similar to: rxgain/txgain in chan_sip

2011.08.23. 15:38 keltez?ssel, Yanick Bourbeau ?rta: > Since I don't have access to different channels as I record a phone call > using a man in the middle approach, there is something else I can use > to equalize the sound ? What I would do then probably is try to manually separate the two channels/talkers; say channel 1 goes from 0 seconds to 13 seconds, channel 2 goes from 13

Hi, I''m a bit unsure as to when one uses a nested resource and when one uses a normal resource. If you have a belongs_to, has_one/many relationship between models is that automatically an indication of a nested resource or can these resources still be represented in the normal resource way? I have a resource (talker) that belongs_to a number of other models (network, data_date,

Hi, Do you have some experience with the "best" values for those parameters in youyr particular case? I mean the best raport between sound level in both direction and echo cancellation. For me, the best result I can get is with: rxgain=10 txgain=15 ... the sound level is good, but the echo is a little bit to strong for my taste. Something interesting is that if I put a txgain value

Hi, I know that this issue has been discused guite a lot, but I haven't managed to get a definite answer. Is those two values supposed to be floats (e.g. 3.5) or integers with the percent symbol (e.g. 20%)? Thanks, Yiannis.

Upgraded to Asterisk CVS-v1-0-12/23/04-18:34:44 and txgian / rxgain don't seem to work any more. Is this a know problem? I am using two X100P cards. -Thanks

Hi, does anyone have test numbers in Germany that would allow me to tune my rxgain/txgain settings? I know there are numbers provided by other providers in UK e.g. but have yet failed to find a number in Germany (esp. by Deutsche Telekom). Kind regards, JP

I am looking for docs on how to diagnose and adjust the rx/tx gain in zapata.conf. The wiki has a link to this article but it no longer exists on the server. http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never

Hi all, I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable. - I park the call using ParkAndAnnounce - plays moh. - accept the call using ParkedCall The following errors are coming on the console and there is oneway audio - no audio after Music-On-Hold at caller's side. Please advice. I am testing using cisco 7902

I am trying to obtain optimum gain settings for a bank of analog lines connected to a channel bank. My telco has provided a 'Type 102' test line to use for incoming level calibration. This is functionally equivalent to app Milliwatt(), but provides tone from the CO inwards. Question is, how should one use this a 0dbm test source with ztmonitor? Am I correct in understanding that a 0dbm

Jean-Marc Valin wrote: >> As you can tell, the AAGC integration with speex was really a classic >> hack. Instead of re-creating the hack, what's probably best here is to >> integrate AAGC back into speex, and have a proper API. >> > > Agreed here. If you can come up with a clean patch to add that feature, > it's something I'd like to see in

Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729

Hello, I am working on a trixbox based system with a TDM410P connected to 3 phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN with some polycom and Aastra SIP phones. In general everything works. the problem I am trying to solve is that if both parties to a call speak at the same time one of the voices gets cut out such that the talker A cannot hear what talker B is

Jean-Marc Valin wrote: >> I hate to be a talker and not a do-er, but I won't be able to write this >> myself, probably someone on the iaxclient team could do it. >> > > Anyway, let me know if/when someone's working on that. > > >>> Hmm, or does that mean the analogue AGC is actually completely >>> independent from the "real"

Hello everyone: I'm a new member of this group. I have a question about "oneway.test". When I use "anova(lm(....))" to analysis the ANOVA, I can get the information about Sum Sq and Mean Sq. (The R code and the results are as follows.)

> I hate to be a talker and not a do-er, but I won't be able to write this > myself, probably someone on the iaxclient team could do it. Anyway, let me know if/when someone's working on that. >> Hmm, or does that mean the analogue AGC is actually completely >> independent from the "real" AGC. Any thoughts? >> > > It's actually a bit more

Hi, i'm looking for an R function to fit a one-way ANOVA with one factor containing 10 levels. The factor levels have different numbers of observations (varying between 20 to 40). For most of the dependent variables i'm testing there are unequal variances among the factor levels. I see the function oneway.test: oneway.test(variable ~ factor, data=dataset) which by default does not

Hello everybody, I use the sweetpotato database included in R package: data(sweetpotato) This dataset contains two variables: yield(continous variable) and virus(factor variable). Due to Levene test is significant I cannot assume homogeneity of variances and I apply Welch test in R instead of one-way ANOVA followed by Tukey posthoc. Nevertheless, the problems come from when I apply posthoc

We started seeing odd behaviour with clients somehow hammering our ZFS-based NFS server. Nothing is obvious from mpstat/iostat/etc. I''ve seen mention before of NFSv3 client dtrace scripts, and I was wondering if there ever was one for the server end, displaying top talkers, writes/reads, or locations of such to nail down abusive clients short of using snoop/tcpdump to nail down via