similar to: RX gain TX gain

Hello! I'm having an echo problem with a TDM card. The TDM card is being fed by a channel bank just 12 or so feet away. When you put an analog handset on the line, both the RX and TX volume seem to be just fine. However, when I use the TDM card, I have to have an rxgain of 13.5, and even then, the audio is relatively quiet. I'm also getting echo on these lines, so I have turned

Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring

I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was rxgain and txgain. below is my current zapata.conf file All help would be grateful. I have tried

Hi, Problem: When talking to someone (from pstn) and this person is not talking loud, the voice is cut by Asterisk. I tried increase RxGain but it changed nothing (was talking louder but voice still cut.) I use XLite as soft phone. I think this is probably a codec setting... but how do I check that on server side? I just don't know what to do. All works fine (asteriskathome) but I always

Someone sent me a question late last night and I briefly looked at his file this morning and couldn't figure out the answer, so I'm posting here. A friend has a a ~275MB 24 bit, 48khz stereo wav file of rock music that when compressed using flac level 8 gives a flac file under 110 MB in size. When I dithered his file to 16/48 and converted that file to flac, the resulting flac file was

> we have a web-based station running liveice and aumix and the levels are all > over the place. is there a way to do automatic gain control on the soundcard > input? > > -peter Run the signal through a Compressor/Limiter before sending it to your soundcard. I use Behringer Ultra-Dyne Pro DSP9024. Very nice. If you want to buy it look here:

Nicholas is probably right about noise. Another factor would simply be the amplitude of the resulting file. A 275 MB 24-bit file which compresses to 110 MB is probably not very loud. I assume that the average level is somewhat low, with few if any peaks that reach 0 dBFS. FLAC is very good at compressing audio that is not loud. In fact, the quieter the recording, the smaller the

I've just Replaygained several of my Vorbis albums with Vorbisgain, ranging from 80's metal to present day soft rock. What I don't understand is why does Vorbisgain actually make all tracks QUIETER? I see an average of -7db on most albums. And after that, not only are they substantially quieter than my MP3s (which is a pain), but it also fails to really "even out" the volume.

Hello list... 1) Is there an IAX softphone that supports any kind of voicemail indicator? 2) I have 2 TDM400Ps installed in a system. I need the audio on the analog phone (FXS modules) to be amplified somewhere between 10 and 15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS interfaces. When a call comes in on the FXO at this setting, the call sometimes has about 20 seconds of

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

I have a very annoying problem that we hear on our end, but the other party doesn't hear. There are random beeps and echo type noises that occur. They are present during voicemails, and present on my end during calls. Is anyone experiencing the same deal? I have asked this a number of ways on the list, and never get a response... Thank you. Sean Garland Mount Shasta, CA

Hello, i'm using a TE410P on some E1/PRI with EuroISDN and experiencing a few audio quality problems with current CVS (both zaptel and asterisk) and the following network ISDN public SIP/zaptel network ---- pri --- ASTERISK GW --- iax --- ASTERISK PBX --- PHONES w/ any codec the rx (public network to local

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

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

Hi All, I have a ticket open with Digium, but based on their previous lack of support for the Asterisk Appliance, I'm not really holding my breath - and, honestly, I'm not 100% convinced it's a Digium issue in the first place (but I don't know where else to point fingers). We have an AEX-804E (PCI Express, 4 FXO ports, Hardware Echo Cancellation) in a Dell PowerEdge 1950 with

hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk roy@karlsbakk.net --- In space, loud sounds, like explosions, are even louder

I have a recorded conversation from an analog trunk. As usual one side is stronger that the other one. In my case, the gap between signal levels are even bigger. How does speex AGC preprocessor will perform on this type of audio recording? Maybe I am wrong and AGC is not really what I need to equalize the two persons in my phone conversation? As I Understand, AGC will perform better if each

does anyone know of or would be interesting in writing a program that is an alarm clock for your computer that plays ogg vorbis files? i used to have one that played mp3z before i heard about vorbis, but it got trashed somewhere and i can't find it again because i forget what it was called. but i would be interested to have one that plays ogg vorbis files, as right now the only alarm clock i

Hi. Today I was asked about a function of asterisk. That's what it should to: a call arrives -> put it in a queue -> remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone

Hello, could I adjust the Rx and Tx gains for SIP and CAPI? If it is possible, how should I do it? Thanks a lot. best regards, Felix -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110215/f36da2eb/attachment.htm>