asterisk users - Jan 2005 - Asterisk @ Home 0.4 w/ Broadvoice + 5 SIP Phones How To

Okay, I just spent some time getting this done for a
project.  I documented the steps performed.  Thanks to
all of those who provided input.  

As time goes on, and people always have different
settings, this may or may NOT work for you.  

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

Asterisk @ Home 0.4 HowTo
William Pool (Puddle)
01-28-05
Description: Getting Asterisk @ Home to work with a
Broad Voice SIP provider
account and Five SIP Software XTen 

Install Asterisk @ Home OR CentOS 3.3+ with the
asteris@home.tar ball and 
install from that.  (Since, Asterisk@Home doesn't let
you partition disks)

Once You're up and loaded and able to get to
http://xxx.yyy.zzz.xyz/maint

Setup AMP the following way:

Trunks 

1.) Click on "Trunk ZAP/g0" change the dialout prefix
to something else.

NOTE: Unless you have a digum card and have a normal
POTS analouge line
you should NOT want this. You should leave it just
incase it ever happens(or just delete it who cares). 
However, change the dial-out # to something odd. 
Chances are it'll never be used.

2.) Add a SIP Trunk:

Trunk Name: BV

canreinvite=no
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=818xxxyyyy
host=147.135.0.129
insecure=very
nat=yes
secret=password
type=peer
username=818xxxyyyy

User Context: sip.broadvoice.com

context=from-pstn
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=password
type=user
user=818xxxyyyy
username=818xxxyyyy

Register: 

818xxxyyyy:yourpassword@sip.broadvoice.com

Asterisk Specific Issues

1.) SIP Issues

The /etc/asterisk/sip.conf doesn't have some basic
behind nat server options.
Add the following after the [general] or in that
section somewhere.

externip=69.yzx.xxx.zzz
localnet=192.168.0.0
localmask=255.255.255.0
nat=yes

2.) Dial plan issues.  Calls from the sip provider
into the Asterisk Server
are hangup/congested and pushed away.  

Inside your [from-sip-external] add the line

include => from-pstn

AMP Specific Issues:

1.) Unable to upload file sizes larger then 500K

Edit /etc/php.ini Increase the varable
"upload_max_filesize" to the max size
you'll "ever" expect to play.  Increase this for
'whole' mp3 albums to "if"
you do that...  Also, edit /etc/httpd/conf.d/php.conf
and change the variable "LimitRequestBody" to
something more meaningful.  I use "12000000"

Security:

Created asterisk user:

adduser -s /bin/false -d /bin/false -c "Asterisk
Daemon User" asterisk

added asterisk user to asterisk group in /etc/group

chown -R asterisk:asterisk /var/log/asterisk
chown -R asterisk:asterisk /var/lib/asterisk

Edit /etc/init.d/asterisk to include the two 

AST_USER="asterisk"
AST_GROUP="asterisk"

MySQL, it's default passwd for Asterisk @ Home has a
LOT to be desired for.  Change this.  The password via
the MySQL DB itself and the following files:

/etc/asterisk/cdr_mysql.conf
/var/www/html/maint/phpMyAdmin/config.inc.php

Update Repository Catalogue

yum check-update <enter>

Update Repository 

yum update

Install mod_ssl since Apache doesn't have it included.

yum install mod_ssl.i386

Setup Apache 2.x ssl configs

mkdir /var/swww

Move the Asterisk@Home to the SSL folder 

mv /var/www/html /var/swww

Create blank HTML Page for people

mkdir /var/www/html
touch /var/www/html/index.html
echo "<TITLE>pbx.domain.com</TITE> >>
/var/www/html/index.html
echo "NOT Authorized maybe you mean <A
HREF=https://pbx.domain.com>https://pbx.domain.com</A>"
>> /var/www/html/index.html
/etc/httpd/conf/httpd.conf

ServerName leave commented out, had issues with
setting it.
Change at your own risk blah blah

/etc/httpd/conf.d/ssl.conf

After <VirtualHost _default_:443>

Uncomment:

 DocumentRoot "/var/swww/html"
 ServerAdmin admin@domain.com

# ServerName pbx.domain.com:443
# NOTE: Look above

Find the 'cgi-bin' directory in the ssl.conf file. 
Edit the path to reflect:

/var/swww/cgi-bin

Add (After the CGI-BIN Directory works)

# Use htpasswd -c /etc/httpd/conf/passwd/plainpasswd
user1
# this will create the user and passwd file
# after that use htpasswd
/etc/httpd/conf/passwd/plainpasswd user1
# I Know digest is better, but it doesn't always
# work

<Directory "/var/swww/html/maint">
SSLOptions +StdEnvVars
AllowOverride All
AuthType Basic
AuthName "Receptionist / Admin access Only!"
AuthUserFile /etc/httpd/conf/passwd/plainpasswd
Require user user1 user2 user3
</Directory>

<Directory "/var/swww/html/meetme">
SSLOptions +StdEnvVars
AllowOverride All
AuthType Basic
AuthName "Meetme access Only!"
AuthUserFile /etc/httpd/conf/passwd/plainpasswd
Require user user1 user2 user3
</Directory>

<Directory "/var/swww/html/admin">
SSLOptions +StdEnvVars
AllowOverride All
AuthType Basic
AuthName "Admin access Only!"
AuthUserFile /etc/httpd/conf/passwd/plainpasswd
Require user user1 user2 user3
</Directory>

Make sure firewall is setup somewhat:

Install redhat-config-securitylevel

yum install redhat-config-securitylevel

This will add the following dependences

[deps: gtk2 2.2.4-8.1.i386]
[deps: pango 1.2.5-2.0.i386]
[deps: atk 1.2.4-3.0.i386]
[deps: pygtk2 1.99.16-8.i386]

Add the following ports:

SSH,WWW,443,5060

443 https
5060 SIP

Edit /etc/sysconfig/iptables

Where you have your SIP (5060) line delete the '-m
tcp' Change '-p tcp' to '-p udp'

Setup NTP correctly

/etc/ntp.conf

Install redhat-config-date to make time setting easier

yum install redhat-config-date

***PHONES***

I've used XTen Lite (Soon to try Pro)

The Call Groups / Extensions are self explainitory so:

XTenLite 
NOTES: The context/username/mailbox MUST be a #.  I
couldn't 
get anything to work with it having any letters.

[200]
username=200
type=friend
secret=1234
qualify=1000
port=5060
nat=yes
mailbox=200
host=dynamic
dtmfmode=inband
context=from-internal
canreinvite=no
callerid="First Name" <200>



		
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Hi

I know that attended transfers are only available in the CVS Head.

I downloaded the asterisk-update.sh script from voip-info.com and ran it
with these parameters

./asterisk-update.sh update dev

It looked as tho CVS HEAD was downloading and compiling, although it
couldn't download the addons.

However, now it's up and running, only blind transfers work with
"#", and I
cannot change the blind transfer key to "##", it only takes the first
character. And Attended transfers still isn't running.

Is there something I've missed.

The version info reports:
Asterisk CVS-v1-0-02/03/05-10:24:22

Any help would be great.

Thanks

Mike 

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