similar to: Best Echo Canceller.

I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked "Receive In" and "Send Out". The 2 connectors on the drop side are marked "Send In" and "Receive

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter > Sent: Thursday, August 11, 2005 12:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped > betweenpstn & norstar > > > I poured over my logs most of

Hello, I have been researching Echo Can's for a while now, and wanted to post this out to the list to solicit feedback (and if my assumptions are correct, hopefully help others out)... If anyone out there knows anything about wiring up an Echo Can, and my outline below is incorrect, please let the me/list know.. I have posted the Documents I received from Tellabs at:

Since putting my Tellabs EC into place around 2 weeks ago, the echo problem has almost been eliminated. Reports of some very faint echo, but everybody is happy. My question is, if I were to also turn on the Asterisk Software EC, would this remove any residual echo that may make it past the Tellabs Hardware EC. Thanks, Doug

To answer my own question for the record: The relevant timing parameters in zaptel.h are #define ZT_MINPULSETIME (15 * 8) /* 15 ms minimum */ #define ZT_MAXPULSETIME (100 * 8) /* 150 ms maximum default, lowered to 100ms */ #define ZT_PULSETIMEOUT ((ZT_MAXPULSETIME / 8) + 50) And the pulse detecion loop that consumes these parameters begins at line 4866 of zaptel.c The

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

Hi All, Greg has been a huge help getting me going with this tellabs echo can, but I'm still having some problems getting it to work... I suspect I wired it up incorrectly, so I thought I'd see if anyone can point me in the right direction. Digium tech support pointed me to this doc for a standard T1 cable: http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm which looks like it

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Another question, Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our analog lines to a PRI, I thought it would be simple stuff moving the EC to the PRI. Changed signaling, made sure that channel 24 wasn't being ECd and everything came up. But, I was getting complaints of random echo on the PRI. Local echo. Also, we weren?t able to do any kind of modem dial-outs (Adit

Hello Dan, Have a look at this link: http://www.adcomcorp.com/asterisk/tellabs I got those pictures up there, may be of help. In essence, 1 pair is either a tx pair or an rx pair. If I recall, Orange and Green should be one side, and blue brown should be the other side. I tried to upload them to the wiki (even 40% size) but they didn't show up for some reason... Files are large on the

I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

Can anyone share any advice / documentation on how to configure a Tellabs 2572 T1 echo canceller? I connected one between a T100P and an Adtran TA750 FXO/FXS channelbank, but when echo cancellation is active I get a LOT of snap-crackle-pop (and other problems) on the line. The 2572 has a bunch of configuration options and they're impossible to guess without documentation, which I cannot find

Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call