similar to: asterisk-users] Increase Volume - SIP

Hi guys, Can someone tell me if there is a way to increase the volume of a conversation that occurs between two SIP channels or between a SIP and an IAX channel ? My headsets are set to the maximum volume but the voice is still low ... I know there is a configuration in zapata.conf for the digium cards, but is there a place I can set this up for RTP conversations ? Thanks,

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

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

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:

Hi, On Tue, Aug 24, 2004 at 10:44:49AM -0700, Shana Cooke (Gitnick) wrote: > The result has my husband speaking clearly, but the echo of myself > talking is garbled and sounds like the Borg Hive from Star Trek. It's > more distracting than the original signal was. I have tried introducing > delays to my echo, but it does not help the signal to become clearer. Try this patch. It

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

hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config

Hmm, I don't have /etc/modprobe.conf, and wctdm is giving me problems. Which device is it talking about? [root@asterix /]# modprobe wctdm /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

I am having trouble with chan_zap.so not loading. When I load it from modules.conf, Asterisk bails out without any error message. When I load it from the console, it just says "Unable to load module chan_zap.so" no matter what verbose level I am using. dmesg says: Zaptel Version: 1.4.4 Zaptel Echo Canceller: MG2 Freshmaker version: 73 Freshmaker passed register test Module 0:

Ok, Here's another bizarre one (no strange curve balls to throw this time :P). I have several mp3 files of some easy listening music that I pulled off some CDs we have. They sounds fine and are at a nice volume level. When I run this script I wrote: (I run it by doing ./script < filename) mpg123 -s --rate 44100 --mono $1.mp3 > $1.raw sox -r 44100 -w -s -c 1 $1.raw -r 8000 -c 1 $1.wav

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Days ago I bought a OpenVox A400P card with a port FXS and another FXO that I am testing with my Asterisk installation in my house. I'm using Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on Debian GNU/Linux Lenny. I was reading "The future of telephony" and this [1] document looking for information about

Hi I have been searching the wiki and mailing lists and I cant see where my config is incorrect. I have a Digium tdm11b (1 fxo + 1 fxs) this is the output of cat /etc/zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0

Hello everyone, I have a Atcom AX-1600P(1) card with a FXO module and I can't configure it. I have four extension with this PJSIP settings: --- /etc/asterisk/pjsip.conf --- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=6001 aors=6001 direct_media=no rtp_symmetric=yes force_rport=yes

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAARRRRGGGGGGHHHHH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get:

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

I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and installed dahdi-2.2.1. kernel modules loaded. lsmod | grep wctdm wctdm 37233 0 dahdi 194985 1 wctdm lsmod | grep dahdi dahdi 194985 1 wctdm crc_ccitt 1549 2 dahdi,isdnhdlc dmesg: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.1

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

Hello I'm trying to install Dahdi through source code on a Fedora 13 host to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv isn't happy. 1. After successfully running "make all; make install; make config", I edited /etc/dahdi/system.conf thusly: loadzone=fr defaultzone=fr fxsks=1 2. Then ran "dahdi_cfg -vv" which says: ------------- DAHDI Tools