<asteriskathome_handbook.pdf>


Version: 1.5
Date: 05/31/05

1 Introduction <#Section_1>
2 Installation <#Section_2>

    2.1 Hardware Requirements <#Section_2.1>
    2.2 Install from ISO <#Section_2.2>
    2.3 Install to an existing CentOS 3.4 system <#Section_2.3>

3 Configuration <#Section_3>

3.1 Quick start <#Section_3.1>

    3.1.1 Set an IP address for your Asterisk@Home box <#Section_3.1.1>
    3.1.2 Configure an extension <#Section_3.1.2>
    3.1.3 Configure a trunk for outbound and inbound calls <#Section_3.1.3>
    3.1.4 Configure Outbound Routing <#Section_3.1.4>
    3.1.5 Configuring Incoming Calls <#Section_3.1.5>

3.2 PSTN interface cards <#Section_3.2>

3.2.1 FXO Cards <#Section_3.2.1>

    3.2.1.1 Digium Wildcard X100P OEM FXO PCI Card <#Section_3.2.1.1>

3.2.2 FXS Cards <#Section_3.2.2>

    3.2.2.1 Digium TDM400P FXO/FXS Card <#Section_3.2.2.1>

3.2.3 T1/PRI Cards <#Section_3.2.3>

3.3 VOIP Service Providers <#Section_3.3>

    3.3.1 Free World Dialup (FWD) <#Section_3.3.1>
    3.3.2 VoicePulse <#Section_3.3.2>

3.4 Additional feature not installed by default <#Section_3.4>

    3.4.1 FAX to E-mail <#Section_3.4.1>

4 Phones <#Section_4>

4.1 Soft phones <#Section_4.1>

    4.1.1 X-Ten <#Section_4.1.1>
    4.1.2 sipXphone <#Section_4.1.2>

4.2 Hard phones <#Section_4.2>

4.2.1 Cisco 7960/7940 <#Section_4.2.1>

    4.2.1.1 Setting up a Cisco phone step by step <#Section_4.2.1.1>

4.2.2 Polycom <#Section_4.2.2>

5 Configuring and using Asterisk@Home features <#Section_5>

5.1 Conferencing <#Section_5.1>

    5.1.1 External access to conferences <#Section_5.1.1>

5.2 AMP (Asterisk Management Portal) <#Section_5.2>
5.3 Flash Operator Panel <#Section_5.3>
5.4 FTP server (vsftpd) <#Section_5.4>
5.5 Cisco XML Services <#Section_5.5>
5.6 Music On Hold (mpg123) <#Section_5.6>
5.7 Fax support (SpanDSP) <#Section_5.7>
5.8 xPL <#Section_5.8>
5.9 Sugar CRM <#Section_5.9>

6 Asterisk@Home add-ons <#Section_6>

    6.1 Click-to-Dial using Microsoft Outlook and AstTapi <#Section_6.1>
    6.2 H.323 add-on <#Section_6.2>
    6.3 Webmin - web based Linux management <#Section_6.3>

7 Useful Info for Asterisk@Home <#Section_7>

7.1 Feature Codes - List <#Section_7.1>


  1 Introduction

Asterisk@Home was created to make installing Asterisk easy.
Experimenting with Asterisk should be fun and not take hours or days to
set up.


  2 Installation


    2.1 Hardware Requirements

the faster the system you use to run Asterisk the more simultaneous
calls it will be able to take a 300MHz PII with 128 megs of RAM should
be able to take a few calls for home use.


    2.2 Install from ISO

Download the latest ISO from http://asteriskathome.sourceforge.net and
burn <http://members.home.nl/lsnoek/iso.htm> it to a CD
Book your Asterisk PC with the CD and press enter

[Warning] Warning

This will erase all data on the hard drive of the PC!!!

After the Linux is loaded the CD will eject. Take out the CD and wait
for the system to reboot
Booting the system might take a while it is building Asterisk from source.
Log in to your new Asterisk@Home box (*user: root, password: password*)

[Warning] Caution

Change your root password immediately by typing *passwd*
Asterisk@Home boxes with default passwords have been hacked!!


    2.3 Install to an existing CentOS 3.4 system

If you have problem with the ISO you can install onto an existing CentOS
server.

Download  CentOS 3.4 ISOs from the CentOS Mirrors
<http://www.centos.org/modules/tinycontent/index.php?id=13>

Make a directory to install from, put the install file there, and install.

mkdir /var/aah_load
cp asteriskathome-0.7.tar.gz /var/aah_load
cd /var/aah_load
tar xvfz asteriskathome-0.7.tar.gz
./install.sh


  3 Configuration

Once you have installed your Asterisk@Home system you can start to
configure it. First you should register a phone to an extension. Next
you can configure a trunk to make outbound calls and accept incoming calls.


    3.1 Quick start


      3.1.1 *Set an IP address for your Asterisk@Home box*

To configure Asterisk@Home you will need access to the Web GUI if you
log into the console you will get a message telling you the IP address
of your Asterisk@Home box. If this field is blank you have a problem
with your network card. Type *netconfig* from the command line. This
will allow you to set a static IP address for your Asterisk@Home box.
After you have set up your network card type *reboot*. Your system will
restart and hopefully you can connect to your system from a browser.


      3.1.2 *Configure an extension*

Go to a pc on your network with a web browser and connect to your
Asterisk@Home box. Click on *Asterisk Management Portal *(*user: maint,
pass: password*) and then *setup*. Click *Extensions* then *Add
Extension*. Use the default extension *200* and type in a password for
registration like "abc123". then enter the name of the person using this
extension.

Go down to the voicemail section and enter a voicemail password. Use
something you can type on a phone keypad like '1234'. Enter an e-mail
address where you would like your voice messages sent and *click add
extension*. Then click on the *red apply bar* at the top of the screen.

*Set up a phone for this extension.* Get a SIP phone an *X-Ten soft
phone* is good for testing. See the section on the X-Ten phone <#X-Ten>.
remember to use your * extension* and *extension password*.

*Make a call from your phone*. (try *45 this is a local echo test)


      3.1.3 Configure a trunk for outbound and inbound calls

Using *AMP* (*user: admin, pass: password*) select *setup* then
*trunks*. Click on the type of trunk you want to create. See the VOIP
Service Providers section for how to configure a trunk for your provider.


      3.1.4 Configure Outbound Routing

Next you need a route to allow calls from your phones to go out on a
trunk. If you have more than one trunk you can set up rules to determine
how a trunk is chosen for each call. Here we will set all calls to go
out one trunk.

Using *AMP* (*user: admin, pass: password*) select *setup* then
*Outbound Routing*. Type in a *name* for your route. Then enter the
following in the *dial pattern box*.

1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX

This will set all calls to use this route.

Next go to the *Trunk Sequence* section. Drop down the box and *select
the trunk you configured earlier*. Click *add*.

That's it. Click *Submit Changes* and then click on the *red apply bar*
at the top of the screen.

Try dialing a number on your phone like 19197543700 (RedHat customer
support) and you should hear "Welcome to RedHat"


      3.1.5 Configuring Incoming Calls

Next you need a route to allow calls from your provider to go someplace.
Using *AMP* (*user: maint, pass: password*) select *setup* then
*Incoming Calls*. Under the *Send Incoming Calls from the PSTN to:
section* drop down the box next to *Extension:* and select the
extension* 200* you created earlier.

Click *Submit Changes* and then click on the *red apply bar* at the top
of the screen.

Call the phone number from your provider with a cell phone or other
non-IP phone. Your SIP phone at extension 200 should ring.


    3.2 PSTN interface cards

There are may type's of PSTN interface cards. These can include a single
FXO card the allow 1 home telephone line to be connected to asterisk, or
a T1 card that allows a digital trunk to be connected to Asterisk. All
of these cards allow you to make calls directlly on the Public Switch
Telephone Network without have to use a VOIP phone service provider.

PSTN cards and also be used to connect PSTN (Non VOIP) phone to
Asterisk.  These can include a single FXS card the allow 1 home
telephone to be connected to asterisk, or a T1 card that allows a
Channel bank with 24 phones or a tie line to a PBX to be connected to
Asterisk.


      3.2.1 FXO Cards

These cards allow you to connect a POTS (plain Old Telephone System)
line to your Asterisk@Home box.


        3.2.1.1 Digium Wildcard X100P OEM FXO PCI Card

These voice modems are available on e-bay for about $7.00 They are not
made by Digium and are of low quality. Having said this they can work
very well in many situations. If your phone line is well balanced there
will work well. If it’s not you can get some bad echos. We recommend
them only for testing not for a production system.

To configure these cards for use with Asterisk@Home first use the zaptel
card auto-config utility to set up the zaptel driver.

Type *genzaptelconf -s -d* from the command line

Next go into the AMP web interface and create a trunk.

There is already a trunk called ZAP/g0 edit this

Enter the *phone number* for you pots line in the *Caller ID field*
Enter *1* for *Maximum channels*
Set a dial rule if you want for this trunk
Select an outbound dial prefix to select this trunk when dialing
Set the *Zap Identifier* to *1* (the default is g0)

This trunk is now configured

*you must add a route for incoming calls or asterisk will not answer
this line*

click on *incoming calls* in amp and set up an incoming route.

to make outbound calls you will need an outbound route. set one up in AMP


      3.2.2 FXS Cards

These cards allow you to connect an Analog phone to your Asterisk@Home box.


        3.2.2.1 Digium TDM400P FXO/FXS Card

This card has 4 module ports that can be loaded with FXS or FXO mudules.
You can auto config this card just like the X100P. *Channel 1* is the
*top RJ-45* on the back of the TDM400P
<http://www.digium.com/index.php?menu=wildcard_tdm400p2> card.

To configure these cards for use with Asterisk@Home first use the zaptel
card auto-config utility to set up the zaptel driver.

Type *genzaptelconf -s -d* from the command line

Next, using config edit, look in the *zapata-auto.conf* file you will
see a list of all your channels. Set up the trunks as trunks and the
extensions as extensions in AMP.

For example if your zapata-auto.conf file looks like this

; Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"
signalling=fxo_ks
; Note: this is an extension. Create a ZAP extension in AMP for Channel 1
channel => 1

signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 2
context=from-pstn
channel => 2

then add a zap extension for channel 1 and a zap trunk for channel 2

you may have to reboot your system to get everything going.

*you must add a route for incoming calls or asterisk will not answer
your trunk*

click on *incoming calls* in amp and set up an incoming route.

to make outbound calls you will need an outbound route. Set one up in AMP.


      3.2.3 T1/PRI Cards

*Setup for Digium T100P
<http://www.digium.com/index.php?menu=wildcard_t100p> and a PRI*

login to the CLI as root

[root@asterisk1 /]# cd /etc
[root@asterisk1 etc]# nano -w zaptel.conf

*Add these lines to /etc/zaptel.conf *

span=1,1,0,esf,b8zs
bchan=1-23 # set this to 1-15,17-31 for E1
dchan=24 # set this to 16 for E1

Add # to front of fxsks=1 so it looks like this:

span=1,1,0,esf,b8zs
bchan=1-23 # set this to 1-15,17-31 for E1
dchan=24 # set this to 16 for E1

#fxsks=1
loadzone = us
defaultzone=us

*Control-X then Y to save zaptel.conf *

Log into Asterisk@home using a browser http://-asterisk-ip-adress/admin
Then click on *Maintenance *
Then click on *Config Edit *
Then click on *zapata.conf *

Comment out with ; to signalling=fxs_ks:

*;signalling=fxs_ks *

Then copy just below it:

signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=national

Then add this callerid=asreceived under ;usedistinctiveringdetection=yes

Change echocancelwhenbridged=yes

Change echotraining=400 ; Asterisk trains to the beginning of the call,
number is in milliseconds

at the end of the file copy and past:

channel => 1-23 ; Set this to 1-15,17-31 for E1

Click Update and Click Re-Read Config

Go back to SSH root

[root@asterisk1 etc]# modprobe wct1xxp
[root@asterisk1 etc]# ztcfg -vv

Do a shutdown and restart the system

[root@asterisk1 etc]# shutdown -r now

You just need to login via web and make your setup.


    3.3 VOIP Service Providers

There are many service providers. Some provide proxy server that make it
possible to connect to other members of that provider. Other providers
offer both icomming and outgoing PSTN to VOIP termination. Here are a
few common providers and how to make the work with Asterisk@Home.

Most providers will give you phone number and a password for that
provider some will also give you a user name. If you get a real PSTN
number from the provider it will be a normal 10 digit number (US
providers). some providers give out shorter number that can only be used
by other members of that provider.


      3.3.1 Free World Dialup (FWD)

*Contact:* http://connect.voicepulse.com/
*Service:* proxy to other FWD users, Gateway to other providers
*Protocol:* SIP or IAX
*Cost:* free

You should have a phone number (123456) and a password (wibble). You
also need to have your FWD account setup for IAX. This is achieved by
visiting http://www.freeworlddialup.com
<http://www.freeworlddialup.com/>, logging in and turning on IAX. It
does take a little bit of time to be set up (10 mins or so), so do that
first. Once you've turned it on and clicked 'Submit' enough times (I
noticed I had to click Submit two or three times before it came up with
'Changes Successful', that may have just been a temporary glitch) you're
ready to proceed below.

Once again, you need to be in AMP, the Asterisk Management Portal. Click
on *Setup* up the top, but this time click on *Trunks* on the left.
Click on *Add IAX2 Trunk*

*Dial Prefix* is, usually, 393 — That's 'FWD' on your phones pad.
Leave *Default Trunk* switched off
*Outbound Caller ID* should (but doesn't have to be) set to your FWD
Number. This is what is displayed when you call someone else through
FWD. They'd normally just see your Extension (200).

*Outgoing Settings*

*Trunk Name:*  fwd  (This is just a descriptive name, and is what
appears on the left of the screen)
*PEER Details:*  (Change '123456' and 'wibble' to be your FWD Number and
Password)
 

host=iax2.fwdnet.net
type=peer
username=123456
secret=wibble


*Incoming Settings*
*USER Context:* iaxfwd  (Pay attention here. Don't change it. or it
won't work)
*USER Details*  (Nothing needs to be changed here, this can be pasted
straight in)
 

allow=ulaw
auth=rsa
context=from-pstn
disallow=all
inkeys=freeworlddialup
type=user


*Register String:* should be set to
yournumber:yourpassword@iax2.fwdnet.net, using our examples above, it
would be *123456:wibble@iax2.fwdnet.net*

Thats it. Click on *Submit Changes*, and then on the big red 'You have
made changes' bar and you're done.

Assuming you've got your username and password correct, you should now
be able to dial '393612', Which will read out the time to you. IF you're
feeling exceptionally brave, call '393613', which is a useful little
echo tester - it'll just bounce back to you everything you say to it.
Ycan then try '393514' which is FWD's 'Coffee Lounge' - I've never
actually sucessfully had a conversation with anyone there, however, or
'39355555', which calls a random volunteer, so you can actually speak to
a live person!


      3.3.2 VoicePulse

*Contact:* http://connect.voicepulse.com/
*Service:* PSTN termination
*Protocol:* IAX
*Cost:* pay

Once again, you need to be in AMP, the Asterisk Management Portal. Click
on *Setup* up the top, but this time click on *Trunks* on the left.
Click on *Add IAX2 Trunk*

*Dial Prefix* 9 if your not alredy
Leave *Default Trunk* switched off (or make this the default if you want
all your calls to use it)
*Outbound Caller ID* should (but doesn't have to be) set to your
VoicePulse Number.

*Outgoing Settings*

*Trunk Name:*  voicepulse-out-01  (This is just a descriptive name)
*PEER Details:*  (Change <your username> and <your password> to be your
VoicePulse Number and Password)
 

host=gwiaxt01.voicepulse.com
secret=<your password>
type=peer
username=<your username>


*Incoming Settings*

*USER Context:* voicepulse-in-01  (Pay attention here. Don't change it.
or it won't work)
*USER Details*  (Nothing needs to be changed here, this can be pasted
straight in)
 

auth=rsa
context=from-pstn
inkeys=voicepulse01
type=user


*Register String:* should be set to <your username>:<your
password>@gwiax-in-01.voicepulse.com example (bob:abc123
@gwiax-in-01.voicepulse.com)

Thats it. Click on *Submit Changes*, and then on the big red 'You have
made changes' bar and you're done.
For a test make a call (try 1-800-555-1212)


    3.4 Additional feature not installed by default

Some of the features of Asterisk@Home are not installed by default. Here
are the steps to install them.


      3.4.1 FAX to E-mail

Type *install-pdf* from the Linux command line. This will install
support for encoding incoming faxes as PDFs and e-mailing them.


  4 Phones

VOIP phones vary widely in price, features, and sound quality. Phones
support either SIP or IAX protocol. For simple testing a free soft phone
like X-ten is good. For day to day use a good hard phone is the best.
Grandstream makes a cheap easy to configure phone with good quality and
feature. Polycom and Cisco make the best phones with the highest sound
quality and feature. If you have and existing analog phone like a
cordless that you want to use and ATA will convert an analog phone to SIP.


    4.1 Soft phones

Soft Phones run on a host computer and use the computer speaker and mic
or a headset for voice. Soft phones are available for most operating system.


      4.1.1 X-Ten

One of the easiest to setup is X-Lite. X-Lite is available for free from
http://www.xten.com. <images/sip_settings.jpg>

If X-Lite cannot connect, the setup screen should open, if not, click on
the "drop down" icon just to the left of the green Off-Hook icon.

Under *System Settings*, select the *SIP Proxy settings*, then
double-click on the first entry. You will see the SIP Proxy settings as
shown here. The settings I changed are as follows:

Username: 200 (my extension)
Authentication User: 200 (my extension again)
Password: 1111 (my extension password)
DomainRelm: 192.168.5.50 (PBX IP address)
SIP Proxy: 192.168.5.50 (PBX IP address)


      4.1.2 sipXphone

The sipXphone project <http://www.sipfoundry.org/sipXphone/index.html>,
formerly known as Pingtel <http://www.pingtel.com>'s instant xpressa
soft phone, is a fully functional SIP soft phone that runs on Microsoft
Windows and Linux. The Pingtel's xpressa phone (no longer a Pingtel
product) is based on this same source base. Windows Download from
http://www.sipfoundry.org/pub/sipXphone/win32/


    4.2 Hard phones


      4.2.1 Cisco 7960/7940

The Cisco SIP IP Phone software allows businesses and service providers
to use the Cisco 7940 and 7960 IP Phone platforms in any standard SIP
network. The SIP software provides for both on-board traditional desktop
services such as Caller-ID, Call Hold, Call Transfer, 3-Way Calling, and
Call Waiting as well as an XML interface to allow for enhanced web based
services. The XML interface allows the phone to transcend the
traditional phone paradigm and become a true Internet appliance. By
supporting web browsing type functionality as well as allowing for
application developers to directly control the user interaction on the
phone and integrate tightly with the Cisco SIP phone, the Cisco SIP
phone is a key enabler of enhanced and rapid application deployment in
any SIP customer's network.

Type *setup-cisco *from the command line to set up a basic
SIPDefault.cnf in the /tftpboot directory.

Copy your Cisco 7.4 SIP firmware into the /tftpboot directory using
WinSCP <http://winscp.net>.

using the *Cisco config* tool in the maint menu to set up your phone.


        4.2.1.1 Setting up a Cisco phone step by step

Connecting a Cisco 7960/7940 to your Asterisk@Home. The easiest way to
use these phones with Asterisk@Home is with the SIP firmware.

Power on your phone and connect it to your network. You will need to
change the IP address of your phone to a free one on your network or
enable DHCP is you have a DHCP server.

*Unlock your phone* by pressing ***#* (for old firmware) or selecting
*unlock config* from the *config menu* and type in the password (default
is '*cisco*')

Change the IP address or enable DHCP. Enter the IP address of your
Asterisk@Home box in the TFTP server field. If DHCP is enabled you will
have to select* alternate TFTP server = yes* so you can edit the TFTP field.

Get the latest firmware for the Cisco phone (7.4) and copy the files
firmware files to your Asterisk@Home box. Put them in the */tftpboot*
directory.

Create a Default.cnf file by using the setup-cisco command and change
the file permissions on the firmware files that you uploaded in the last
step. Type the following.

setup-cisco
chmod 777 /tftpboot/*

Add a Cisco config file for your phone. Use the wed GUI and click
*maintenance -> Cisco Config -> Create *a new phone configuration file.

Type in the *mac address* of your phone. It’s located on the *bottom of
your phone. (starts with 00)* Enter a name for your phone in the Phone
Label box. Enter *200* in the *Line Name* and the *Display Name* fields
for *Line 1*. Click *add*.

Click on *Edit phone configuration files* and click *edit* next to your
phone. Go to* Line 1* and type 200 in the *Auth Name* box. Then enter a
password like abc123 in the *password box*. Click *change*.

Create extension 200. *setup -> extensions* use *abc123* for the
*extension password*.

Reboot your phone by disconnecting the power or pressing *-6-settings.

Your phone should upload the new firmware and extension. Make a test
call like *65.

_*Trouble shooting*_: The Cisco phone can be difficult to upgrade. If
your phone does not seem to upload the firmware try this. From command line.

cp /tftpboot/cisco_util/* /tftpboot 

Then reboot the phone. After the phone has upgraded the firmware. Type
the following so the phone will *boot normally.*

rm /tftpboot/xmlDefault.CNF.XML
rm /tftpboot/XMLDefault.cnf.xml


      4.2.2 Polycom

http://www.voip-info.org/wiki-Polycom+Phones


  5 Configuring and using Asterisk@Home features


    5.1 Conferencing

Each extension you create will have its own conference. For example
extension 200 has conference 8200. Just dial 8200 to get into your
conference. If you dial from 8200 you are the moderator.

You need to set a password for the meetme user. Type passwd-meetme from
the Linux command line.

Once you have started a conference you can manage it from the WebMeetme
page. Just enter the extension for the conference.

External access for conferences is disabled for security reasons. It is
not always desirable for people outside the system to have access to
conferences


      5.1.1 External access to conferences

If you want dial in access to conferences you need to be add an option
on one of the auto attendants for this.

First go into the *Maintenance* page and select *Edit configs*. The file
you want to edit is *extensions_custom.conf *add these lines to the
bottom of the file. *Update* the file and *reload* configs.

[custom-meetme]
include => ext-meetme 
exten => s,1,BackGround(enter-conf-call-number)
exten => h,1,Hangup()

Next go into *setup* and *Create a Digital receptionist* if you have not
already done so. Record a main message for your receptionist. Something
like "press # for a company directory or press 1 to enter a conference"
click continue and Select 1 for number of menu options.

Next you will have to select an *Action* for this menu item. Click *
Custom App *and type in* custom-meetme,s,1 *then click continue.

That should be it when you dial in you select 1 from the main menu and
you will be prompted for your conference number. You can dial *7777
*from and extension to simulate an incoming call.


    5.2 AMP (Asterisk Management Portal)

AMP is a web GUI that allows the configuration of Asterisk without the
need for editing config files. If you use AMP be careful not to make
manual changes to the config files that will affect AMP’s operation. You
can learn more about AMP from their web site.
http://amp.coalescentsystems.ca/


    5.3 Flash Operator Panel

Flash Operator Panel is a real-time web interface to Asterisk. You can
see what all of your extensions, trunks, and conferences are doing. We
have extension 200 set up for testing. This is a live extension and will
work as soon as a phone is registered to ext 200. The rest of the
buttons are dummy objects for examples.

The op_buttons.conf file has all of the configs for each button in Flash
Operator Panel you can edit this using the Config Edit button on the
maint menu. You must reboot the Asterisk or restart the panel service
after you make changes. The normal config files for Flash Operator Panel
end in .cfg we changed them to .conf so they would match the other
Asterisk files.

See the Flash Operator Panel we site for more info on Flash Operator
Panel.  http://www.asternic.org/


    5.4 FTP server (vsftpd)

Asterisk@Home has an FTP server that is used primarily for configuring
phones that use FTP such as Polycom. For security reasons only one user
is configured by default to user ftp. This user is ftpuser (password:
asteriskftp). It can access file in the /var/ftp directory. If your
would like other users to have ftp access add them to the
/etc/vsftpd.user_list file.


    5.5 Cisco XML Services

Asterisk@Home has a Cisco XML Services that can be used to look up
contacts in a database. Click on CRM from the Asterisk@Home main menu.
(user: admin password: password) and enter all of your contacts and
their phone numbers.

Use the *setup-cisco* command to set up SIPDefault.cnf for your Cisco
phones. The setup-cisco script will set your Services URL to
http://ipaddressofyourasteriskbox/cisco/directory/services.php you can
then use the Services button of your Cisco phone to call your contacts.


    5.6 Music On Hold (mpg123)

Asterisk@Home has now uses mpg123 for music on hold. Put a call on hold
and they hear music not dead air!  See voip-info.org
<http://voip-info.org> for more info.

www.voip-info.org music on hold info
<http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf>


    5.7 Fax support (SpanDSP)

SpanDSP allows Asterisk to receive faxes. This service must be
installed. Type *install-pdf* from the command line. You can set this up
in AMP.
 

    5.8 xPL

xPL is a very simple and powerful home automation protocol. It can be
used it interface Asterisk into an existing home automation system. For
example if you get a call your TiVo
<http://www.xplproject.org.uk/modules.php?op=modload&name=Sections&file=index&req=viewarticle&artid=30&page=1>
can display the caller ID info on your TV screen and turn down the stereo.

If you have a Microsoft based home automation system see the xPL project
page (http://www.xplproject.org.uk/).

If you have a fully Linux based system MisterHouse
(http://www.misterhouse.net) has a good xPL interface.

One of the best features of xPL is the ability to mix and match
applications. A few Windows apps, MisterHouse on Linux, Asterisk@Home on
CentOS, and a TiVo can all be on the same network and share xPL information.

xPL is not running be default. To enable it type cd /usr/src/xplast then
type ./install.sh

Once xPL is installed and running it will send out xPL heartbeat
messages. xPL will also send out notification of new voicemail messages.
xPL can also send out callerid info for any incoming calls. you must add
a line to extensions.conf to make it do this. Add this line to
extension.conf at a location where you want the callerid sent out.

exten => s,1,AGI,xplring.agi

For example a good place to add this line is to the
[from-pstn-timecheck] section. Be sure to renumber the following lines.

 
[from-pstn-timecheck]
exten => .,1,Goto(s,1)
exten => s,1,AGI,xplring.agi
exten => s,2,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:)
exten => s,3,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:)
exten => s,4,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten => s,5,Goto(from-pstn-afthours,s,1)


    5.9 Sugar CRM

SugarCRM <http://www.sugarcrm.com/home/> is a complete CRM (Customer
Relation Management) system. Although most of the feature of this
software package are beyond the needs of home users the contact
management module in this package is the best we have ever seen in a web
application. SugarCRM also has sophisticated user management.

Asterisk@Home uses SugarCRM mainly to manage contacts. SugarCRM is
integrated into the Cisco XML phone directory making it possible to look
up CRM contacts from a Cisco phone.

We have also added a *click-to-dial* function to CRM so that you can
dial contacts directly from the web interface.

The default account for CRM is *admin* with a password of *password*.
Once you have logged in you can create CRM users. You should make a CRM
user for each extension. Enter the extension of a phone when you create
a user so Asterisk will know where to send click-to-dial calls for that
user. *Use standard Asterisk notation for extensions*. For example for
sip extension 200 enter *SIP/200* in the extension field.


  6 Asterisk@Home add-ons

These add-on packages are not included with the Asterisk@Home iso. They
can be installed to add functionality to your Asterisk@Home system.


    6.1 Click-to-Dial using Microsoft Outlook and AstTapi

AstTapi <http://sourceforge.net/projects/asttapi/> is a Microsoft TAPI
<http://msdn.microsoft.com/library/default.asp?url=/library/en-us/tapi/tapi3/telephony_application_programming_interfaces.asp>
to Asterisk bridge that makes it possible to do click-to-dial from
Microsoft Outlook and other TAPI compliant applications.

Download it from http://sourceforge.net/projects/asttapi/
<http://sourceforge.net/projects/asttapi/>

First make a login for AstTapi. Use the *config editor* to edit *
manager_custom.conf*
There is a default AstTapi account you can use. Just remove the *#* from
the *permit* line and change the *192.168.1.0* to the * network address*
of your network. Then reload Asterisk.

*Install AstTapi*

In outlook click on a contact and click on the phone icon. Select a
phone number.
In the *"new call"* dialog box select dialing options.
In the "*connect using line*" box drop down and select Asterisk.
Click on line properties.

*Host:* <ip of Asterisk server>
*Port:* 5038

*User:* AstTapi
*Password:* AstTapi
*User Channel:* sip/200 (your extension)

Select *dial by context*

Enter *outbound-allroutes* in the context box

Click Apply

You can now make calls from Outlook!


    6.2 H.323 add-on

This package adds H.323 support to Asterisk it also install the GnuGK
H.323 gatekeeper.

*Installation*

Copy the asteriskathome-h323.zip
<http://sourceforge.net/projects/asteriskathome/> file to you
Asterisk@Home server using WinSCP <http://winscp.net>. Unzip the file by
typing

*unzip asteriskathome_h323.zip*

from the command line. Next type

*./install.sh*

When the install is done reboot your Asterisk@Home system.

*Testing*

register a SIP phone with Asterisk@Home

open Microsoft Netmeeting click on tools -> options -> Advanced calling
Under Gateway Settings check off Use a gateway and enter the ipaddress
of your Asterisk@Home system.
Click OK

The calls from the gatekeeper will be put in the *from-pstn context*.
Make sure you have an incoming call route set up so the calls have some
place to go.

go back to the main NetMeeting screen and type in the extension of your
sip phone. You should be connect to your incoming call route.

*For support try*

The asterisk-oh323 project home page
http://www.inaccessnetworks.com/projects/asterisk-oh323

GnuGK home page
http://www.gnugk.org/


    6.3 Webmin - web based Linux management

Webmin in a great package for managing a Linux box from the web. Webmin
make it easy to configure application like SMTP mail. To install Webmin
download the latest RPM from their web site and install it.

http://www.webmin.com/

Example

wget http://unc.dl.sourceforge.net/sourceforge/webadmin/webmin-1.200-1.noarch.rpm
rpm -Uvh webmin-1.200-1.noarch.rpm


  7 Useful Info for Asterisk@Home


    7.1 Feature Codes - List

*411 Directory
*43 Echo Test
*60 Time
*61 Weather
*62 Schedule wakeup call
*65 festival test (your extension is XXX)
*70 Activate Call Waiting (deactivated by default)
*71 Deactivate Call Waiting
*72 Call Forwarding System
*73 Disable Call Forwarding
*77 IVR Recording
*78 Enable Do-Not-Disturb
*79 Disable Do-Not-Disturb
*90 Call Forward on Busy
*91 Disable Call Forward on Busy
*97 Message Center (does no ask for extension)
*98 Enter Message Center
*99 Playback IVR Recording
666 Test Fax
7777 Simulate incoming call

------------------------------------------------------------------------

If you found this documentation helpful please consider making a
donation to the Asterisk@Home project:

http://sourceforge.net/donate/index.php?group_id=123387

<http://asteriskathome.sourceforge.net/handbook>