similar to: ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P

Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs:

Hi, Is it possible to use 4 TE405 boards in one server ? It mean, to have 16 E1s on just one server. Can somebody tell me how many boards is it possible to have on one server ? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060605/29e9d42d/attachment.htm

Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA

I have a TE405/TE410P card that was working on 1.0.X I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to 1.4.5 and libpri. I copied all the zaptel and zapata and extensions.conf files from 1.0 I did update extensions.conf from 1.0 to 1.4 commands. I cannot get the card to work in 1.4.10. AHHH! I see with zttool that the T1 is in Green, I see calls coming in as the bits go

I'm trying to apply a function defined in the VW R docs, that attemps to convert a data.table object to Vowpal Wabbit format. In the process i'm getting the error in printf mentioned in the subject. The original function is here: https://github.com/JohnLangford/vowpal_wabbit/blob/master/R/dt2vw.R Below there is a small example that reproduces the error. The function works great with

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Currently I only have one machine backing up to another. I am backing up to a filesystem that is mounted as nosuid. The syncing process works perfectly except it fails on files that are suid. Example proc/ root/ tmp/ mknod tmp/mysql.sock : Invalid argument mknod tmp/orbit-daud/orb-12573780511489051058 : Invalid argument mknod tmp/orbit-daud/orb-139425434385535813 : Invalid argument mknod

Dear R user, After I imported data (csv format) in R, I called it out. But it is in non-numeric format. Then using "as.numeric" function. However, the output is really awful !!!!! > PE[1,90:99] V90 V91 V92 V93 V94 V95 V96 V97 V98 V99 1 16.8467742 17.5853166 19.7400328 21.7277241

Hello, Does anybody know how I can configure icecast so it doesn't disconnect a client which "has fallen too far behind". We are using icecast2.01 streaming software for our church home-listening system. Our client we use a netgem-netbox. We are streaming 25kbps MP3 on V90-56kbps modem connection After around 45 minutes after starting the cleint the icecast error.log reports:

Hi All, i'm going crazy trying to make asterisk work with the following hardware: 02:05.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 17 I/O ports at d000 [disabled] [size=8] Memory at ec000000 (32-bit, non-prefetchable)

Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my Asterisk 1.2.1 box. Also when compiling Asterisk 1.2.5 and tried to run it with the Asterisk-addon,

Hi. I have a simple (I think) question My dataset have these variables: names(data) [1] "v1" "v2" "v3" "v4" "v5" "v6" "v7" "v8" "v9" "v10" "v11" "v12" "v13" "v14" "v15" "v16" "v17"

Hello. Has anyone been able to successfully make data (dialup modem) calls through a TDM2400E? We're able to make fax and credit card calls fine, but cannot successfully make modem calls using a 56K modem connected to a patch panel connected to an FXS port which then gets bridged to an FXO port connected directly to a phone line. We have 'echocancelwhenbridged=no' set in

Wouldn't it make sense to have this in the man page? The 8192-byte limitation for 'fmt' is mentioned but not this one. Thanks, H. On 08/25/2015 02:08 AM, Prof Brian Ripley wrote: > From the sources: > > #define MAXNARGS 100 > /* ^^^ not entirely arbitrary, but strongly linked to > allowing %$1 to %$99 !*/ > > > > On 22/08/2015 04:21, Martin

I was playing around with the Flash Operator Panel, and came smack into a brick wall. We have a * box linked to a legacy Meridian System using a EuroIDSN link (TE405p) with 10 channels enabled. I also have several SIP extensions. What I wanted to do was to have a button for each of our (say) 32 users, 5 of which are on SIP. That leaves the other 27 on Zap. A potential of 27 users on 10

Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. PRI/T1----- | | | ---------------- | | | Channel Bank | | |

I'm unable to get faxes working over tdm400p (4fxs modules) Too many errors sending and receiving faxes with an analog fax 1) echocancel=no on the zap channels 2) ztmonitored the channel for a good/low audio volume I'm trying to send fax between zap fxs channels. No way to get it working right Has someone else the same problem?

Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061031/0f74ec67/attachment.htm

Hello, i have a little Problem with compiling asterisk-addons the failure is: app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: ?SR_LIST_REMOVE' undecalred (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is

Hello, I get an Error every minute on the second card of two installed TE410P Cards in our System. The error is: PRI got event. HDLC Abort (6) on Primary D-channel of span 5(-8) PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 5(-8) Is it possible that there are known problems with 2 cards in one system? I'm running Asterisk/Libpri/zaptel from SVN branch-1.2-16008 I was