises, echo is removed, but sometimes you can hear some noises instead of echo. I was trying with many different parameters for buffer length (40ms and 20ms), filter length (from 100ms to 4s) and echo tail (2 to 5 buffers), but could not find the right setting. problem 2: it happens that the filter

Hello, Need some help using the speex_echo_cancel. I've read the documentation about the speex_echo_cancellation function: speex_echo_cancellation(echo_state, input_frame, echo_frame, output_frame); (in) echo_state => speex internal state. (in) input_frame => audio captured by mic. (in) echo_frame => the signal that was played in the speaker. (out) output_frame => the

_________________________________________________________________ Trouvez l'âme soeur sur MSN Rencontres ! http://g.msn.fr/FR1000/9551 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No

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 made a CD-backup with cdparanoia | oggenc and I tried to use different values of -q listening to the effect by cdparanoia | oggenc -o - | mplayer -. The best quality was 8 -- why? Oggenc version was the precompiled 1.0 from Mandrake 9. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

I was just wondering if the original audio is always the best audio. I'm sure every compression format including vorbis is based on trying to make the output sound the closest to the listener to the input. I was wondering if there is any possibility that there would be a way of modifying the huffman tables or something in some way to make the output sound better than the original? ---

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

This is off-topic, mostly, but I figure you guys will have some knowledge in this sound-quality-related area. I'm sitting here looking at the most recent "Musician's Friend" at headphones and thinking about getting a pair. They've got products from AKG, Fostex, Audio Technica, Nady, Sennheiser, and Sony, at price points ranging from $16 to $130 (list prices $20 to

Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500