similar to: fixed point mec3

I've got mec3 working at my house now. I haven't tested it on anything but my FXS ports. If any of you have echo issues and/or don't like the AGRESSIVE_SUPPRESSOR stuff, I'd like your feedback on the new mec3 echo can. Remember it is HIGHLY EXPERIMENTAL, but find me on IRC or contact me off-list with your impressions of it. 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

Heya all, I'm experiencing some problems with a TDM40P and was wondering if anyone else on this list has similar experiences, or maybe even a possible solution. My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected to an analog line to my telco, and a TDM40P with analog phones connected. The TDM-card is not sharing any interrupts, but the X100P is, with the 2 Adaptec SCSI

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. Best Regards; Wassim _________________________________________________________________ Windows Live Spaces is here! It?s easy to create your own personal Web site. http://spaces.live.com/signup.aspx

Hi, I'm trying to get asterisk running on kernel 2.4.20 however trawling through the archives I've found a few references to patches to remove i4l's dtmf detection, but have been unable to find the patch itself (I think it is isdn_audio.c). Can anyone point me in the right direction? The problem I'm seeing is connecting a SIP softphone (tried a few) to an external number via an

Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to

Hi All, I have an equipment loaded with 4 X100P (numbered 1-4)) and one T400P (numbered 5-8). Everything works fine except that I cannot use one of the FXS ports (number 5). If I configure zapata.conf to recognize it, the whole system voice quality suffers. I've tried already to switch PCI slots, with no results. Below is a snapshot of my /proc/interrupts, maybe this can shed some light on

Hi, I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and several SIP phones and ATA's. We have a terrible delay on calls between the PSTN (isdn BRI) and the SIP phones. All internal calls are fine. My first thought was that the transcoding could cause the delay but all of the SIP phones default to ulaw so there should not be any transcoding needed. I also checked the load

Please read the reported bug and bjorn answer.. which does not help any to a solution of fix, or explenation. But the big question now is, does someone somewhere know what bjorn is talking about. i did search for "gencache" but no go here.. just from old documentation. https://www.samba.org/samba/docs/man/Samba-HOWTO-Collection/tdb.html gencache.tdb Generic caching database.

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

Hi, I currently use? Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network

Greetings, today I'm recompiling asterisk on a new machine with the same isdn card (not the same model, the *SAME* card) using CVS downloaded yesterday night. When I try to start *, it gives me the error: [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem

Has anyone made spandsp to work with a digium tdm fxo card? I finally got the rxfax and txfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in

Hi, I'd like to realize the following setup traditional phones --> pbx (non ip ) --> asterisk --- ISDN network is it possible to set up an asterisk server in such way that it acts as gateway between a traditionnal pbx and an isdn network ( I'm in France ). this to let people continue with their existing pbx + phones and enjoy voip features ? outgoing calls: if yes,do I need

Hi all, this is a rather "newbie-oriented" question, so please bear with me... The system running Asterisk has been provided with an AVM FRITZ!Card PnP. SuSE Linux 9.0 recognizes it right after booting the system and it seems to be configured (MSN) correctly... The hwinfo looks like this: --- pbx:/etc/asterisk # hwinfo --isapnp 11: ISA(PnP) 01.0: 10300 ISDN Adapter [Created at

Hi all, I'm new to asterisk and not too knowledgeable on ISDN, so please be gentle :) I have a dual-channel Eicon Diehl Diva card in a Debian Woody box with kernel 2.4.27, connecting to a Telstra (Australia) Onramp Home Highway ISDN line. I'm pretty certain the card and line both work since they've been used in this machine for PPP before this (but with an older kernel with DoV

Hi, i have a debian-system with the asterisk 1:0.9-1 packages and chan-capi 0.3.1-2 installed. My chan-capi seems to be out-of-order. Capi and I4l work in general. I can use isdnlog and capi4hylafax. Using chan_modem_i4l with the sam context work fine too. But no incoming calls are answered by asterisk over chan-capi. If have 2 isdn-card: a passive avm a1 (i4l/hisax) and a active avm b1