similar to: Disabled echo canceller because of tone (rx)

Hi guys, Being a true Linux geek, I've never been too much into sounds or sound files other than a few .mp3 songs I got. My question is pretty straightforward and simple. I see that the music format of choice for asterisk is .gsm. What can I use to listen to files in .gsm format and what is the most effective way of recording files into .gsm format? The last part of the question is

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that

I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or

Anyone know of anybody doing VXML with Asterisk and/or Linux? Tia Kevin

Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic and

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt

Hi,When captured speech contain sudden burst of sound like a bell ringing or a loud click sound along with echo,  echo canceller seems go wrong in latest svn version. Is it because of sudden  change from normal speech to say some high frequency sound?ThanksJam -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes,

This patch adds a 'hairpin' (also called 'reflective relay') mode port configuration to the Linux Ethernet bridge kernel module. A bridge supporting hairpin forwarding mode can send frames back out through the port the frame was received on. Hairpin mode is required to support basic VEPA (Virtual Ethernet Port Aggregator) capabilities. You can find additional information on VEPA

This patch adds a 'hairpin' (also called 'reflective relay') mode port configuration to the Linux Ethernet bridge kernel module. A bridge supporting hairpin forwarding mode can send frames back out through the port the frame was received on. Hairpin mode is required to support basic VEPA (Virtual Ethernet Port Aggregator) capabilities. You can find additional information on VEPA

This patch adds a 'hairpin' (also called 'reflective relay') mode port configuration to the Linux Ethernet bridge kernel module. A bridge supporting hairpin forwarding mode can send frames back out through the port the frame was received on. Hairpin mode is required to support basic VEPA (Virtual Ethernet Port Aggregator) capabilities. You can find additional information on VEPA

You probably ought to be using the raster package. See the CRAN Spatial Task View. -- Sent from my phone. Please excuse my brevity. On July 5, 2017 12:20:28 AM PDT, "Anthoni, Peter (IMK)" <peter.anthoni at kit.edu> wrote: >Hi all, >(if me email goes out as html, than my email client don't do as told, >and I apologies already.) > >We need to downscale climate

Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call

> > hi, > > How good or bad is the EC in Asterisk? > > Can anyone prove that it works at all and what it's limitations are? > > I ask cause I have some problems with this myself which variate from > call to call, and I see from others that Echo Cancel is a quite common > topic. > > Jan > > I have installed systems that had absolutely no echo

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make