similar to: txgain for SIP?

Hey everyone, I have been trying to figure this out and I am just getting no where with it. The office is using Polycom IP 601 phones. Everything sounds great in terms of quality on both heads. However, users of the phone are having trouble with their headsets and handsets. Some users are hearing their voices back come back on the phone. If they adjust their volume lower, then it goes away,

> > 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

Hi! When I try to load the ztdummy driver via "insmod ztdummy", I get the following errors: /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_register I'm using zaptel-1.0.8 and

>The call always has echo on it. The Asterisk sip extension >hears them selves echoing. The remote party does not notice any >difference. >I have also notices that the Asterisk >levels are very hot from T1-PRI to Sip. I have had good success fiddling with the txgain and rxgain values in zapata.conf on my PRI. In my setup, cranking the gain down a LOT eliminated most of the

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

Hi I've whined about this before, but now I feel the overwhelming urge to whine some more. Forgiveness begged... :) I've spent some very frustrating times trying to get a microphone to work when doing skype calls. According to the various sound tools available on Centos all the various mics I have (built into webcam, analog headset, USB headset)produce sound just fine. but when using

Hi, I'm running Asterisk 11.7.0 with app_jack and I'm finding a software echo. When I answer a call, I get two dahdi jack ports to connect the call. One for the phone mic and the other one for the phone headset. The audio that I send to the phone, comes back in the other jack port. For example, If I create a jack metro that beeps every second and I connect it to the jack port that goes to

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

Hi I've whined about this before, but now I feel the overwhelming urge to whine some more. Forgiveness begged... :) I've spent some very frustrating times trying to get a microphone to work when doing skype calls. According to the various sound tools available on Centos all the various mics I have (built into webcam, analog headset, USB headset)produce sound just fine. but when using

Hi All, I have set following milliwatt and dial in from my mobile phone. and i adjust txgain value in chan_dahdi.conf but no effect on dahdi_monitor value is same. even i change its to 20 and more but monitor still showing same value. But i can hear tone variation on my mobile when i am changing txgain... is there any specific value i can set in txgain or leave it default 0.0 ( # = Audio Level

Hi! I'm searching for a 1-800 number that simply plays music for a long time (>3mins) and no one picks up. I've bothered the AT&T lines so far when trying out my SIP->PSTN connection but then always someone answered :-) Anyone have a number? Christoph

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

Is it possible to adjust the rx/txgain values per device? I have a mishmash of different phones (soft phones & sip hard phones) and each of them sound different w/the same rx/txgain settings. Is there any way to adjust these via asterisk? I'm having the most difficulty with sjphone & these zultys zip2 phones, vol is right on one, but way off on the other. Thx again...

I'm having a problem getting a good clear output sidnal from Ekiga to a VOIP conference call using the Ekiga.net free conference call system. I'm told that each time I speak, my voice is clear & intelligible for about .5 - 2 seconds, but then it starts to be garbled, sounding like the sounds R2D2 makes. I've used 2 or three mic/headsets - two plug into my audio I/O sockets on my

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

Hi! I have two Asterisk installations, one being a 1.09 bristuff installation and one 1.08 installation. In the 1.09 installation I have the "Dial" command available on the CLI, in the 1.08 installation I don't. My question is now: was that a new feature in 1.09 or is it a bristuff specific thing or is my 1.08 installation simply lacking features? Thanks, Christoph

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, 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

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

>> I'll need to add support for saving audio to my program, so I can give you >> the "actual" sampled loudspeaker and mic files, and I'll also need to get >> hold of a test person again. (I had a friend with a friend who has an >> exceptionally clear voice. My own "aaaaaa" is far too muddy to cause >> this). I'll try to get this done