similar to: Echo on SIP -- not on analog.

I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Thanks for clearing this up. Alan

Hi, I am currently working on a thin-client "low-cpu" usage telephone application. The current setup is that you can make phonecalls using an alsa supported usb headset and a smartlink based usb modem (a sweex usb modem, which -in large quantities- goes for 17.95 euro's) for analog lines. Unfortunately, it is really made as a modem and not for audio purposes, so, there is a pretty

Hi, I'm using a 4-port BRI card (b410p) to make and receive calls (via chan_misdn): http://www.digium.com/en/products/digital/b410p.php This card supposedly has hardware echo cancellation. How can I check that echo cancellation is actually ON (taps, etc) on a given misdn channel (like with the "zap show channel X" command). "misdn show channel X" doesn't seem to show

Ok, I searched the lists and found no definitive answer. I'm assuming the IAXy has some primitive form of echo cancel, is there anyway to adjust this? Or any ideas on what to do instead. Here's the setup, this will not be a typical setup for our company however, well whatever. Anyway it looks like this: IAXy --> Netgear router --> Charter --> Level3 --> Asterisk --> VOIP or

Have a customer running a 3rd party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are utilizing Polycom phones. They are experiencing frequent static on the line, and overall insufficient volume on conversations. They are in a bit of a rural area, I was curious if anyone thinks it could be an issue with

I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940, but not when using a POTS phone plugged into a TDM400 FXO port. It doesn't seem to matter if the call

Perhaps 90% of my calls -- over a Uniden and a Grandstream -- are fine. The other 10% get some nasty echoes. Is there some magic something I should be tweaking? I kind of thought that echo cancellation was static, rather than dynamic, and that it's difficult to be able to cope with something like this. Am I (hopefully) mistaken? Any pointers on what to do about echo cancellation would be

I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken

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

OK I have been battling with echo problems with asterisk on ISDN for a few weeks now, and still can't solve it (although I think I have tried everything I can find.) I will try a post everything I think is possibly relevant that I can remember with the hope someone can point me in the right direction. current spec I am using (Trixbox 1.0 was still in beta at the time an didn't detect

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

I'm very, very confused. Dialing out, through VoicePulse, with both gsm and ulaw CODECs, most of my calls are great. However, calling my (non-Asterisk) voicemail at my job, and calling my cell phone both produce horrendous (~ 1/3-second delay) echo. I've tried with different phones (Polycom and Grandstream), different IAX CODECs (as described, above), different network

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on 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 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

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 :

I noticed I get echo on line when my internet connection is congested (during file downloads, for instance). Can you verify this isn't your problem? > -----Original Message----- > From: Doug Reid - Stormcorp [mailto:doug@stormcorp.co.za] > Sent: Tuesday, December 14, 2004 3:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

Hi All, Last week I started hearing a huge amount of local end echo on incomming calls. I am using a Sipura SPA-3000 as my FXO connected to an SBC POTS line. Echo cancellation is enabled in the SPA firmware. As a test I switched to a Digium X100 card the still lives in my server but the echo was about the same. I have both Polycom IP600 and SNOM 200 phone, which both hear the echo. I'm

[Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones "inside?" [In other words, I obviously need a NIC -- but what would be the