similar to: (euro)ISDN: complete silence / can't hear a word.

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

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:

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

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

Well, it has been a long time since I used my NetJet S cards with asterisk (moved to the quad PRI card) but I am trying to get this working again for home use. Basically, what I am using is this: Linux 2.6.8.1: config lines: # ISDN subsystem CONFIG_ISDN=y # Old ISDN4Linux CONFIG_ISDN_I4L=y # CONFIG_ISDN_PPP is not set CONFIG_ISDN_AUDIO=y # CONFIG_ISDN_TTY_FAX is not set CONFIG_ISDN_DRV_HISAX=y

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

Hi all! I'm new with telephony cards and DAHDI drivers. I have installed Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. I'm following the installation guide of Atcom [1] for AX1600P analogic card, modules are loaded [root at pbx ~]# lsmod | grep -E "hisax|netjet|dahdi" netjet 14618 0 isdnhdlc 4523 1 netjet mISDNipac

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

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

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,

Hi, I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP). The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss") which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1". Everything works fine except that I can not see the called number/MSN of incoming calls within Asterisk and because of this I can not route incoming calls

Hi all, Here another guy working with ztdummy and having problems with music. MP3Player does not work for me both from a telephone entering through a passive ISDN-adapter as well as a SIP-client in the LAN. Ztdummy works with conferences. Here are my messages. I have seen several discussion threads in the mail-archive regarding this subject, but could not find an answer. Is this something I

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

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

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 --

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

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

Hi, I am using a Netjet-s ISDN Card, and am having some trouble dialling out (Incoming Works Fine). TRUNK=Modem/ttyI0 exten => _90XXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm) exten => _90XXXXXXXXX,2,Congestion I get the following when diallingout: -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "Modem/ttyI0/04XXXXXXXX||Ttm") in new

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

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