similar to: PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM

Hello! Is there a way to use AVM Fritz!PCI as a ZAP interface and have it configured for ZAP channels? Thanx in advance! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hi I got a simple patch (merely extra PCI IDs for Junghanns douBRI miniPCI: https://issues.asterisk.org/view.php?id=18169 Sadly the reporter did not follow up on the report. I'd like to get at least one positive report before I commit this. Anybody has this device and wants it to work with DAHDI? Please update the bug report. -- Tzafrir Cohen icq#16849755

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

Hi, I don´t think that this is a ext3 Problem, since you have problems partitioning that disk. Perhaps a hdparm -i /dev/hdX could shine some light on this, also a fdisk -l /dev/hdX could be useful. For now - let me guess: <guess> To me, it seems, that the disk is clipped to 32GB, perhaps the mainboard bios can not cope with a disk >32GB - just a thought. Since the manufactorer

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

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:

Hello, We're about to deploy an Asterisk system. So far we have the following (below) configuration but before we start anything we would like to hear some suggestions on it, specification or recommendation too... Thank you! miniITX board dual core CPU 2 LAN 1x IDE (or more) 2x SATA (or more) 2x PCI slots (must work with Linux correctly in regards to IRQs) Optional: On board hardware RAID

I have an IBM Thinkpad with this beastie in it. The card was immediately recognized and uses the airo driver. No matter what I do, I can't get it to successfully connect to my WAP. Sometimes it sees the access point (as evidenced by an iwlist wifi0 scan seeing the network) and sometimes I can actually get it to connect ( as evidenced by the iwlist command showing the WAP ID).

Changes at my day job (we have a big merger being finalized), has me in the market for a new laptop. They may require me to only run the approved XP build on the system they provide. I do not know if they will fund another unit for me or not, so I am looking to go cheap. Here is what I am looking for: Runs Centos (duh!) Supports Suspend to memory as well as to disk. Bootable from

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

Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent "generic" board out there that I could be investigating? It would be such a shame to waste a PCI slot that

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

At 01:54 PM 1/8/2015, John R Pierce wrote: >On 1/8/2015 11:32 AM, david wrote: >>The price point of Intel's NUC unit makes it attractive to use as a >>server that doesn't have significant computational load. In my >>environment, a USB connected hard-drive could provide all the >>storage needed. I wonder if anyone has had experience with it, and can answer:

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To: