similar to: CLIR and isdn4linux

Hi, Apologies in advance for the lengthy email. I'm new to asterisk and have trouble with isdn4linux. The setup is very basic like this: winxp ------- asterisk -------- winxp x-lite | x-lite | pstn The hardware involved is: Compaq EVO with RH9/kernel 2.4.20-30.9. Fritz!Card PCI v2 Asterisk CVS-04/17/04-21:36:18 Basically

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, 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

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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello! After LOTS of research on this list and internet in general I managed to get an old Teles PCI card working with Asterisk throught ISDN4Linux. No echos, no delays, simply perfect -- electronic poetry ! :) eheheheheh I just didn't get it to work with CAPI and "chan_capi" but, since isdn4linux is doing such a good job, I'll

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

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) --

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

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

Hello all, This afternoon I had a BRI line installed by Telstra (our telco in Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux driver. Incoming and outgoing calls with Asterisk work fine (and with no echo - my main reason for getting ISDN). However, I can't seem to get outgoing DTMF working (incoming works fine). I made a call from my desk phone (Cisco 7940G)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm having some problems using the asterisk modem driver with an ISDN4Linux card (AVM Fritz). (I realise that it's better to use CAPI, but unfortunately this card doesn't have any CAPI drivers). Every so often the ISDN just stops working (it neither dials out nor accepts incoming calls). Trying to dial out just logs: --

Hi everybody ! I've a Pentium IV with a USR ISDN PCI Card W699. Card I've installed asterisk 1.0.0 and it's working fine with outgoing call to POTS But it never answers to my incoming call from POTS and I don't understand why. I tested the modem also with the minicom utility and I never saw the "RING" status It seems that the modem didn't know somebody is

Hi all, I'm playing with Asterisk and I've already configured all needed .conf files. It works quite well, but now I need your help to tune the system: when I place a call from a softphone to the PSTN, I can't hear directly Telco's tones and I can't use its services, e.g. a mobile's answering machine. I don't know if I have to modify the dialplan or if it depends on my

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

Hello all, I currently have an Eicon Diva Client isdn card using i4l. Outbound dtmf doesn't work (and never has), but there has been an annoying problem with false dtmf detection in calls (that could be triggered easily by blowing into the receiver on the remote end). I looked through the list and found two patches that need to be applied - 1 to isdn_tty.c in the kernel, and another to

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, 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,

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now