similar to: voicemail.conf

Hi, I'd really appreciate it if you can explain this to me. I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. I'm new to IP Telephony and telephony and general and I researched a lot but still confused about what I really need. I know that I can setup an IP-Telephony for my LAN using a SIP server and SIP compatible software phones. But the challenge is how can I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I am running an * box with just 8 extensions connected to our old Alcatel BCN 5200 PABX. The requirement is that we now scale it up to handle about 300 lines and get rid of our old PABX. Is there a way of hooking up 300 phones to asterisk without going via the PABX. I am more of a network person than a telecomms one so i may not fully

I don't completely understand this.. Lemme try it out.. [default] exten => 1112223333,1,Macro(happy-did) [macro-happy-did] exten => s,1,Goto(${MACRO_EXTEN},1) exten => _XXXXXXXXXX,1,NoOp(Normal "s" exten stuff here) So when this is ran it will cut the cdr and the s will show the actual DID not the s correct? But then the NoOp would be something like: ....

All; I am running Asterisk 11.6-cert16 and I have voicemail setup so voicemail messages are sent as email attachments. That works fine. However, the body of the email contains the CallerID(name), but is missing the CallerID(num). For example, the email body looks like this: Just wanted to let you know you were just left a 0:21 long message (number 13) in mailbox 101 from WIRELESS

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

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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 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.

Okay, my last post for the night. So, after tweaking my iptables rules I placed a test call from my * -> Vonage softphone account, to my Vonage main account (which uses the Motorola device). Once connected I tried punching a few DTMF's to check the quality and found I had been transferred to a fax machine. ???? Did I find a secret entrance to some fax machine at Vonage, or

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

I created some scripts to logrotate. I am having a problem. After I do it, I am sending kill -HUP to the process its not using the newly created messages file again. Could someone help me out with how I can rotate asterisk's log's without killing the process? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.