similar to: sidetone noticeably loud on analog handsets on T100P

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 volume button but it also control the

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

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 calls) I get the side tone, but its delayed by a second or

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

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

Hi, We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2 extensions to iaxmodem devices for fax2email. We are rapidly growing and will be adding an additional PRI trunk and grow to about 150 SIP & IAX2 extensions towards the end of the year. We have two Digium Wildcard TDM800P cards (8 x

Paul, Ah, I see. Our echo is largly under control now. It took me a while to figure out the gains and get them tuned, and now the echo only leaves very small artifacts. Nonetheless, this still provokes the odd complaint here and there. We use VOIP for outgoing calls when our POTS lines are congested, and we find zero echo during those calls. Therefore, I assume that our handsets (Cisco

Brent- There is no good way to say what changing the hardware and PSTN hookup will probably do for the echo problems. I'm not sure if you mentioned (lost in the past history of your post now) what sort of hardware you're using for PSTN connection now- TDMs, X100s, ATA's, etc- but that could also be a potential cause. I've heard tell of aftermarket X100s and certain ATA's

Anyone got any pointers on where to find USB handsets or headsets that can be used as the audio device on a softphone? Later..

I'm looking for a usb cordless handset to pair with a softphone ( probably ekiga) on a pc linked to an asterisk server. I've loooked at the bluetooth headsets, but they seem overkill for just home phone extensions. I've found a number of handsets that work with skype, but they seem locked into skype. I've also found the Uniden WIN1200, which works with MS Messenger. It

I am having trouble getting my T100P and ADIT 600 talking. I have 2 FXO 8 port cards in the ADIT and a T1 controller card. I have a T1 x-over cable connecting two together. Here are copies of the relevant bits of my zaptel.conf and Zapata.conf Zaptel.conf span=1,0,0,esf,b8zs loadzone=us defaultzone=us Zapata.cfg context=default signalling=fxs_ks group=1 channel => 1-24 My

Hello Headsets we use are Plantronic's which are great, but I am still looking for a USB handset. Regards Nick

Peter - 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs for this. If you have ISDN you can also get DDI to the extensions.I would strongly recommend abandoning the analogue PSTN lines and using ISDN. The setup pain you will go through will be significantly less, combined with better audio, more features (like DDI numbers!) and much more robust connections. You should look

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

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

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

Hi guys, I'm new to the telco game and still pretty new to Asterisk, although I've been using it for a couple of months now and like most of what I see. At my office, we've got a small two extension setup w/ two Digium cards for a single FXO line and three FXS extensions, but I'm also working on designing a larger installation for a customer which will involve ~16 analog

Hi, I'm currently working on a * server connected to an. ahem. "Wireless Communication Server". The HWCS has an NMS AG 4000C. I'm using NI2, net side on the * box. The D-Channel comes up. The B-Channels come up. The first call to the * box goes through. After which, the HWCS with the AG4000 seems to either get a state-machine screwed up, or * does not send the correct