similar to: Callerid Modem I4l and outgoing spool

Can anybody recommend an ISDN2e card that is a reasonable price. Also, does anybody know if the AVM B1 V4 card has the echo problems, I believe I saw somebody mention they had echo problems on the V3 card. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/c7cc9e15/attachment.htm

What is this? chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1 This is why I can't start asterisk in the background -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, please where i can get channel modem dsp patch - for correct use with i4l and patch to solve problem when the calling party can hear voices from last call, for a little moment regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hello, i have asterisk with 2 internal isdn cards - handled by isdn4linux and i need to setup whol system like this route some call beggins with 0 or 00 - long distance through first card, route calls to mobile network via second card ( tehere is isdn gsm gateway connected).how i can do this using only isdn4linux (/dev/ttyi) ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza

I used to get bad echo and also voice breaking up during a conversation, so I upgraded the machine (from a PII 300 to a Athlon 1.8GHz) and have resolved the voice breaking up issues but still seem to have a fairly bad echo on *both* sides of the conversation. Currently I am only using the system for outbound calls, so I am always calling from the digium USB port phone and out to the telco using a

To whom it might concern, The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7

okay.. i'm on my winbox, looking at stuff. connecting to my samba server works fine.. browsing through files and access is fine, but when i try to print, it doesn't work at all. it recognizes the printer, the printer type and such, but it just doesn't print. [printers] comment = All Printers path = /tmp printing = /usr/local/bin/lpr print ok = Yes

Hi. I'm working with i4l with asterisk CVS-02/21/03-13:59:12, plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19 patched to disable dtmf). All seems ok (apart some echo issues that seems gone with mec2 aggressive suppressor), but outgoing dtmf doesn't work . or at least I hear the very first part of the dtmf, but then it seems suppressed. here's my modem.conf [interfaces]

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Greetings, today I'm recompiling asterisk on a new machine with the same isdn card (not the same model, the *SAME* card) using CVS downloaded yesterday night. When I try to start *, it gives me the error: [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem

Hi, I'm trying to get asterisk running on kernel 2.4.20 however trawling through the archives I've found a few references to patches to remove i4l's dtmf detection, but have been unable to find the patch itself (I think it is isdn_audio.c). Can anyone point me in the right direction? The problem I'm seeing is connecting a SIP softphone (tried a few) to an external number via an

Hi all, i have noticed that i can't hear a ring tone if i make a call from my TDM40B to a chan_modem_i4l endpoint. I had a look in the code from chan_modem_i4l and there is a function calling "i4l_handle_escape" that gives a AST_CONTROL_RINGING frame back. But this seems not work ...(or i4l is not signaling it ?) Til now i have used the Dail app like Dial, Zap/g1:XXXXXX|60|r so it

Hi, Anyone using ISDN4Linux (Eicon Diva Hisax ) card. If yes, please help me out. After configuring extension.conf and modem.conf I could make outward calls correctly from gnophone and kphone. Still the inward call to the configured MSN is correctly reaching Asterisk and also to the configured Context. But the issue is, it was hanging on 'Answer' application and throwing out

-------- Original Message -------- Subject: Answer App hanging in I4L Date: Tue, 25 May 2004 13:49:50 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com Hi, Anyone using ISDN4Linux (Eicon Diva Hisax ) card. If yes, please help me out. After configuring extension.conf and modem.conf I could

Hello everybody! After checking the complete wiki and the mailinglist archives I still haven't really found out why the following constellation does not work. We have an asterisk-System with some SIP-Phones and an old ISA Fritz-ISDN-Card used with i4l. The whole system is integrated in out (ISDN-)PBX for testing. The ISDN-Card is properly configured, as we are able to phone out, receive

Hi, will spandsp work with i4l, or should I use capi? tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89

I have been trying to locate the patch that is supposed to cure the problem of hearing sound from the previous call when dialing through i4l and an hfc card. Does anyone have it? It is mentioned briefly in this post: http://lists.digium.com/pipermail/asterisk-users/2003-February/007530.html Thor

HI I.m working on echo on my asterisk and I'm wonder if is it possible to use Zaphfc (BRI) exactly like i4l? How to attach msn number to such card? Thanks in advance for any help. Regards, Corvin

God save this mailling list :) Which is the best settings for the best quality of Audio ? I'use isdn4Linux driver and SIP client but bad quality PSTN <-> Asterisk and SIP <-> PSTN

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