similar to: Current echo status?

Hi list, would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side,

Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none

Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and

Hi Everyone, I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

I came into this with my eyes wide open. I have read ABSOLUTELY EVERYTHING there is to be found on the net about avoiding echo problems BEFORE I even attempted to create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production

Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone.

Hello, Does anyone have experience with a %100 all VoIP * setup? I imagine an office with 50 extensions or so, a full T1 connected to a decent ISP and an account with NuFone (IAX2 trunking, G729a). I would need to support at least 25 simultaneous outgoing (7960G -> Asterisk -> NuFone -> PSTN) phone calls. I would probably keep four or so analog lines for local calls, 411, 911,

We have two small business that run out of our office. One business has 3 phone lines, and the other has only one. In a perfect world, Asterisk would indicate WHICH line (or group) the outsider caller called, so that we would know which way to answer the phone. The incoming calls would go through a voice mail menu first - different for each company. (Press one for sales, two for

Noob here, my apologies in advance. Recently my employer decide to stop paying for my home POTS line, so I ordered packet8 for a home line instead of another POTS line, since I really dislike my local phone company, and the POTS line without any long distance would cost more than the $20 to packet8 with unlimited US calling. Anyway, this whole thing got me started thinking about VOIP a lot

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

I have an x100p which suffers from echo (no surprise there apparantly :), and a few of the things I've read about tuning out echo say the first thing to get right is the tx and rx gain, and for that you need a few different types of test lines from the telco (Telstra, Optus, whatever). Does anyone have any numbers in Australia for such lines? I think they just need to send out a tone of a

So far no one I have talked to has either had this issue or does not know a answer. I currently run asterisk 1.0.9. I have two issues to deal with. 1. The caller waiting caller ID does not show on the uniden hand held that is hooked to a sipura spa 1000 or my Polycom ip500. 2. When a caller is calling in and I hear the caller waiting beep on the line when talking with somone, is there no way

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 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

We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

Any help on this would be great I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune

Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS