similar to: incoming & outgoing call

Where can I get some asterisk books.. or tutorials..? I?ve been searching in google.. but I find just some tutorials explaining how to fast set up an asterisk server. I want to learn how to use it and how to make my own configurations. So, the thing is that I want to know what is the best book or tutorial that you know? recomendations? Thanks to everyone... Danko Miocevic

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 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

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

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 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

Hi, I'd like to make small 20 users setup with BTs. I'd like each of them to have its own context (for recording prompts, conference, ...). For them to have same extensions I should put them in separate contexts and let BT call them offhook. But these contexts are pretty similar (for instance dial to conference on 5 goes to different conf. number for each user, ...) How could I describe

I can make them logon but how do I make it so if I dial a number it will make the other one ring, is it under the sip config? what shouuld I include so for instance if I dialed 1111 it would ring phone 1 or 2222 would ring phone 2. thanks -chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

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I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

I have a SIP phone that doesn't support three-way calling. Is there a way to do three-way calling from a SIP phone server side instead? TKS __________________________________ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250

I'm curious is it possible to direct a call to an extension that takes you straight to music on hold, but NOT the standard music on hold. The boss suggested something he wondered if it was possible. Example: Someone calls (Telemarketer), we answer tell them to hold while we 'redirect' them to extension (Someone Important) 666 which is a separate music on hold pool of mp3's from

Hello: I have set up samba PDC using samba 3.0.11. When I use the usrmgr.exe tool on NT4 machine to add user to the domain, it prompts and says: "The following error occurred changeing properties of the user mary: fail find the group" (The error msg actually appears as Chinese in my computer, I just translate the error message). Then I check with the samba log file and it said: group

Has anyone had any echo issues with the Digium TDM02B FXO card? I purchased a clone wildcard on E-Bay for $8.00 and have had horrible echo issues.... I'm assuming it is because of impedance issues? Just wondered what people's takes were on the TDM02B FXO and echo?

I have an installation that connects in a [very] good day at 22kbps, but the normal is about 18kbps. I use de ILBC codec, and also change in iax.conf the trunkfreq = 20 to trunkfreq = 30 It works, you can understand well the other person, but don't expect miracles or an outstanding sound quality. > Dear Dan; > > Thanks alot for your kindly reply. > > Well, what u advise us

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400