similar to: New to IP-PBX

I have delete=yes and attach=yes. But my messages are not getting deleted after they're sent. I'm running asterisk as root so it can't be a permission issue. Any ideas?

Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ? Maybe the Digium TDM410PLF card without any daughter card would do the job ? Thank you very much for supporting. Have a nice week-end, Mike

Hello, The company I work for deploys and manages telecom hardware for small- to medium-sized businesses. My boss has asked me to investigate Asterisk as a possible PBX for deploying to customers along with IP phones. The general layout would be: -------------- ---------- | trunk | ---------- | LAN, | | (T1/analogs/ |

I found an old Dialogic card in an abandoned PC, that I think is a Dialogic D/41EPCI based on some googling.. The lspci output says: 00:09.0 Bridge: PLX Technology, Inc. PCI <-> IOBus Bridge (rev 01) Subsystem: Dialogic Corp: Unknown device 0529 I'm just getting started with Asterisk to build a SOHO system. Any suggestions for what I can do with this card? It would be

I just found a Dialogic VFX/41JCT-LS (4 analog ports) in a drawer. I can use it in my house with asterisk at home project. Can I use that with asterisk? Where can I download proper drivers? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama -------------- next part -------------- An HTML attachment was

Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .....Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas

Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

Does anybody know if there is some tool to convert from WHIRL to LLVM? maybe some project under development? a similar project? Thanks > > --- Duraid Madina <duraid at octopus.com.au> wrote: > > Date: Fri, 18 Mar 2005 12:45:54 +0900 > > From: Duraid Madina <duraid at octopus.com.au> > > To: ahs3 at fc.hp.com, LLVM Developers Mailing List <llvmdev at

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Hi all, This is just to announce that I have a FreeBSD x86 machine running the test suite more or less continuously: http://kinoko.c.u-tokyo.ac.jp/~builddonkey/ Of note is that it now runs Dejagnu tests. (Thanks, tonic+co!) I'm also tracking CVS breakage (internally, for now). At some later point, I'll make available more real-time information on "is CVS alive and if not,

Hi, I have A101DE Sangoma Card( http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html ) lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card [root at asterisk ~]# lspci -vvv -s 03:04.0 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: Unknown device

Hi everyone, I've just checked in an IA64 backend to LLVM! Be warned, it's pretty rough right now. Here are some of the known defects: - No varargs - No alloca - No instruction scheduling/bundling of any sort ...or in other words, it breaks often and when it does work, it's a dog. On the plus side, it _does_ have a tasty new pattern instruction selector. :) Beyond fixing the

Oleg Smolsky wrote: > I've just tried building CVS/HEAD of llvm using gcc 4.0.0 that I have > installed to /opt/gcc ... then you should either add /opt/gcc/lib to /etc/ld.so.conf and rerun ldconfig, or add /opt/gcc/lib to your LD_LIBRARY_PATH . However, GCC 4.x definitely has issues building LLVM, at least on ia64. If you want to use LLVM in anger, I'd stick with 3.4 for now.

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

Can someone point me to a page about writing a text file to call an external number and play a TTS with cepstral? I know it includes the creation of a .call file but beyond that im a bit lost.

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these

Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C