similar to: Asterisk documentation..

Just to close the thread. The problem was that I was using an old version of the code. If anyone has the same problem, you can download the code from here: http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz Good luck, Danko ----- Original Message ----- From: "Danko Miocevic" <danko@santirso.com.ar> To: "Asterisk

Hello, I?m trying to use my phone with asterisk to get GSM connectivity but I can?t compile the code. I got the asterisk, chan_bluetooth, zaptel and libpri sources, the last two ones compiled perfectly. I have added this to the /usr/src/asterisk/channels/Makefile: include /usr/src/chan_bluetooth/Makefile and I?ve also added chan_bluetooth.so to the CHANNEL_LIBS var. When I do "make

I could use a few pointers to get this working please? I have asterisk installed on my linux server. It is setup to register with sipgate and works for incoming calls. I have xlite installed on my windows pc and this connects fine with the asterisk server and can get the incoming calls fine. Now I want to be able to make outboun calls from xlite via sipgate. I also need to be able to dial

Hi All, I've installed digium TDM02B. The PSTN line connected to this card. At the IP network side, I have SIP phones registered sucessfully to asterisk server. How do I configure asterisk, so once there is incoming call to the TDM channel from PSTN, the caller will hear another dial tone from asterisk then to key in the extension of intended destination (SIP phones number) that already

Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. Maybe what I want is too much, after all it is a open project, not commercial product. If I want to get

I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk don@moskaluk.com www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date:

How are you? Fristly I would like to introduce myself in a shortly. I'm a newcomer at Newcomtech Co,. Ltd. Nowdays I'm working at TDM40B and TDM04B cards. I have installed Linux hedhat 9.0 and Asterisk software. I have configured the cards. I typed asterisk -vvvc command in the command line and then asterisk give us "asterisk is ready". Now I have to configure my asterisk

Hello All, Please forgive the lack of understanding as of yet but I have been trying to follow the mailing list messages over the last few days and would like to know if someone could wither point me into the right direction or possibly give me a brief overview of the complete process. Basically, I see that the Asterisk PBX systems can run on linux and seems to offer the engine base that is

Hello my brothers and sisters, Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" and "Asterisk" Package? or i need also to other cards or packages? and if "X100P" card not suitable to VoIP, please recommend a another card, (please take in your account that i would like to connect standard analog line to the card

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

I am trying to get an initial setup up and going which I assume is a very common question here. My basic questions are the following: Can I get Asterisk up and going without voice cards using it with SoftPhones internally as a proof of concept. (just calling extensions and leaving voice mail) Is there a jump start config that would accomplish this? What is the recommended SoftPhone that is

Hello, Quick question. I have written a C function - I would like to make it return two vectors to the R environment - one STRSXP vector, and one INTSXP vector. Is this possible/ easy to do using the API? I looked, but could not find the answer to this question in the "Writing R Extensions" guide. Thanks very much for your help! Charles [[alternative HTML version deleted]]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1

Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2.

I have compiled chan_bluetooth and I have used it many times before... but now I have a problem and I can?t find a solution. I have a bluetooth dongle (MSI) connected to my asterisk and I use a V3 phone to make calls to another cell phones... the thing is that I just get one way audio.. I can hear people from my IAX softphone but they can?t hear me... I don?t know what could be.. anyone had

Hi, I need some information on DID and Callback. Please read-on: Question on DID (User1 Calling User2 via normal Telephone line and sending its CLI: Connectivity is as below: User1 ==PSTN==> DigiumE1/Asterisk1 ==INTERNET==> DigiumE1/Asterisk2 ==PSTN==> User2 1. Can User1 make a single stage call to User2 via Asterisk1? Currently User1 is able call User2 on Two Stage basis (Asterisk

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

ONLamp.com has finally published part 2 of my Rails article! You can find it here: http://www.onlamp.com/pub/a/onlamp/2005/03/03/rails.html It would be really nice if we could get this slash dotted again before the weekend. >From what I understand, you''re more likely to get accepted when multiple people submit something. So, to anyone who has the inclination: if you would submit

as i am trying to use asterisk and install my newly purchased ( got it yesterday) digium cards. i am following the very detail steps of http://www.automated.it/guidetoasterisk.htm. but one thing did not seems right so i wanted to let enveyone know the page says: Once compiled make sure there is a copy in /usr/bin/mpg123 i think the location is /usr/local/bin/mpg123