similar to: [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone

Hi Martin, I saw your problem listing on the Asterisk mail archives. I seem to have the same problem with the ISDN 'lacking dialtone' message I still have not been able to get it working, could you share your modem / extension / sip conf files? Thanks in advance! - Patrick --------------------------------------------------- This email was scanned by MyMail of

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 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 ---------modem.conf :---------- msn=240862922 incomingmsn=240866365,6365 device => /dev/ttyI2 group=1 device => /dev/ttyI1 ; ttyI3, ttyI4 ---------extensions.conf ;------- [sip] exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076xxxx result : --Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack -- Called g1:024076xxxx --

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 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

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) --

Hi, I've been playing round with Asterisk on Redhat 9 (2.4 Kernel) and was experiencing bad echo problems using Budgetone 100's when calling analogue lines in uk (Isdn4Linux / Digi Datafire). Calls to other ISDN and mobile network seemed ok although not much testing done. I've tried installing Asterisk on a faster processor (P4 3.0 GHZ) with a 2.6.8-24 kernel to see if that helps.

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004",

hi, I am new to * and setting up a test system. here my setup : - debian (from knoppix 3.3) - Asterisk 0.7.1 (from the debian package) - AVM Fritz card used with i4l - softphone I use for testing SJphone on windows - I can make great softphone - softphone calls - I can call from an outside line * and get connected to a softphone here my problem: I can not make outbound calls. I place a call

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

I posted a message a little while ago but got no response (that I can recall), I've also seen other people mention this issue. Basically, when you have music on hold, it doesn't play the music on hold, the debug info shows it is starting and then stops straight away.. # My extensions.conf ... exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten

Hi, i install the Asterisk PBX on a linux machine with i4l to connect to PSTN (EuroISDN). And i configure a very simple dial plan in extension.conf. After this, i connect with a SIP program to asterisk and would call my cellular phone, but got this error: -- Executing Ringing("SIP/ACzerniak-0904", "") in new stack -- Executing Dial("SIP/ACzerniak-0904",

Hi, try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. A known problem ??? Thanks in advance Michael i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus) and oh323-0.6.0 Here are my config's ############## # modem.conf # ##############

Hello! Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl over POTN. Yesterday i configured asterisk to be able to hear a song (with Playback() command) with iaxcomm, and it was wonderful. Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a plain HFC card) with asterisk. i configured modem.conf this way, after reading

In this message I try to summarize what I have learned in these last two weeks. My primary sources of informations were the * list archives and linux ISDN docs. I ain't no * master, so don't trust too hard. Relevant messages from the * list for the current discussion are: 009177.html 009268.html 0498.html 0849.html My setup is an Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2)

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

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 passive isdn port configured in modem.conf in extention.conf i use this two channel (ttyI0 and ttyI1) with the string: exten => _NXXXXXXXXX,1,Dial,Modem/g1:${EXTEN}|60|r how can i hide my msn? is it possible to activate the clir with the @ before the ${EXTEN}? thanks Cristian