similar to: Increase Volume - SIP

Hi Marc; Are you using VPN between the two sites? If that is the case, then you wil have a low volume and I faced this problem. Just try without VPN. Another issue: the call is originated from the Mobile (or PSTN) and you call to Zaptel and then do the call via SIP (or IAX) Trunk? Or you are facing a low volume when you call from FXS (or from IP Phone)? There is a way to increase the volume on

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With

Hi All; In the zaptel, we were increasing the gain of the voice volume at the hardware level from the /etc/zaptel and /etc/modprob.conf files, but now we are using DAHDI, so where to do the same thing? I am looking actually to increase the volume at hardware level and not software to avoid the DTMF detection problem and to have better voice quality. Any advise? Regards Bilal

I have tried both linear and non-linear (residual echo) parts of SPEEX AEC on MS Windows platform. Great job. Most echo can be cancelled. However, the voice volume becomes too small. Does any one have suggestions that makes voice volume louder but echo still can be cancelled? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am testing speex 1.1.6's echo canceller. I am using testecho.c, with a few modifications to get it to run on Windows. My problem is that I am unable to get the echo cancellation to work correctly. I am working on an audio conferencing software, and one issue we have is sometimes the microphone picks up what is being played through the headset, resulting in an echo of the other

I just noticed that quality of .gsm files for using with asterisk is not that good.. is there any way to make then sound better? asterisks sample voices sound way better than theones recorded using applications like wavepad or with asterisk like unavailable messages... any tips? Do you know the command line for sox to adjust the volumen levels or gsm files (make the louder)? Also, do they have

Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there appears to be no change on the phone volume. I can see that the

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

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 All, I have been wrestling with this problem for months now and at my wits end. The sound is always very low and if I try to bring the sound louder by using the sound references icon on the top panel, the sound is choppy with lots of static. Output of lspci for the sound device is as follows: 01:00.1 Audio device: NVIDIA Corporation GF108 High Definition Audio Controller (rev a1)

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

Hi, I just started with Asterisk as I am very unhappy with the functionality of my current PBX at home. I try to understand everything and play around, but it is not as easy as I thought. So please be patient if this is a too easy question for You. I installed Asterisk 1.4.26.3 on a Debian Lenny with IP 192.168.2.147 My extension.conf looks like this: [default] exten => 1001,1,Answer()

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 everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device

Hello, Can you please tell me what the overload point is for OPUS in dBm0 or dBV? As an example, the G.711 A law codec has an overload point of +3.14 dBm0 across a 600 ohm circuit. In ITU-T G.100.1 section 5.8 you can read more about the relationship between dBm0 and dBov. In that same section you can see that the G.711 A law, u law, and G.722 codec overload points are also defined.

Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

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

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