similar to: Cisco ATA Advanced CallerID

Good day, I want to have Asterisk as my gateway to the outside world and use another PBX to connect my existing phones. How do I specify the dial string to forward the original Caller ID to over the ISDN to the second PBX? Right now, my extensions.conf looks like this: exten => ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN} How do I transfer the caller Id information initially coming

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/

Hi all, i tried to write a call generator script. It generate i file like sample.call in ast src tree. But when i set the callerid and make a call over Modem[i4l] then the caller id is not set for the outgoing call. Is it impossible to set the callerid at this time for ttI devices? Thanks for help, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing.

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

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between

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

Is there any kind of Configuration guide available? I've been trying to get a SIP Soft phone to work and all I get is: NOTICE[81926]: File chan_sip.c Line 4716 handle_request): Registration from '<sip:xxx@xxx.xxx.xxx.xxx> failed for 'yyy.yyy.yyy.yyy' I know I'm missing something in the configuration. But don't know much about * yet. Thanks, Kyle --------------

Can anybody recommend a reseller in Europe (Netherlands) for modules for the X100P (FXO and FXS modules)? Cost, support are important. Also, what is a reasonable price for an X100P in Europe? Is there a difference in price between OEM and Boxed versions? Thanks, Pieter

Hi folks, I just used cvsup to upgrade my system to 4.8 stable. Now I have problem with gdm. When I have logged in as other user, I will log in as root. That happens when I automatically start gdm after booting. But if I start gdm myself after boot, I won't have this problem. I don't know why. Does somebody have the same problem before? Thanks! Peng

Hello everyone, Last week the CVS version of tinc gained the ability to run under Windows, in a Cygwin environment (http://www.cygwin.com), making use of the virtual network driver from the CIPE project (http://cipe-win32.sourceforge.net/). Preparations were also being made to support Windows without Cygwin. Today James Yonan of OpenVPN (http://openvpn.sourceforge.net/) anounced a port of

Hello-- I've been playing with the privacy options on my home/home-office system since August last year, and have some results, gleaned from my CDR records, which over the last 13 months, number a total of 8672, which includes incoming, as well as outgoing calls. Before I start spitting out numbers, let me note that with the current setup, I haven't had to tell a single telemarketer

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

We have a little soekris box running freebsd that uses racoon for key management. It's used for setting up an ipsec tunnel. I noticed that one of these devices lost the tunnel this morning. I looked in the log and saw this Jul 23 01:37:57 m0n0wall /kernel: pid 80 (racoon), uid 0, was killed: out of swap space I reproduced this problem using this code. #include <stdlib.h> int