similar to: Using AVM Fritz!PCI as zap interface

Hi, is it possible to randomize extension which would be choosed by Dial()? I would like to forward phone calls to one of sales rep in randomized way (not to harm anyone;) ). tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89

Hi, I've seen some reports about ruuning Intel modem with 537 or MD3200 chipset running with Zaptel drivers as a FXO port. Did anybody managed to set up a PCI faxmodem based on Intel536ep chipset to work with * and Zaptel drivers? Modem seemd to work just fine with Linux, but the driver says no;) some more info: Linux 2.4.26 mazuchna:~# cat /proc/pci | grep 536 Communication

Hi, I am setting up a firewall on CENTOS 4.4. I have done default block iptables -P INPUT DROP iptables -P FORWARD DROP iptables -P OUTPUT DROP I have enabled ICMP to www.google.com iptables -A OUTPUT -p icmp -d 64.233.189.104 -j ACCEPT iptables -A INPUT -p icmp -s 64.233.189.104 -j ACCEPT Ping works fine as below [root at firebox rc.d]# ping 64.233.189.104 PING 64.233.189.104 (

I was able to remove the NTFS partition and repartition the HD as a type Linux (83). Next I created the ext3 file system. If I reboot with the USB drive attached, the boot process hangs, however if I turn off the drive, boot and then turn it on, all is fine. # cat /etc/fstab shows the drive, but I am not sure how to use it for a backup nor do I know how to access it. Any help would be

Hello! I want to use "asterisk -rx "show version"" from a php script called in the browser using the local apache, which runs as user "apache". Asterisk is running as root. I added the following line to /etc/sudoers using visudo: apache ALL = NOPASSWD: /usr/sbin/asterisk When i am on the command line of my linux box it looks like this:

Hello! Is it possible to run Asterisk as a SMS Service Center and does it work over a directly connected ISDN (CAPI) interface card? Did anyone already use Asterisk for that? Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Thanxxxx! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hi to all, we are searching for a hardware based DSP solution for use with Asterisk based on PCI or MiniPCI to reduce main processor load and to use embedded boards with Digium E1/T1 cards like TE410P. does anyone know about any manufactorer of those cards or someone who is able to develop/build such cards? Specifications: PCI or MiniPCI up to 120 concurrent transcodings Codecs: G.729/G.729A or

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a

Hi to all out there using HP DL380 G4 servers, i found a way to get the Digium TE410P with older firmware running on a HP-Compaq DL380 G4 Server! Here's the step-by-step description: 1. download the latest BIOS (in my case it was 4.04 from date: 06/02/2005) for the HP-Compaq DL380 G4 using the "Systems ROMPaq Firmware Upgrade Diskette for HP ProLiant DL480 G4 (P51) Servers" Link:

I have a plain vanilla Centos source (not 100% sure where we got it from), but I can't seem to get a build that installs correctly. I tried the method described at http://wiki.centos.org/HowTos/Custom_Kernel, but rpm doesn't like the output file, and even if I install it with -F, nothing seems to happen - no new kernel, now initrd, nothing. When I tried following the directions in the

HI, basic question. I've got a TE110P card and I'm trying to set it up with ztcfg with polish zone. ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 206: Unable to register tone zone 'pl' I've got loadzone and defaultzone set to pl, and there is a definition of that zone in zonedata.c but it doesn't work. Any hints? tia

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, July 03,

Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

You can send both paties to a meetme conference with Manager Redirect. Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -----Original Message----- From: Roland Zagler

Hello everyone! Has anyone had success in configuring the "Fritz!Card DSL" card, a combination of an ISDN adapter and a DSL modem from the german AVM GmbH (http://www.avm.de/) in a dom0 Domain (Xen 2.0.7)? I had success in compiling the modules from http://www.avm.de/de/Download/index.php3 and they load fine with "modprobe fcdsl". But with the Xen dom0 Kernel they seem to

Hello all, I would like to know about the support for this ISDN passive card AVM Fritz-PCI v2.0 with Asterisk: 1. which drive should I use with Asterisk: isdn4linux or capi4linux from AVM (I read some messages on the asterisk-users archives but I didn't find the answer) 2. if the drive is isdn4linux then HiSAX support (for passive cards) is enough ? and the device /dev/ttyI0 (for