search for: loud

Working on a paging system for one of my sites and running into something I can't believe is this hard. In one of the zones, they want to have three different extensions ring over the pa system, using it as a loud ringer. Now the paging system does have a loud ringer built in and I can easily have it do a simultaneous ring, but all of the extensions will sound the same over the loud ringer. Of course, we want them to have different rings over the pa system so that all three people don't have to check...

We are currently considering the Linksys POE switch for a small Asterisk office deployment. There will be no separate wiring closet to put it in. Can anybody tell me if this switch has a loud fan? Users would not be able to tolerate a loud noise close by. Otherwise we will go with a fanless desktop switch. Thanks, -- Andres

What would be a good parameter to return that would better represent the loudness of the signal? As for the your other comments, I'll make the necessary changes and resubmit. Thanks, Mihai On Oct 25, 2007, at 7:45 PM, Jean-Marc Valin wrote: > I'm in favor of the idea, but not of the current implementation. There > are two problems: > 1) The st->loudn...

Hi, I've been looking a bit at the VAD/AGC situation. I noticed a leftover variable declaration. The AGC used to use nb_loudness_adapt to count the number of frames so far, but it now uses nb_adapt which is shared by all the functions. However, it still allocates and updates nb_loudness_adapt. This patch saves 4 bytes of memory and a partial cycle of CPU power ;) -------------- next part -------------- Index: libspeex...

...actual volume keys doesn't stop the distortion, so that points to a problem with asterisk (the volume can be very loud, barely audible, but you can still hear the clipping occuring). By clipping, I mean the static that happens when you have a signal that's too loud. The thing is, when you call directly into the Polycom over the PRI, it's fine. This ONLY happens during a conference call with g.722, though...

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike

...g 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 still could be louder. Maybe a light-up option would be better. The old phone system here had some huge loudspeakers that someone had wired right into the speakers of the old digital phones. I haven't figured out yet if they need a different voltag...

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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: http://lists.digium.com/pipermail/asterisk-us...

I am having a problem with and Asterisk installation where two ports connected to a TDM400 card will have a very loud noise when you try to dial. The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank 8 fxs. It is running Zaptel 1.4.11 and Asterisk 1.4.18. The problem always happens with two ports (34 and 35) which are connected to two GSM gateways. They will work fine for a week and then when you t...

...s the in and out channels become out of sync. It seems like one channel happens faster or has data missing when sox mixes them together. Digging around, I found MixMonitor, which skips the whole soxmix process. I figured that removing that step could only help. Now it seems that if I'm really loud on a call, MixMonitor stops recording. The wav file stops growing. The log says nothing. When you hang up the call, MixMonitor reports that it is exiting, even though it hasn't been recording since that loud noise. Has anyone experienced such a problem with MixMonitor? Is MixMonitor well teste...

Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090128/fb8b49ce/attachment.htm

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pi...

...and ATA-186 extensions work fine, of course. They use ulaw encoding. It's just the MOH that fails to answer. There is no NAT or firewall involved -- all hosts are local. I'm testing with dedicated extensions like this: exten => 302,1,WaitMusicOnHold(5) exten => 303,1,MusicOnHold(loud) exten => 304,1,MusicOnHold(default) Here's the asterisk output from a working ATA-186 call: *CLI> -- Executing MusicOnHold("SIP/200-990d", "loud") in new stack -- Started music on hold, class 'loud', on SIP/200-990d [ beautiful music emanates from ea...

Hi, Does someone know if I could use SpeexPreprocessState::loudness or SpeexPreprocessState::loudness2 to create a VU meter. And if so, how? <p>Thanks, Tom Moers --- >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 'speex-dev-request@xiph.org'...

Mihai Balea wrote: > What would be a good parameter to return that would better represent the > loudness of the signal? > As for the your other comments, I'll make the necessary changes and > resubmit. Could just be to first take the 5th root, a bit like in the code. Jean-Marc > Thanks, > Mihai > > On Oct 25, 2007, at 7:45 PM, Jean-Marc Valin wrote: > >> I'...

...ers leave. Every message that a caller leaves is very faint. I've already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? One interesting artifact is that at the end of every message, there is a loud pop (I think I usually hear a faint click when the caller hangs up and then I surmise that the loud pop is some sort of local connection ending. If I do: sox msg0004.WAV zz1.WAV stat it always tells me that the volume correction should be 1.0000. I surmise that the volume peak is triggering...

...al 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. This is especially true for the chirp for call-waiting which is excruciating for someone who is having a conversation. There is also a very loud chirp and the very end of Voicemail recordings. Is it possible to turn down these levels? How about a software limiter? Or a filter that cuts out hu...

Sorry. I forgotten the words volume or loudness. But it is know as microphone stroke too, i think. If something can tell me something about that procedure it would complete my pleasure. To bring back memories, i only wanted to know wheather i can change a variable that holds the sound intensity (loudness) needet to start "encoding >&...

Two days ago, I was talking on the phone from the FXO, to a SIP phone. After some time (like 1h30m), all of a sudden, there's a huge noise, like a buzz... Really loud. So I hungup, and called my asterisk box again... All I could hear was that sound. Someone called me from the internet, and as Asterisk dialed the FXO, all she heard was that noise too. So, I logged in my Asterisk server, restarted the Asterisk (just the software). Didn't work. So I stopped it...