search for: sidetone

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the vol...

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 make any difference if I adjust the rx/txgain in zapata.conf (on either the inside Zap channels [handsets] or the external [POTS lines] ones (which are routed...

Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it doesn't matter if we call a local, a mobile or a longdistance call. Only we hear the echo, not the called party. We do not ear any echo in internal call to each other extensions. I tryed every poss...

Hi all, I'm not getting any sidetone on my Snom 370. I searched the web and the snom wiki, but I don't see any place to enable/adjust it. Callers say I sound great on the other end, but I don't hear myself so it is a little off-putting. Any suggestions would be appreciated. On a related note, some times (maybe 1 out of 10...

Hi - I've got a new install with a Sangoma A200 and a few GXP2000's. When users are talking over the Sangoma, they get a lot of sidetone (local echo). Internal calls are fine. Where do I adjust that? I assume its in zapata.conf somewhere? thanks Todd

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

I am using GS 101 and as I am new to Ip phone arena. I am finding it a bit annoying to hear sidetone, especially when both parties are talking over each other occassionally. In that case, I cannot hear the other party's conversation. Is there any way to suppress it? Is it only GS or it applies to more expensive phone eg Cisco 7960 as well? -- David Kwok Iaxtel/FWD # 17001813482 ext 100...

...Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to voip are fine. * Outbound calls that do not bridge with the X100P are fine. * PSTN ->*->VOIP calls have so much echo on the called party side (sidetone) that it is almost impossible to have a conversation. I've played with rxgain txgain, echocancelwhenbridged,etc nothing seems to work. Yes Im completely restarting asterisk and running ztcfg. Anyone figured this out? Heres my zapata.conf : ; ; Zapata telephony interface ; ; Configuration fi...

....gain.rx.analog.ringer.IP300="-6" voice.gain.rx.digital.handset="0" voice.gain.rx.digital.headset="-21" voice.gain.rx.digital.chassis="0" voice.gain.rx.digital.chassis.obs="0" voice.gain.rx.digital.ringer="-21" voice.gain.rx.analog.handset.sidetone="-24" voice.gain.rx.analog.headset.sidetone="-24" voice.gain.tx.analog.handset="3" voice.gain.tx.analog.headset="3" voice.gain.tx.analog.chassis="3" voice.gain.tx.analog.chassis.obs="6" voice.gain.tx.digital.handset="0" voice.gai...

...occur in >the IP path due to network issues, and can occur on any sort of digital or >analog circuit due to various electrical or audio components. (one of the >more commonly neglected causes is poor handsets, that do a bad job of >isolating the speaker and microphone, or attempt to add sidetone incorrectly >(sidetone is the slight echo you should hear of your own voice- it's very >hard to hear, but without it, you get the feeling that you're talking to a >dead wire). Conversion to BRI/PRI is a last step only, in my opinion, >unless you have other compelling reasons to...

Hi all, I just purchased a Polycom 301 for my home office and I believe I have it setup correctly as I can dial out, receive calls in, etc. However, I'm having the following issue: When calling a local number over a Zap line, I hear a lot of feed back on the line. I had a Grandstream configured with the same information before I got the 301 and never had that kind of feedback noise.

Howdy: Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? If so, for the same reasons? This is a part of our consideration to transition to BRI. Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

...occur in >the IP path due to network issues, and can occur on any sort of digital or >analog circuit due to various electrical or audio components. (one of the >more commonly neglected causes is poor handsets, that do a bad job of >isolating the speaker and microphone, or attempt to add sidetone incorrectly >(sidetone is the slight echo you should hear of your own voice- it's very >hard to hear, but without it, you get the feeling that you're talking to a >dead wire). Conversion to BRI/PRI is a last step only, in my opinion, >unless you have other compelling reasons to...

...mittant calls. It is not isolated to a single extension or calls to/from a single destination. It seems that on random occasions, there are very bad echo (calls via the PRI trunk), it sounds like Silence Supression is being used, and there seems to be isolated cases that seems to sound more like a sidetone than echo on actual conversations / handsets. All these are only occuring on external calls, internal calls are working flawlessly. We've pretty much had this issue intermittantly since day one, had certified Digium consultants look at it, independent audits was done, various settings has ch...

On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's

Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". The echo I mean is when I hear myself while talking to another party. I have heard

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the