Asterisk@Home Handbook Wiki Chapter 9

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Chapter 9 Software that is not installed with Asterisk@Home

The following software is not installed with Astersik@Home but you definitely may want to consider using it.


9.1 Click-to-Dial using Microsoft Outlook and AstTapi

AstTapi is a Microsoft TAPI to Asterisk bridge that makes it possible to do click-to-dial from Microsoft Outlook and other TAPI compliant applications.

9.1.1 Download AstTapi and install it

Download this software from Sourceforge at http://sourceforge.net/projects/asttapi/.
Be sure that outlook is turned OFF before installing it. When finished, reboot the PC as requested.

9.1.2 Modifying the "Manager_Custom.conf" file in A@H (don't panic! this is easy!)

We need to make a quick edit to a A@H text file. You can do this from inside AMP or you can do this from the CentOS command line.

First we have to make a login for AstTapi. Click Maintenance under "AMP" and then click "Config Edit" and then click manager_custom.conf. In this file there already 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 your A@H server is on. This is NOT an IP address! It is the NETWORK address.

Now there is a chance that your phone may not even BE on the same network as the asterisk server. If this is true, you'll have to use 0.0.0.0/0.0.0.0 (which means any IP address from any subnet that has access to port 5038 can login). If this is true, you better use a better password then what is already there. Remember this login and password. Save the file. Then reload Asterisk.

9.1.3 Configuring AstTapi in outlook

Now we need to configure AstTapi inside of outlook.

Start outlook and select and click on a contact. There is a phone icon on the bar above the contact. Click that phone icon. A small window will appear with the following (use your imagination):

Number to dial
Contact: (Contact Name Field) (Open Contact Button)
Number: (Phone Number Field) (Dialing Properties Button)
(CHECK BOX Create new Journal Entry when starting new call)

(Start Call Button) (End Call Button) (Dialing Options Button) (Close Button)

Now that you can see the name and number you want to dial, click the "Dialing Options Button". In the "Connect Using Line" field, arrow down to "Asterisk" and click the "Line Properties" button right next to it.

Use the following entries in the next window:

Asterisk Server:
Host: ip of Asterisk server (you can use an IP address here or even a DNS address)
Port: 5038

User Information:

User: AstTapi (this might be case sensitive)

Password: AstTapi (or the password which you have chosen in the "manager_custom.conf" file)

User Channel: sip/200 (This is your extension. This is the number that will ring, requiring you to pick it up and get connected to the contact's number)

Context:

Select "Dial by 'Context'"

Context: outbound-allroutes (note, this is in every guide I can find, but didn't work for me on a default install. I had to use "Caller ID" and use my extension's CID and also check off "Attempt to set outgoing ID") (your milage may vary). (If neither work, try empty "Contect - Dial by Contect" or "Dial - Dial out by using the Dial application" fields)

Click Apply and OK and OK again

Now Click "Start Call" in the "New Call" box to begin the call.

Your extension will ring and once you pick it up, asterisk will connect you to the number of the contact you've chosen.


9.2 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 file to you Asterisk@Home server using WinSCP. 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/


9.3 Webmin - Web Based Linux Management

Webmin in a great package for managing a Linux box from the web. Webmin make it easy to manage all types of different service in a linux box (file manager, change passwords, etc). To install Webmin download the latest RPM from their web site and install it.

http://www.webmin.com/ - Webmin Homepage
http://sourceforge.net/projects/webadmin - Sourceforge Page

From the CentOS command line type the following:
(please note, check to see what is the LATEST version and file name of the webmin "noarch.rpm" file. If it is different from the example, be sure to use the latest file name. At the time of this edit, the file name was webmin-1.260-1.noarch.rpm)

wget http://heanet.dl.sourceforge.net/sourceforge/webadmin/webmin-1.270-1.noarch.rpm

rpm -Uvh webmin-1.270-1.noarch.rpm

Once it is installed, you can use the following to connect to the web console.

HTTP://PutYourAsterisk@HomeIpaddressHere:10000

Remember, Webmin uses TCP port 10,000 and the SIP RTP uses UDP port 10,000 so there is no chance for conflict

9.4 How to use Shorewall Firewall to protect your A@H Server

As you probably already know, SIP and NAT don't play well together. If you're like me, you don't even want to deal with that mess and would rather just place the asterisk directly on the net to avoid any NAT issues. If you decide to go that route you better use a firewall on your asterisk server. I know that CentOS is a very secure operating system. However, you must still use a firewall on the server itself to have some peace of mind. Shorewall is a robust solution for our firewall needs on the A@H server.

I would like to thank Samy Antoun for his input. His tutorial on shorewall http://samyantoun.50webs.com/asterisk/firewall/firewall.htm helped me write this section.


9.4.1 What is Shorewall

According to the Shorewall Site: http://www.shorewall.net/ The Shoreline Firewall, more commonly known as "Shorewall", is a high-level tool for configuring Netfilter. You describe your firewall/gateway requirements using entries in a set of configuration files. Shorewall reads those configuration files and with the help of the iptables utility, Shorewall configures Netfilter to match your requirements. Shorewall can be used on a dedicated firewall system, a multi-function gateway/router/server or on a standalone GNU/Linux system. Shorewall does not use Netfilter's ipchains compatibility mode and can thus take advantage of Netfilter's connection state tracking capabilities.


9.4.2 How do I download and Install Shorewall?

We will download shorewall to the tmp directory and install it from there. Please be sure to check the shorewall web site for the latest RPM file and edit the following commands according to the lastest versions they offer.

At the CentOS Command Line type in the following commands to download and install Shorewall.

cd /tmp

wget http://www.invoca.ch/pub/packages/shorewall/2.2/shorewall-2.2.5/shorewall-2.2.5-1.noarch.rpm

rpm -ivh shorewall-2.2.5-1.noarch.rpm


9.4.3 How do I configure Shorewall

There are several files that need to be edited to setup our newly installed firewall. You can use nano at the command line to edit the files. They consist of the following files:

/etc/shorewall/interfaces
/etc/shorewall/masq
/etc/shorewall/policy
/etc/shorewall/routestopped
/etc/shorewall/rules
/etc/shorewall/shorewall.conf
/etc/shorewall/start
/etc/shorewall/zones

Each file is a text file with a pretty good description of what options you have and examples of how to use those options. I will not be cut and pasting the descriptions here because the wiki would become pretty large. I will however cut and paste a simple network setup and how it would be configured.

Our example will be a simple internet connection (it doesn't really matter if it's DSL, Cable, T1 or whatever). We will have a Static Internet IP address of Eth0 1.1.1.1 255.255.255.248. Remember, we don't.....


9.4.3.1 The Interfaces File

You must add an entry in this file for each network interface on your firewall system.

Our Example:
  1. ZONE INTERFACE BROADCAST OPTIONS
net eth0 detect routefilter,norfc1918,tcpflags
loc eth1 detect tcpflags
  1. LAST LINE — ADD YOUR ENTRIES BEFORE THIS ONE — DO NOT REMOVE

9.4.3.2 The Masq File

Use this file to define dynamic NAT (Masquerading) and to define Source NAT (SNAT).
(not sure why eth0 and eth1 need to be listed that way since there is no NAT occuring).

Our Example:
  1. INTERFACE SUBNET ADDRESS PROTO PORT(S) IPSEC
eth0 eth1
  1. LAST LINE — ADD YOUR ENTRIES ABOVE THIS LINE — DO NOT REMOVE


9.5 How to use IPCOP firewall to protect the A@H Server









9.6 The definitive guide to Sound Card Installation A@H 2.0-2.1

I would like to thank Tracy Carlton for an amazing well done guide and code.

Step 1: Edit the channels makefile to allow the compiling of chan_oss.so
Open/Edit /usr/src/asterisk/channels/makefile
On or about line 16 find:
CHANNEL_LIBS=chan_sip.so chan_agent.so chan_mgcp.so chan_iax2.so chan_local.so chan_skinny.so chan_features.so

Following chan_features.so add {space} chan_oss.so

Save and exit the file.

Step 2: Recompile Asterisk to build chan_oss.so
From the linux command line (via SSH/Putty or console) change to the Asterisk source code directory:

cd /usr/src/asterisk

Execute the rebuild by typing the following:

make clean {enter}

make {enter}

make install {enter}

Step 3: Checking on the chan_oss.so file
After the recompile completes verify that a chan_oss.so file exists in the correct directory: /usr/lib/asterisk/modules

Step 4: Edit the Asterisk/AMP startup script to allow Asterisk access to the soundcard device.
Open/Edit /usr/sbin/amportal
On or about line 34 find:
chown -R asterisk:asterisk /dev/zap
chown asterisk /dev/tty9

Insert the following line between these two lines:

chown -R asterisk:asterisk /dev/dsp

After which the lines should exactly match this:

chown -R asterisk:asterisk /dev/zap

chown -R asterisk:asterisk /dev/dsp

chown asterisk /dev/tty9

Now save and exit the file

Step 5: Create the Asterisk configuration file for chan_oss.so: oss.conf
Copy and paste exactly the text below:

;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console?  Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=from-internal
;
; Set overridecontext to yes if you want the context specified above
; to override what someone places on the command line.
;
overridecontext=yes
;
; Default extension to call
;
extension=s
;
; Default language
;
language=en
;
; Silence suppression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary.  Use the echo test to evaluate the best setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; On half-duplex cards, the driver attempts to switch back and forth between
; read and write modes.  Unfortunately, this fails sometimes on older hardware.
; To prevent the driver from switching (ie. only play files on your speakers),
; then set the playbackonly option to yes.  Default is no.  Note this option has
; no effect on full-duplex cards.
playbackonly=yes
;


Save this file to the Asterisk configuration directory via WinSCP or a Samba file share as: oss.conf in /etc/asterisk

After saving the file change its permissions to allow Asterisk to access it. From the linux command line (via Putty or direct) type the following:

cd /etc/asterisk {enter}

chmod 0777 oss.conf {enter}

chown asterisk:asterisk oss.conf {enter}

Step 6: Enable Asterisk to automatically load chan_oss.so upon startup.
From a web browser enter the following URL to directly access the “Config Edit� web editing utility:

http://xxx.xxx.xxx.xxx/maint/phpconfig/phpconfig.php

Substituting your Asterisk server’s IP Address for xxx.xxx.xxx.xxx.
Login as maint as you would normally.
Find the file modules.conf and click to open it.
On or about line 30 find:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
noload => chan_oss.so
;


Change the line containing the reference to chan_oss.so from noload to load.

load => chan_oss.so

Verifiy that the line containing the reference to chan_alsa.so is set to noload as shown below:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
load => chan_oss.so
;


Click the “Update� button but DO NOT click on the “Re-Read Configs� link yet.

Step 7: Add a custom dial plan script to access the soundcard as a paging port.
While still in the “Config Edit� web interface find and click to open: extensions_custom.conf
On or about line 16 find:


exten => *60,1,Answer
exten => *60,2,Playback(at-tone-time-exactly)
exten => *60,3,SayUnixTime(,,IMp)
exten => *60,4,Playback(beep)
exten => *60,5,Hangup


Insert the following lines directly ABOVE this block:

exten => *52,1,Dial(console/dsp)
exten => *52,2,Playback(custom/bosun)
exten => *52,2,Hangup()

Allow one blank line between each block.

The custom/bosun.gsm file is the bosun’s whistle sound effect used as an intercom/paging alert tone aboard the USS Enterprise from Classic Star Trek. Any short “alert� type sound such as the default beep.gsm can be used.

Click on the “Update� button and then click on the “Re-Read Configs� link.

Step 8: Hookup and Test the paging function.
If not already connected, connect the soundcard’s primary speaker output (if it has more than one) to the inputs of the paging amplifier or for testing use common computer speakers or headphones.
IMPORTANT NOTE: The soundcard may not produce ANY sound output unless there is some sort of resistance (load) being put on the output jack. For this reason always attach speakers, headphones, or an amplifier to the sound card and reboot before testing.

Reboot the Asterisk server.
After rebooting test the paging by dialing the “Paging Code� *52 from any handy telephone. If after adjusting volume levels on the speakers/headphones/amplifier you still have no sound at all skip to Step 10. Otherwise continue on to Step 9.

Step 9: Finish the Installation.
If not already connected to the final device do so now and adjust volume levels to suit.
YOU ARE DONE! Congratulations!!!

Step 10: Correcting Centos’ volume level issues.
Centos 4.x appears to set the soundcard’s outputs to muted and also a zero volume level requiring a mixing/tuning utility to correct these issues.
From the linux command line (via Putty or direct) execute the following command:

yum list | grep alsa {enter}

NOTE: This REQUIRES access to the internet.

Due to the fluid nature of linux the list returned may not match exactly the list below:

alsa-lib.i386
alsa-lib-devel.i386
alsa-utils.i386

Execute the following command:

yum –y install {item 1} {item 2} {item 3} {item x} {enter}

For example:

yum –y install alsa-lib.i386 alsa-lib-devel.i386 alsa-utils.i386 {enter}

After the download and installation completes, reboot the Asterisk server.

Step 11: Un-mute and tune the soundcard.
It is suggested that a pair of headphone be used for this step for best results.
After the reboot enter the following command from the linux command line (via Putty or direct) - For this step a keyboard and color monitor connected directly to the Asterisk server is highly recommended:

alsamixer {enter}

This SHOULD launch the alsamixer utility application. Alsamixer is a text based soundcard mixer/tuner. It is functionally identical and visually similar to the MS Windows sound card mixer/tuner. Each “channel� and/or “feature� is listed with it own control. Because of the large number of soundcard makes and models the exact number, labeling, and order of these channels will vary. However, there are only TWO channels that need to be adjusted and they are a constant: The MASTER channel and the PCM channel.

NOTE: It is recommended that prior to adjusting the levels that the paging code be dialed and an active paging session be underway. This will allow for “on-the-fly� real-time volume level setting to be done.
On-Screen, each channel consists of a “Label Box� at the base of a “Slider Bar� with a “Mute Status� indicator at the top. The channel currently being edited will have its name in the label box listed in red as opposed to white for all others. For switching or scrolling through channels use the left and right arrow keys.

To adjust the level of a channel use the up and down arrows.

To mute or un-mute a channel use the mute toggle key the “M� key.

To exit and save you level adjustments press the “ESC� key.

The levels of both the MASTER channel and the PCM channel must be adjusted for satisfactory volume levels and both channels MUST be UN-MUTED for correct functioning.

Starting with the far left or first channel, this should be the MASTER channel. Use the “M� key to un-mute the channel if it is muted (It most likely is muted.). Adjust the level to 55. This is a good “middle of the road� value to start testing with.

Using the left and right arrow keys scroll through the channels until you find the PCM channel and, if necessary, un-mute the channel then adjust its level to 55 as well.

If headphones are being used, test this level as being of “average� loudness and not to loud and/or distorted. If the level is to high adjust each channel down slightly until a “comfortable� level is reached. If the sound level is too quiet then adjust up the level in the same fashion.

After the levels have been adjusted to a satisfactory level press the “ESC� key to save the level settings and exit. These values are stored in a file that is parsed during startup making them, in effect, permanent. Once set, no changes should be required unless a hardware change occurs. Return to Step 9.


9.7 The definitive guide to Sound Card Installation, configuration and usage with A@H 2.2+

I would like to thank Tracy Carlton for writing this exceptional guide.

Step 1: Edit the Asterisk/AMP startup script to allow Asterisk access to the soundcard device.
Open/Edit /usr/sbin/amportal
On or about line 34 find:
chown -R asterisk:asterisk /dev/zap
chown asterisk /dev/tty9

Insert the following line between these two lines:

chown -R asterisk:asterisk /dev/dsp

After which the lines should exactly match this:

chown -R asterisk:asterisk /dev/zap
chown -R asterisk:asterisk /dev/dsp
chown asterisk /dev/tty9

Now save and exit the file.

Step 2: Create the Asterisk configuration file for chan_oss.so: oss.conf
Copy and paste exactly the text below:


;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console?  Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=from-internal
;
; Set overridecontext to yes if you want the context specified above
; to override what someone places on the command line.
;
overridecontext=yes
;
; Default extension to call
;
extension=s
;
; Default language
;
language=en
;
; Silence suppression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary.  Use the echo test to evaluate the best setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; On half-duplex cards, the driver attempts to switch back and forth between
; read and write modes.  Unfortunately, this fails sometimes on older hardware.
; To prevent the driver from switching (ie. only play files on your speakers),
; then set the playbackonly option to yes.  Default is no.  Note this option has
; no effect on full-duplex cards.
playbackonly=yes
;


Save this file to the Asterisk configuration directory via WinSCP or a Samba file share as: oss.conf in /etc/asterisk

After saving the file change its permissions to allow Asterisk to access it. From the linux command line (via Putty or direct) type the following:


cd /etc/asterisk {enter}
chmod 0777 oss.conf  {enter}
chown asterisk:asterisk oss.conf {enter}


Step 3: Enable Asterisk to automatically load chan_oss.so upon startup.
From a web browser enter the following URL to directly access the “Config Edit� web editing utility:

http://xxx.xxx.xxx.xxx/maint/phpconfig/phpconfig.php

Substituting your Asterisk server’s IP Address for xxx.xxx.xxx.xxx.
Login as maint as you would normally.
Find the file modules.conf and click to open it.
On or about line 30 find:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
noload => chan_oss.so
;


Change the line containing the reference to chan_oss.so from noload to load.

load => chan_oss.so

Verifiy that the line containing the reference to chan_alsa.so is set to noload as shown below:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
load => chan_oss.so
;


Click the “Update� button but DO NOT click on the “Re-Read Configs� link yet.

