similar to: SIP<->*/ISDN<->PSTN calls almost impossible because of echo

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, 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

Sorry for a slightly off-topic question, but I'm getting desperate. I tried to use asterisk to talk w/ ISDN line and tracked the problem to i4l. I have a Diva PCI 2.02 (S/T) card and am running RH8 w/ 2.4.20 kernel (first version that supports the 2.02 w/ hisax driver). The switch is a 5ESS running NI-1 protocol. The card sync's up and reports other traffic on the line, but fails to

The card is an ASUSCOM ISDNLink PCI (passive) and the circuit is from Qwest (in the US). I will be using this circuit only for voice (I'm doing this because of the poor quality of my POTS lines). I've compiled Hisax (as a module) into my 2.4.25 kernel, and with 'modprobe hisax type=35 protocol=4 id=hisax' I get the following: Apr 7 10:34:24 dev kernel: HiSax: Linux Driver for

We have setup an asterisk box to let everybody call into the university internal network, but I get unexpected hangups when doing an outbound call from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the call. ----------the dial and the problem----------- -- Executing Dial("SIP/57966-a19d", "Modem/g1:96121||rt|") in new stack --

I have * with i4l installed and working - on a dumb eicon card. It seems in order to get DTMF out of the BRI (for business banking - etc) - I should change from i4l drivers to capi drivers. wiki help seems to be for the Fritz card only...??? I have ticked the suggested boxes in 'menuconfig', explored the capi4linux websites - etc... but am missing some magic. I've "modprobe

Hi There I am running SuSE Linux 9.3 and installed Asterisk from the RPM's. I have a Diva ISDN card, making use of the HiSAX drivers. Configuration seems to be OK. I can receive a call coming in on the ISDN and route it to a SIP (softphone) on my PC. As soon as I hangup the call, then asterisk stops working. I am forced to kill the process with -9, a normal kill does not work. The last

I just wanted to let you know that it's possible to use Eicon DIVA PCI 2.01 ISDN cards (not "server" divas) with asterisk. First thing to do is to load the module. If you have two of these cards, you should do it like that: modprobe -v hisax protocol=2,2 type=11,11 And now you can have up to 4 incoming calls with two cards (try calling yourself and see if anything gets into

I'm trying using Asterisk over an isdn line. I configured my Eicon Diva card by the hisax module and it works with the Asterisk's chan_modem (and chan_modem_i4l) driver. The problem is that I can receive calls (on /dev/ttyI0) but when I make a call I always have the line busy (on ttyI0 but even if I use an other device like ttyI1). This is a part of my consol output: --

I'm trying to get a basic Asterisk configuration together for ISDN incoming / outgoing calls. I have two Cisco 7905g phones working (at least talking to each other) and have purchased a NETjet-S PCI ISDN card for routing calls to / from ISDN. The state I've managed to get it to is:- -- Executing Ringing("SIP/PHONE2-d557", "") in new stack -- Executing

I need HELP pls! BRISTUFF: Bad Sound quality CAPI : PTP Mode dont supported mISDN : kernel is 2.4.x and not 2.6.x HISAX : PTMP ok, PTP incoming ok but in outgoing asterisk dont compose number(i listen dial tone and than i can compose number via dtmf) Asterisk CLI (g3 is group of Modem[i4l]/ttyI0 and ttyI1): Called g3: 3453444444

Hello, I am trying to use 2 netjet cards under asterisk and isdn4linux. I am having a hard time trying to get them to work in terms of dial out. Does anyone have a working config I could look at for even one card (tried that, not much luck either). When i dial out: -- Accepting AUTHENTICATED call from 172.16.11.2, requested format = 2, actual format = 2 -- Executing

Hi Thomas, I saw your mail to Asterisk mailing list regarding ISDN BRI problem. It was about answering the call. I'm facing a similar problem. I had briefed the problem below. Please tell me what are the parameters to check/change to get it working. I have Hisax ISDN BRI card. I configured extension.conf, modem.conf correctly and could make outward dialing. But when I make incoming call

I can't find solution anywhere. I googled and find people with the same problem but there was no answers on how to fix this. I have W6692 based PCI cards that uses hisax driver (card type=36). Card is working fine under asterisk with i4l modem driver for incoming calls. If I want to dial out using some sip phone I don't get ring indication. Phone is ringing and I hear only silence until

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

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

Hi, I have a problem that I just can't figure out how to solve. I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in * I get the demo-greeting, listen for a few seconds and hang up. I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should. Right now I have removed all codecs but codec_gsm.so

I have a North American BRI configured as National ISDN (NI-1) on an SBC (Nortel?) switch. SBC calls this "FastTrak ISDN", and it's surprisingly inexpensive -- less than two POTS lines. I've been trying to find an inexpensive PCI interface solution to connect this to Asterisk as voice lines. I prefer ISDN BRI over POTS lines because of the improved signalling, fast dialing,

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

G'day, New to Asterisk alert! I have a Netjet card running on linux 2.4.27 kernel using the HiSax module, and trying to use it for incoming/outgoing calls from *. I've tried playing with modem.conf and extensions.conf every which way I can think of, using samples and whatever I can find off the net, and I get the same message everytime I try to dial in. The complete message is: