similar to: Echo cancellation and DTMF from the Asterisk console?

Hello all, I'm a hardware person who has recently found himself thrust into a software role. So, please forgive any ignorance in the following questions. I've tried to do my homework reading the relevant manual pages and investigating the Doxygen documentation. :-) We're investigating acoustic echo cancellation solutions for a speech application. Our hardware will be a TI C64x+

Hello all, I'm looking to troubleshoot some echo issues and possibly "studdering" while using the speakerphone with the Intellitouch ITC3002. On the hardware end, i have a 2621 setup as the router with some policy maps for qos, and a 2900XL with cos priority set to 5. I have setup created 2 vlans currently (really small office), 1 for corp/data, and 2 voice vlan. Asterisk is on

Hi guys, I'm running Asterisk 1.2.13 on a Debian Linux system (that was just the version that was packaged for it). I've been using monitor() to record calls, with fairly satisfactory results - at least until the last few months. I've been recording VoIP calls, and using monitor() with no arguments, so I'm getting separate wav files for each leg (both use ALAW, BTW), and

Hi, I'm getting some strange behaviour on Asterisk 1.4 running on Debian Stable (Lenny). I suspect it's something to do with my setup, rather than a bug, but I'm struggling to see it, and would appreciate any input. Setup: PC with two ethernet cards: eth0 goes to local network, including two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes to router and

Hi all, We have discussed so many about sampling rate asynchronous (or offset) between rendering (D/A converter) and capturing (A/D converter) of most PC soundcards. It seems all acoustic echo cancellers, include AEC in speex, can not deal with this trouble, because it causes a drift of echo path and also buffer overflow and underflow which jumps the delay of echo path seriously. Unfortunately,

Simply to say, in a quiet room, you can play a impulse signal and then find it's impulse response signal from the microphone. For example, if the delay between the impulse signal and its response signal range from 500 to 3000 cycles, you can buffer the far-end signal to 0-300 cycles and set the filter length to 4000. It is also called to align far-end signal and near-end signal. BTW: Speex

hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090827/8d6c680a/attachment.htm

Hi Who should insert dahdi_echocan_oslec.ko module in Ubuntu 14.04? dahdi start/stop service? I have a test installation that is not launching it when asterisk/dahdi starts. Clues? Regards Ethy

Mike Ball a ?crit : > We?re investigating acoustic echo cancellation solutions for a speech > application. Our hardware will be a TI C64x+ series processor (DM6467) > which has both a DSP and ARM9 core. Our audio will be sampled at 8kHz > and fed in through the I2S ports on the DM6467, echo cancelled, and then > encoded in G.711. As such, I?m really interested in the echo >

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the

From: Steve Underwood <steveu at coppice.org> > It seems some cards use a PLL for their ADC, so they can lock to an > incoming SPDIF signal, but always use a local crystal clock source for > their DAC. These cards do not have their ADC and DAC synchronised. Do common on-board or PCI sound card lock to some incoming signal? Yes, there is a crystal oscillator and a PLL or divider to

Then why ONE sound card have different capture and playback rate? It must be ONE single physical clock generator which is used by both ADC and DAC in the sound card, isn't it? If you are a hardware engineer. Will you design two different physical clock for ADC and DAC seperately? What on earth causes this problem? Who knows its intrinsic real reason? Isn't there any other solutions? For

On Thu, 22 Oct 2015 15:06:59 +0300 Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Wed, Oct 21, 2015 at 01:33:27PM -0200, Ethy H. Brito wrote: > > > > Hi > > > > Who should insert dahdi_echocan_oslec.ko module in Ubuntu 14.04? > > dahdi_echocan_oslec should be built by default. What you may miss is the > 'echo.ko' (OSLEC) kernel module.

These patches fix various issues which were reported or found with the MIC driver. Changelog ========= v2 => v3: * Reorder patch 5 in v2 to patch 4 in v3. * Split patch 4 in v2 into patches 5 and 6 in v3. v1 => v2: @ https://lkml.org/lkml/2013/11/26/376 * Address review comments @ https://lkml.org/lkml/2013/11/26/443 in patch 5. v1: Initial post @ https://lkml.org/lkml/2013/11/26/321

These patches fix various issues which were reported or found with the MIC driver. Changelog ========= v2 => v3: * Reorder patch 5 in v2 to patch 4 in v3. * Split patch 4 in v2 into patches 5 and 6 in v3. v1 => v2: @ https://lkml.org/lkml/2013/11/26/376 * Address review comments @ https://lkml.org/lkml/2013/11/26/443 in patch 5. v1: Initial post @ https://lkml.org/lkml/2013/11/26/321

>>>> Hi, >>> >>> I have a scenario in a mobile VoIP app that requires echo cancellation but >>> is somewhat different from what's described in the docs. >>> >>> Audio is received from and sent to the network at 8000Hz. Each packet >>> contains 160 samples worth a playback of 20ms. >>> >>> But the hardware

So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230908/dee530c8/attachment.html>

All I can say for now is that: 1) My diagnostic tool does funny things that need to be fixed 2) It was probably also getting confused by the clicks at the end of the files 3) There seems to be odd things with your recordings, though I can't say what that would be. Are you changing something during the recording by any chance? Note that if a person close to the mic of speaker moves, that

I have Asterisk running on my LAN with softphone clients (SJPhone) and Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the caller places his SJPhone on

On 01/19/2011 06:44 PM, LiMaoquan2000 wrote: > > Hi all, > > We have discussed so many about sampling rate asynchronous (or offset) > between rendering (D/A converter) and capturing (A/D converter) of > most PC soundcards. It seems all acoustic echo cancellers, include AEC > in speex, can not deal with this trouble, because it causes a drift of > echo path and also