similar to: Experimental MEC3

I've been working on a fixed point mec3 echo can. The old "mec3" which crashed a lot of peoples machines because it was floating point in the kernel is now available as "mec3-float.h". I'd appreciate any testing / feedback on the new fixed point mec3 echo can. Thanks! Mark

I was fooling around with echo cans today. I found that I could reproduce this problem regularly t400p adit600 for fxs mec3 enabled Two speaker phones side by side within about 8" of each other. Put phone 1 offhook with speakerphone Dial phone 2 Phone 2 rings and is loud enough to be picked up by the mic in phone 1 By the second ring the echocan thinks it's figured things out, but the

Whilst in a call using the mec3 echo canceller today I had period of about 20 seconds of speech distortion. It's hard to describe but to me the call sounded as though we were having the conversation in a bathroom with some extra noise bursts and echo thrown in. I could carry on the call, with difficulty, and my correspondent didn't complain of any noise at all. After that 20 seconds

Hi Has anyone had any luck in putting two of these im a box? May I ask how.... I am starting to go bald.... I do have a suggestion is make another box and link them with iax2 Thanks Samantha r2d2:/home/s/samantha# ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2

I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked

Hello, does anyone else have a problem with Asterisk crashing right after a valid password/PIN is entered when trying to access voicemail in the 1.4b3 version? Not sure if this is anything to do with "realtime" per se but I keep getting the asterisk process bail on me as soon as a valid PIN is entered. Anyone? Cheers Ranj

Shoval, Interesting Mention. I agree, most people don't have CO exp. And I wish daily I had enough. Understand that what I mean by my e-mail is consumer side FXS ports, in broader terms, I mean, customer picks up a phone line, it signals a channel bank which signals *. 24 of those channels. Not channels equipped to Send Signal to the CO that a loop has been made.. meaning FXO. 24

Hello All, I am getting a set of errors when I boot Asterisk that I have not been able to solve. What is causing these error(s)? Asterisk boot output: ============== Asterisk CVS-04/10/04-21:44:51, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= [

Ahh Mr Carbuyer ... you should have _specified_ you wanted tires with that new car We can still help you though, it will just be an extra $$ above the price we quoted you I understand the concept. I see it in many industries until a company comes along that cares about it's customers I still think that digium is the best buy (for the small scale stuff that I'm interested in anyway) ...

ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE.... it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxxxxxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall -------------- next part -------------- An HTML attachment was scrubbed...

Hi List Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly phonegc:/home/samantha# asterisk -r Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Connected to Asterisk CVS-05 currently running on

What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI> A No such command 'A' (type 'help' for help)

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

I've added an important new application: app_transfer. This application is designed to allow Asterisk to request the transfer of an incoming call to a different extension. Consider the following diagram: Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App A caller calls an extension on PBX1 which forwards to PBX2. PBX2 executes app_transfer, which requests that hte

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools

Can anyone help me with this: - root@foxy:~ > asterisk root@foxy:~ > asterisk -r Asterisk CVS-v1-0-12/28/04-11:37:32, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> ========================================================================= Connected to Asterisk CVS-v1-0-12/28/04-11:37:32 currently running on foxy (pid = 1751) foxy*CLI> foxy*CLI>

Hello, Does anyone know what driver to use with an "Eicon Diva Server 4BRI" card? My modem.conf: -------------- [interfaces] context=remote driver=i4l msn=16453 device => /dev/ttyds01 -------------- I get the following error: -------------------------- == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem Driver) WARNING[8192]: File chan_modem.c, Line 383

I've merged some changes from Michael Manousos that should improve sound quality on non-zap channels, including music on hold. I'd like to hear back on or off list about your experiences with the new code. Thanks! Mark