similar to: Asterisk & Euro-ISDN

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 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,

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

-----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

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

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

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

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

asterisk-users-request@lists.digium.com is believed to have said: [setup tool] >Sorry, I cannot answer that one. I don't know enough about these cards and >their drivers. Armin, thanks alot. One has to do some research and experimentation on his own every now and then; and see if there is anything interesting that might even end up in the wiki... ;-) >> So in the end there

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

Hello everybody, I have installed the Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new

Hello everyone. My land-line provider (Romtelecom) has a very nice offer for ISDN. All in all they offer me a digital land-line with 1 base number + 2 MSN's and that would make a grate addition to my full-time home office. Romtelecom say they're providing EURO-ISDN and the line is compatible with any euro-isdn compliant equipment. They say they'll install a NT at my office and this

Hello to all, I'm trying to build a PBX using Asterisk. I have a single BRI ISDN line and I need to connect 4 internal normal phones and a couple of softphones on PC. I have bought a single port Billion S0 card and a TDM400 with 4 FXS modules for the intenal phones. ISDN lines here in Greece terminate to an NT1 terminal called NetMod

Hello, After searching the archives, I''ve found only two messages regarding this case, but there were no solution. I have a similar problem, so I try to ask here. I have twi HFC chipset ISDN cards in my pc: 0000:00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus

O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling me while startup: [chan_capi.so] => (Common ISDN API for Asterisk) Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr1 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr2 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr3 Jan 24 14:30:47

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

I have installed a Billion ISDN card in my Asterisk. Calls between sip and isdn work. The i4l channel has MSN=26. I also put incomingmsn=26,27 in modem.conf. extensions.con: [incoming-isdn] exten => s,1,Dial(SIP/701,20,Ttr) ;will make my extension 701 ring, while exten => 26,1,Dial(SIP/701,20,Ttr) ;will cause an error message in the Cli interface: WARNING[196621]: pbx.c:1814

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