similar to: Severe hum in recording

I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.

Hey guys, I am trying to diagnose a hum in my FXO lines I am using an Adtran 750 with 8 FXO ports. I am getting a pretty bad hum on the line during a call. I have checked the Telco side of the 66 block and there is no hum there so it's my problem to fix. I have tried to lower the gain but that reduces the call volume to much. Where else should I be looking? Setup as follows Dell

HI, We are experiencing the hum noise when the conversion of 2 parties is established. How can we eliminate that noise? ango

Rich Hi, After Qwest pronounced my circuits as within spec. (yes, disconnected from the house) I listened on the lines with my butt set. Clear of all noise and hum. I then got my box of Cat3 and laid a circuit around the outside of the house and into my lab. Still clear of all noise and hum. I then connected the circuit to the Adtran and made a call. Lots of hum and buzzz. As I said,

Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

I've recently begun integrating an Asterisk system into my house. I purchased the Dev Kit a year or two ago when Digium was selling it as a Z-plex 10 channel bank with the T100P. I've recently found that when I connect the serial monitoring port to my system it introduces a noticable hum on my incoming FXO lines. Anyone know (or have suggestions about) how to prevent this? -- Jamin

> I read your excellent post on the subject. I too am hearing the 60 Hz > hum on the FXO ports of my TA750. One Interesting thing is that it > goes away if there's an analog phone on the same POTS line, and you go > off-hook on it. I guess the analog phone has the repeat coils built > into it, and they balance the line for the FXO port. That's exactly correct; the

Hi, I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and 1 SIP phone. I've noticed some horrible buzz/rasping type of sounds! These seem to occur when * is trying to play back some audio or sound to me? E.g. If I have an exten rule which plays one of the music on hold files when I dial 800 lets say, I get a really loud buzz for about 2 seconds and then the music plays.

On 6/23/2015 2:05 PM, Frank Cox wrote: > I'll have to dig out the manual for my piano and do a bit of reading to see what's really going on with those plugs and whatnot. The only thing I've ever plugged into it so far is headphones. if you can plug in headphones, then you can plug it into a computer line-in jack with a suitable cable, just set the headphone volume on the piano

Bryan Walls wrote: > > I've set up Liveice on a Yellowdog Linux laptop (YDL 2.0). I'm trying > to stream to an icecast server on another YDL 2.0 system. > > Streaming seems to be fine, but encoding is pure garbage. When I > listen to the stream, it basically sounds like white noise. The > intesity of the hum depends on the strength of the sound being > encoded,

On 6/23/2015 10:14 AM, Frank Cox wrote: > On Tue, 23 Jun 2015 10:15:35 -0400 > Lamar Owen wrote: > >> >The USB MIDI port won't give you audio, just MIDI text. > That's what I thought. To this point, I've never done anything MIDI and I really don't know much about that; I just use my piano for the purpose of playing the piano. > midi is the electronic

Hello, We have one production server that is identically configured to another production server except for the fact that they each use a different type of digium quad T1 card(one has T400P and one has TE405P). The server with the 405 has been developing delay problems with Sipura SPA-2000 phone adapters after a call has gone on for 15-30 minutes (up to 1 second audio delay). I asked Sipura and

Hey Geoff, thanks for the reply. I had a chance to listen to it for quite a while yesterday and I realized that it's not a constant problem. It starts out really soft, so soft that you wouldn't even notice it if you didn't know what to listen for. I have two types of songs playing at the moment, loud music (rock/punk) and very soft music (middle eastern style). It seems to always

I've set up Liveice on a Yellowdog Linux laptop (YDL 2.0). I'm trying to stream to an icecast server on another YDL 2.0 system. Streaming seems to be fine, but encoding is pure garbage. When I listen to the stream, it basically sounds like white noise. The intesity of the hum depends on the strength of the sound being encoded, though. If I stop playing the CD, it drops to a gentle

I was fooling around with echo cans today. I found that I could reproduce this problem regularly t400p adit600 for fxs mec3 enabled Two speaker phones side by side within about 8" of each other. Put phone 1 offhook with speakerphone Dial phone 2 Phone 2 rings and is loud enough to be picked up by the mic in phone 1 By the second ring the echocan thinks it's figured things out, but the

Good day everyone, A Solution now exists for those of you still looking for the first model of Quad T1/E1 Interface Cards - the T400P and E400P. The T400P and E400P were manufactured by Digium until a few months ago, and were designed and released under the GNU Public License as the Tormenta II by Zapata Telephony. Varion, Inc. is now in full production manufacturing these boards. With the

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

Hi, I'm looking into support for SS7 and I found an article (http://www.openss7.com/news13022002.html) which says that OpenSS7 supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI interface cards. It also says that Linux Support Inc is the primary sponsor of Asterisk. However I cannot find these cards on the Asterisk hardware page

Hi All, I am calling mobile numbers from Soft-phone and recording the call. In recording the level of sound from the receiver's side is perfect (loud enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear me. But in recording my part of conversation is barely audible. I am recording using MixMonitor(). Is there anything that can