similar to: Re: Euro-ISDN

asterisk-users-request@lists.digium.com is believed to have said: >chan_capi does not set the NT-mode. Your cards driver need to do that. >E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl >or set NT-mode in the config wizard. >chan_capi does not (need) to know anything about what protocol the card is >doing. CAPI is independent here. Ok.

asterisk-users-request@lists.digium.com is believed to have said: > >The active cards do the ISDN protocol stuff on board, so the host CPU/driver >does not need to do that -> better performance, less interrupts. >The AVM cards do not have such DSPs on board, so no echo-cancel. >But the Eicon DIVA Server cards do. They do analog Fax/Modem, echo-cancel, >DTMF-detection, voice

asterisk-users-request@lists.digium.com is believed to have said: >> While we are at the subject another couple of simple related question. >> >> Are HFC-S cards active? I got one for a very low price, so that I >> imagine it will be NOT the case... > >No, these cards are passive. The protocol handling must be done by >the hosts CPU / driver. OK; price

Hi folks! Our company are going to buy an E1 line with Euro ISDN and 30 lines (channels). This is how it will be configured: 3 Lines, of the total of 30, is going to be for the company phones, and share one phonenumber (eg. 555-12340). 1 Line will be dedicated to a specific unique phonenumber (Fax) (eg. 555-54321). The rest of the lines/channels (26) will be used by (by, not for) our customers,

asterisk-users-request@lists.digium.com is believed to have said: > >Hey Dan!! > >Give us a clue as to what hardware/setup & network provider you have there, >and we might be able to help :) > >Paul > Hello Paul, hello everybody! I have, too, an inbound call problem. I am using an ISDN Fritz Card PCI 2.00, together with chan_capi 3.5.x . As I call my number I get

Hello, I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550 PBX. As a BRI connection has 2 channels and allows 2 simultaneous calls, numbers/MSNs 6391 and 6392 were for provisioned for each channel. The system is working (partly, read on), the trick is the correct cable wiring and setup the PBX's port

Where can I get E1 and/or Euro-ISDN specifications/data sheets? Are there specs for other E./G./Q./etc. protocols as well? Thanks!

I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 My zaptel.conf file: -------------------------------- span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us

Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom

I'm trying to setup my Asterisk-PBX (using asterisk@home) to work as an ISDN-VoiP-Gateway. For this to work, I'm trying to setup a trunk using my Billion PCI-ISDN-Card. And yes: I'm an Asterisk-Newbie. ;-) I can't get it working and was wondering if there is someone out there with the same setup that could send me his modems.conf or tell me if there's anything special I should

The way I understand things, there's no way for a analog line to "reject" a call (give the caller an busy tone) if the line is not actually busy. Would a digital EURO-ISDN line give me this ability? Thanks, Cosmin Prund

Ok, now I'm really getting places. Something else has got me stumped - asterisk does not receive any callerid information. If I set up an extension as follows: exten => 444444,1,Sayalpha(${CALLERID}) exten => 444444,2,Hangup I get the application saying Executing SayAlpha("Zap/1-1","") in new stack which implies that the ${CALLERID} variable is blank. Do I need

Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain

this happened to me on a cvs update, rebuilt a clean chan capi cm and all is well. Greg ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Voicomm User Sent: Monday, September 19, 2005 3:29 AM To: Armin Schindler Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk ISDN: Problem

Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in

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

Hi all, sorry bothering again. I am still stuck in compiling asterisk. Learning (or trying to) from the first problem and first hint, when I got this error: gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm getting a strange problem with a Mediatrix 3631 Gateway connected to the PSTN via an E1 PRI link configured for Euro ISDN signaling. The Mediatrix sends incoming calls from the PSTN to an Asterisk server via SIP: this works fine. But when the caller hangs up, the Mediatrix doesn't send "Bye" to Asterisk, so the call is

asterisk-users-request@lists.digium.com is believed to have said: >I don't mean to be rude to everyone who responded to this question, but >I think that everyone is answering the wrong question. The point is that >the message waiting indicator doesn't light up, at all, ever. All that >happens when messages are waiting is that the display blinks and the >phone gives a

asterisk-users-request@lists.digium.com is believed to have said: >Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a >protocol for the phone to talk to any Bluetooth headset, no matter who made >it. This protocol would have to include something to allow voice to pass >from the phone to the headset and vice versa. It might also include >something for