similar to: using voip software client as public address system. Low volume

Hi, i want send sound to my yamaha amplifier trough digital output from my audio-card. When I make: speaker-test -Dplughw:0,1 -c2 i can hear the sound. If I set the DevicePCM1 to default i can hear sounds from analog output trying to set hw:0 hw:0,1 plughw:0,1 didn't produced any sound (analog nor digital). How should I setup Alsa Driver in the wine registry? asound.conf and .asoundrc is

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)

Something perhaps noteworth, since this is a multihomed system I bound the transport to 172.31.253.4:5060 I don't *think* that would cause Asterisk to use that IP in the FROM...at least it shouldn't. -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 2:58 PM To: 'Asterisk Users Mailing

I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW, the CONTACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message----- From: Eric Wieling [mailto:ewieling at

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

type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888 at

> As the thing is to encode for human ears (AFAIK), I'd say that 4kHz is already "quite high", > and I wonder who can actually hear pure 20kHz sine. If you read the beginning of RFC 6716, you learn that Opus never encodes any frequencies that are higher than 20 kHz. So at some medium or high bitrates, anything above 20 kHz is filtered out, not because of the bitrate but

On Sat, Oct 15, 2016 at 05:02:55PM -0600, Frank Cox wrote: > On Sat, 15 Oct 2016 18:49:53 -0400 > Fred Smith wrote: > > > Any clues? > > Is it possible that the Connector setting under the Output tab is wrong? The volume control might be controlling the volume on an unused output. Well, I don't think so... there are only two choices there, Headphones and Line Out.

You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:

I am setting up a test domain to learn bind, and how it,s working. The master domain is bbb.ttt.domene 10.11.12.13 the child domain ftp.bbb.ttt.domene 10.11.55.77 The thing I don't understand is, why don't the query report additional info (ip address)? Her is two queries(the first the one that troubles me), cut of named.conf, test of zone file, the zone file and cut from log file.

On 16-01-13 21:52, Clemens Ladisch wrote: > Rene Herman wrote: >> I'm out of touch of with alsa -- but should dmix/dsnoop still be >> enabled by default on hardware that doesn't do hardware mixing? > > Yes. > >> I believe that it should be from /usr/share/alsa/cards/ICH4.conf, >> but I don't get mixing on this setup when using the

Hi all, I am trying to extract from the aplay -l output the Card and Device numbers. I had a script that was working but found another motherboard that has an EXTRA word. HDA Nvidia and HDA ATI HDMI is what I am talking about below. how do I search for a line that has HDMI (which I know how to do), then ignore everything upto the device work and take the next field after that? Thanks, Jerry

You may also want to check this out: plot(ttt, type = "p") points(ttt, col = ifelse(ttt < 8, "black", "red")) Eivind K. Dovik Bergen, NO On Tue, 1 May 2018, Christopher W Ryan wrote: > Excellent! Worked like a charm. Thanks. > > --Chris Ryan > > On Tue, May 1, 2018 at 4:33 PM, William Dunlap <wdunlap at tibco.com> wrote: > >> The

Hi, I found the main reason for sound distortions on my system is guest timer configuration. The working one is: <clock offset="localtime"> <timer name="hypervclock" present="yes"/> <timer name="hpet" present="no"/> <timer name="rtc" present="yes" track="guest"

temp.ttt <- "ttt <- 1\nttt" conn.ttt <- textConnection(temp.ttt) source(conn.ttt, echo=TRUE) ## name of variable is echoed close(conn.ttt) cat(file="c:/temp/temp.R", temp.ttt) ## name of variable not echoed source("c:/temp/temp.R", echo=TRUE) temp.abc <- "abc <- 1\nabc" conn.abc <- textConnection(temp.abc) source(conn.abc, echo=TRUE)

The ts method for plot() is quirky. You can use the default method: plot(as.vector(time(ttt)), as.vector(ttt), type = "p", col=ifelse(ttt<8, "black", "red")) Bill Dunlap TIBCO Software wdunlap tibco.com On Tue, May 1, 2018 at 1:17 PM, Christopher W Ryan <cryan at binghamton.edu> wrote: > How would I color points conditional on their value in a plot

We have frequent timeout problems after upgrading our imap servers from dovecot 1.1.16 to dovecot 1.2.11. One server acts as proxy only, and the other one is the "real" imap server". The credentials for the proxy service are stored in a remote MYSQL database. There were no trouble with dovecot 1.1.16. But now, with the most recent version, we get frequent login failures. It

At times we want to convert a two-sided formula to a one-sided formula. In S we can do this by dropping the second entry in the formula. In R that object no longer has a formula class. R> ttt <- score ~ age | Infant R> class(ttt) [1] "formula" R> length(ttt) [1] 3 R> ttt[-2] [[1]] ~ [[2]] age | Infant R> class(ttt[-2]) NULL R> do.call("~",

Excellent! Worked like a charm. Thanks. --Chris Ryan On Tue, May 1, 2018 at 4:33 PM, William Dunlap <wdunlap at tibco.com> wrote: > The ts method for plot() is quirky. You can use the default method: > > plot(as.vector(time(ttt)), as.vector(ttt), type = "p", col=ifelse(ttt<8, > "black", "red")) > > > Bill Dunlap > TIBCO Software

Hi: I have some problems when I use the function "solve" function in a loop. In the following code, I have a diagonal martix "ttt" whose elements change in every iteration in a loop. I defined a "dpoMatrix"class before the loop so I do not need to define this class every time in the loop. The reason is to save some computing time. The code is below. The inverse