similar to: Fake Ring Tone/Compile Addon

Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406

Dear All, Sorry to be posting again. I have solved my problem. The problem is that when exiting from the macro, the priority number is still in effect. For example, priority 1 is at the start before entering macro after the macro the priorty will be 2. Since there isn't any other dialplan command, the switch statement would be search for a priority 2 in the Realtime extensions table. One

Dear All, It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

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

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

Hello, I am pretty new to R. I am working on neural network classifiers and I am feeding the nnet input from different regions of interest (fMRI data). The script that I am using is this: library (MASS) heap_lda <- data.frame(as.matrix(t(read.table(file="R_10_5runs_matrix9.txt")))*100000,syll = c(rep("heap",3),rep("hoop",3),rep("hop",3))) library(nnet)

Hi All, I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

I''m starting to play with the tooltip.js library from http://tooltip.crtx.org/index.php?page=HomePage. The libarary seems to work pretty well except that when I use a tooltip on a web page that has a form and submit buttons, the submit buttons stop functioning and won''t respond to click events anylonger. So far I''ve only been able to reproduce this under Safari.

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

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 Guys, I''ve got a rails application that uses activeldap to authenticate users. Its been working great for about 6 months now, but a user from mexico started using it recently, but can''t seem to login. I''ve narrowed it down to the fact that his password contains a ñ character in it. This problem is also present in a test user account I''ve setup. ActiveLDAP

I'm looking to get something of a generic CLI prompt function/libarary into Syslinux for use in a COM32. I'm looking for the following: 1) Line Editing with proper wrapping. Currently, gPXE and HDT both have provisions in place for line editing however they both can not handle line wrapping appropriately. 2) Tab completion. Tab completion of filenames would be a goal but not a

Hi all. We are getting our first PRI line to use with Asterisk and one of the technical specifications is about framing, structured or unstructured. The main difference about them is almost clear for me: http://ckp.made-it.com/g704.html says: "G.704 is the framing specification for G.703. A carrier can 'steal' a 64kbps time slot (TS0) from a 2.048 Mbps line and use this to

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I

I'm sure that this question gets asked frequently, but a quick perusal of the list archives shows that it hasn't been asked in a least a month or so, so pardon any repetition. What is the current state of asterisk on linux 2.6? I ask, because I spent yesterday giving it a whirl, and everything seems to go just fine till the very last minute. Zaptel, libpri and asterisk compile just