The basic Asterisk VoIP

Hello,
I am under debian on a Dell, I use a card Digium TDM400P with 1 FXO + 1 FXS module and I have a classical analogue FT. Foremost, it must physically install the card: just plugger card in a PCI, then you must install the card drivers. It is managed by the Zaptel driver.
Here is how you can do this

Code:

# Apt-get install zaptel zaptel-source linux-headers-`uname-r`

you must provide dependencies that must of course accept. Therefore, we have the tools zaptel, source code drivers, and our dear linux-headers.
to compile the driver for the card:

Code:

# Module-assistant auto-install zaptel

module-assistant and starts preparing a package that installs itself thereafter. To verify that everything is ok, do this

Code:

Ztcfg-vv Zaptel Configuration ====================== Channel map: Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart ( Default) (Slaves: 04) 2 channels configured.

In my case, I see the 2 modules I mentioned above +. It acts in fact the type of signal that travels over the modules. wholesale shipments of FXS FT us from their central, and our standard phone listening FXO (they are a little taken as males and females). Basically, an FXS module will be useful only if you want to connect an analog phone directly on your server (in which case you must connect an additional power module FXS PCI card). If you got up there, your hardware is installed and you can enter the heart of the matter.

Code:

# Apt-get install asterisk asterisk-doc asterisk-prompt

There 's other packages asterisk for a multitude of functionality such as: faxing, voicemail, voicemail, but for now, already use the basic elements
after apt-get to do his work, test that the server is correctly installed

Code:

# Asterisk-vvvc

there, you must have a whole bunch of lines that scroll across your screen for a prompt Style:

Code:

Asterisk Ready. * CLI>

Go out and stop the server, there are several methods:
ctrl + c (a bit brutal for a service)
gracefully stop (stops the server is no longer used)
now stop (stops the server immediately), your server is running, there is more to configure it, I'm in testing period and I discover the functions of a little asterisk + every day. I use Twinkle to phone headset with microphone + (apt-get install twinkle) I have at my disposal a hardphone. I do not use the FXS module for the moment because I do not have enough power cable.
/ etc / asterisk / files are all server configuration, there are a lot ...
I have configured as 3:
- zapata.conf (asterisk indicates how to use the modules of the card)
- sip.conf (states rights and the configuration of each IP phone)
- extensions.conf (Asterisk indicates the actions to follow when it gets a request)
In each file, there are plenty of examples for configuration help. it is best to save before doing anything:

Code:

# Cp / etc / asterisk / zapata.conf / etc / asterisk / zapata.conf.OLD # cp / etc / asterisk / sip.conf / etc / asterisk / sip.conf.OLD # cp / etc / asterisk / extensions.conf / etc / asterisk / extensions.conf.OLD

Here are the three files once configured

Code:

; Configuration file zapata.conf ; [trunkgroups] ; AC is pointless ... ? [channels] language = en; definition of the default language = switchtype national to the analog line rxwink = 300; Atlas Seems to use long (250ms) winks usecallerid = yes; use of caller ID hidecallerid = No, do not hide the caller ID callwaiting = No; "no" in the config guillaume usecallingpres = yes; number of active presentation callwaitingcallerid = yes, communication is waiting caller ID threewaycalling = yes; Conference to 3 transfer = yes; transfer allows canpark = yes;??? cancallforward = yes; forward the call is authorized callreturn = yes; ringback active echocancel = yes; attenuation of the active echo echocancelwhenbridged = yes; attenuation of the echo in bridge mode active ; rxgain = 0.0; gain reception, default: no comment ; txgain = 0.0; gain Emmis, default: no comments immediate = no;??? busydetect = yes; default: comments busycount = 4; default: comments ; definition of channels , the configuration of the line analog phone ; = fxo_ks signaling ; callerid = "asterisk fxs" ; context = from-sip ; channel => 3 ; configuration FT telephone line signaling = fxs_ks = callerid asreceived context = FXO channel => 4

as you can see, I have not yet entered into operation, but is commented for later, to respond.
notice any of the same context = FXO. this context is defined in the file

Code:

extension.conf: ; Extension.conf file for asterisk , [general] static = yes; interidiction modify this file WriteProtect = yes; interest to find, protect in writing autofallthrough = yes; if = No, problem when no corresponding action clearglobalvars = No; variables are conserved [globals] CONSOLE = Console / dsp; Console interface for demo ; CONSOLE = Zap / 1 TRUNKMSD = 1; MSD digits to strip (Usually 1 or 0) [from-sip] exten => 520.1, Dial (SIP/520, 15, tr); hardphone IP exten => _0., 1, Dial (Zap / 4 / $ (EXTEN), 45, tr); No meeting starts at 0 exten => 521.1, Dial ( SIP/521, 15, tr); IP softphone [FXO] ; exten => s, 1, Goto (context, extension, priority) exten => s, 1, Goto (from-sip, 521.1)

- if you have received a request for No. 520 then opens a communication using SIP appointed to the post 520
- if You receive an application for a No. starting with 0 then opens a communication using the zaptel driver to No. $ EXTEN (No. complete).
- If you receive a request for 521 then opens a communication protocol using SIP appointed to the position 521

Re: The basic Asterisk VoIP

For the sip.conf file, here is what I configured:

Code:

; Configuration file sip phones , [general] context = from-sip, defined for signaling context = fxo_ks bindport = 5060, UDP Port to bind to (SIP standard port IS 5060) BindAddr = 0.0.0.0, IP address to bind to (0.0.0.0 binds to all) srvlookup = yes; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; flexible loan to place SIP calls based on domain , names to Some Other SIP users on the Internet language = en; Default language setting for all users / peers ; This May Also Be Set for individual "users / peers localnet = 192.168.1.0/255.255.255.0; All RFC 1918 addresses are local networks nat = No, Global NAT settings (Affects all peers and users) ; yes = Always ignore info and assume NAT ; no = Use NAT mode only LUKAS to RFC3581 ; never = Never Attempt NAT mode or RFC3581 support ; route = Assume NAT, do not Send RPORT ; (work around more UNIDEN bugs) hidecallerid = no; allows the display of the Appalachian no ; ************************ ******************************************** ; ***** ************************************************** ************* [520] type = friend; right to emit and receive context = from-sip; context name secret = test; password username = 520; user asterisk callerid = "hardphone <520>; ID phone host = dynamic; type of award IP canreinvite = yes; to precise ... dtmfmode = RFC2833; to precise ... restrictid = yes; to precise ... incomminglimit = 5; Max received 5 calls simultaneously disallow = all; to precise ... allow = alaw; protocol selected = alaw ; *************************** ****************************************** ; ******* ************************************************** ************ [521] type = friend; right to emit and receive context = from-sip; context name secret = test; password username = 521; user asterisk callerid = "softphone" <SOFTPHONE> ; ID phone host = dynamic; type of allocation of IP canreinvite = yes; to precise ... dtmfmode = RFC2833; to precise ... restrictid = yes; to precise ... incomminglimit = 5; received 5 calls simultaneously Max disallow = all; to precise ... allow = alaw; protocol selected = alaw

With this configuration, the communications function and sound quality is correct. You still have all the same with phone setup to do, but this is by far the least difficult. in this case, your asterisk server is used to both "register "to" SIP Proxy 'and' outbound server. Hope this will help you.

Re: The basic Asterisk VoIP

C# open source VoIP softphone

This solution is also capable to connect to Asterisk. It has an SDK and with
it C# softphone, or VB.net softphones can be created.
I can strongly recommend it