similar to: Modem channel in Asterisk

Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute

Hi to everyone! I am new to this list and i want to greet you all! I am trying to install Asterisk on a Fedora Core 3 Distribution with a Digi Datafire Micro V ISDN Card. I know that I have to use the ISDN4Linux so in modem.conf I set the I4l driver ,the msn and device. Modem.conf [interfaces] context=remote ;driver=aopen ; modem by AOpen driver=i4l ; isdn4linux - an

Hello List! I finally got asterisk with capi working, and its already answering my call as well! :) Now i would like to call a number from my shoft phone (kphone). This is my extentions.conf: --- [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password

I finally picked up a "real" modem (not a winmodem) and put it in my machine. I pulled down the hsfmodem driver from the source, got a license and installed the driver at full power. If I use wvdial, I can get out through the modem, so I know that works. However, sendfax is rejecting the fax capability (from /var/log/sendfax.log): 11/30 00:31:52 sendfax: interim release 1.1.33-Apr10

I'm still playing with * at the moment. Just trying to figure out how much it can do. Anyway, it looks like * supports voice modems. So here come the obvious questions... How well does * support vmodems? How do I set * to use my vmodem for incoming calls? Has anyone gotten it to work? Thanx, Mike Diehl.

I have a modem on a port of a channel bank dialing out a FXO port to the PSTN. It wont train at any speed. If I put another modem on a another port of the channel bank, it will connect to it, but at @31Kb. Anybody try this and have it work ?

I've tried this before, with no luck. I've got to try again this evening, and I'm looking for some help. Here's my configuration -- pretty simple, really. Asterisk box -> T100P -> TA750(20FXS/4FXO) -> phones and outgoing lines I have an analog modem (Ok, it's a TIVO) that I need to be able to dial out. Right now, I have the modem connected directly to an outgoing

Hello, I am new to Asterisk, so RTFM answers welcome too (just include the FM's link :). I'd like to build a simple dialout server based on Asterisk. I installed 0.4.0 from package (a Debian SID machine, "server"). The client is gnophone (a Debian SID machine too, "client"). My modem is a GVC 56k voice modem connected to the server's serial port. I modified

Which modems are supported by Asterisk for full-duplex (as a VoIP-PSTN gateway) operations? I've tried Asmax Mustang (based on TI RK 56000 chipset) but it's 4-bit only, also tried Zoltrix on Rockwell 56k chipset, hacked a lil' bit chan_modem_aopen (so it chooses 7200Hz instead of 8000), sound was 'understandable' but only one-way :/ .Does anybody has experience in using

Hi, I have a strange problem on some isdn modem channels. (* 1.0.9 / chan_modem / 2xHFC-S cards). Everything works fine but when the 2nd (and 3rd etc..) call comes in and * answers and there is about a 1/2 second of sound from the previous call (ivr) before the sound from the new call is heard. It just sounds bad and is quite annoying. I am assuming this is sound that is still in a buffer

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to

Hi guys, I think I posted on this issue before, but didn't get a response. I've still not been able to resolve the issue. I've got a small installation of Asterisk running one 4 port FXS Digium card and 1 FXO Digium card. I'm having difficulty routing modem call through one of the extensions out through the FXO card. By difficulty, I mean it won't work. The calls

Hi list, I've browsed through the wiki pages and finally would like to get some answers from peple who perhaps had used such card. I have the sangoma s508 card. It has no CSU/DSU. It is an ISA card with DB9 and DB25 connectors (for RS232 and V34/X21/EIA interfaces). They claim it is a frame relay and syncronous ppp card. Can I use this card with asterisk in any way but without

Hey All, I'm still (since April) having problems getting RxFax to work over an ISDN4Linux channel. Just wondering if anyone has had any luck getting it to work? I have done a CVS update today (about half hour ago) and made sure I have the latest version of spandsp according to Steve's website (spandsp-0.0.1k). When I was compiling asterisk, I got the following warnings:

Hello, I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2 compiled and installed. I have modules alsa 0.9.8 compiled and installed My PC have and audio card ac97 chipset intel i810 in motherboard. The list of the modules loaded is: namor:/etc/asterisk# lsmod Module ? ? ? ? ? ? ? ? ?Size ?Used by ? ?Not tainted snd-pcm-oss ? ? ? ? ? ?35652 ? 0 snd-mixer-oss ? ? ? ?

I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk. Oh323 Module compiled without errors. But When I try to stary Asterisk with the Oh323.so file in the modules folder, Asterisk is dying with the following error. [chan_oh323.so]Aug 2 14:08:14 NOTICE[18873]: res_musiconhold.c:490 monmp3thread: Request to schedule in the past?!?! => (InAccess Networks OpenH323 Channel Driver) ==

Hi, I was using Asterisk with Hylafax via IAX Modem. It works fine until I upgraded to Asterisk 1.2.7.1 I didn't change any configuration but it seems that Asterisk does not get the call from IAXModem anymore. I'm doing something like this Asterisk <--> IAXModem <--> Hylafax Usually when I use sendfax -n -d 260XXX somefile I'll see Asterisk receiving the call in

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

Hi! I want to connect my asterisk system to the PSTN. Now I'm thinking about using an analog modem as FXO device. Which modems do run? How is the voice quality? (I looked into the wiki, but on <http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware> no regular modems were mentioned. Or should I use a special FXO-card? I don't want to spend 99$ for such a thing but

I want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? (On usual modems, I can type ATS11=50 to get tone dialing much faster (50ms instead of default 90ms). After that, I can write configuration to nvram (AT&W) to be permanent)