Step 4: Add a custom dial plan script to access the soundcard as a paging port.
While still in the “Config Edit� web interface find and click to open: extensions_custom.conf
On or about line 16 find:

exten => *60,1,Answer
exten => *60,2,Playback(at-tone-time-exactly)
exten => *60,3,SayUnixTime(,,IMp)
exten => *60,4,Playback(beep)
exten => *60,5,Hangup

Insert the following lines directly ABOVE this block:


exten => *52,1,Dial(console/dsp)
exten => *52,2,Playback(custom/bosun)
exten => *52,3,Hangup()

Allow one blank line between each block.

The custom/bosun.gsm file is the bosun’s whistle sound effect used as an intercom/paging alert tone aboard the USS Enterprise from Classic Star Trek. Any short “alert� type sound such as the default beep.gsm can be used.

Click on the “Update� button and then click on the “Re-Read Configs� link.

Step 5: Hookup and Test the paging function.
If not already connected, connect the soundcard’s primary speaker output (if it has more than one) to the inputs of the paging amplifier or for testing use common computer speakers or headphones.
IMPORTANT NOTE: The soundcard may not produce ANY sound output unless there is some sort of resistance (load) being put on the output jack. For this reason always attach speakers, headphones, or an amplifier to the sound card and reboot before testing.

Reboot the Asterisk server.
After rebooting test the paging by dialing the “Paging Code� *52 from any handy telephone. If after adjusting volume levels on the speakers/headphones/amplifier you still have no sound at all skip to Step 10. Otherwise continue on to Step 9.

Step 6: Finish the Installation.
If not already connected to the final device do so now and adjust volume levels to suit.
YOU ARE DONE! Congratulations!!!

Step 7: Correcting Centos’ volume level issues.
Centos 4.x appears to set the soundcard’s outputs to muted and also a zero volume level requiring a mixing/tuning utility to correct these issues.
From the linux command line (via Putty or direct) execute the following command:

yum list | grep alsa {enter}

NOTE: This REQUIRES access to the internet.

Due to the fluid nature of linux the list returned may not match exactly the list below:

alsa-lib.i386
alsa-lib-devel.i386
alsa-utils.i386

Execute the following command:

yum –y install {item 1} {item 2} {item 3} {item x} {enter}

For example:

yum –y install alsa-lib.i386 alsa-lib-devel.i386 alsa-utils.i386 {enter}

After the download and installation completes reboot the Asterisk server.

Step 8: Un-mute and tune the soundcard.
It is suggested that a pair of headphone be used for this step for best results.
After the reboot enter the following command from the linux command line (via Putty or direct) - For this step a keyboard and color monitor connected directly to the Asterisk server is highly recommended:

alsamixer {enter}

This SHOULD launch the alsamixer utility application. Alsamixer is a text based soundcard mixer/tuner. It is functionally identical and visually similar to the MS Windows sound card mixer/tuner. Each “channel� and/or “feature� is listed with it own control. Because of the large number of soundcard makes and models the exact number, labeling, and order of these channels will vary. However, there are only TWO channels that need to be adjusted and they are a constant: The MASTER channel and the PCM channel.

NOTE: It is recommended that prior to adjusting the levels that the paging code be dialed and an active paging session be underway. This will allow for “on-the-fly� real-time volume level setting to be done.
On-Screen, each channel consists of a “Label Box� at the base of a “Slider Bar� with a “Mute Status� indicator at the top. The channel currently being edited will have its name in the label box listed in red as opposed to white for all others. For switching or scrolling through channels use the left and right arrow keys.

To adjust the level of a channel use the up and down arrows.

To mute or un-mute a channel us the mute toggle key the “M� key.

To exit and save you level adjustments press the “ESC� key.

The levels of both the MASTER channel and the PCM channel must be adjusted for satisfactory volume levels and both channels MUST be UN-MUTED for correct functioning.

Starting with the far left or first channel, this should be the MASTER channel. Use the “M� key to un-mute the channel if it is muted (It most likely is muted.). Adjust the level to 55. This is a good “middle of the road� value to start testing with.

Using the left and right arrow keys scroll through the channels until you find the PCM channel and, if necessary, un-mute the channel then adjust its level to 55 as well.

If headphones are being used, test this level as being of “average� loudness and not to loud and/or distorted. If the level is to high adjust each channel down slightly until a “comfortable� level is reached. If the sound level is too quiet then adjust up the level in the same fashion.

After the levels have been adjusted to a satisfactory level press the “ESC� key to save the level settings and exit. These values are stored in a file that is parsed during startup making them, in effect, permanent. Once set, no changes should be required unless a hardware change occurs. Return to Step 6.


9.8 AsteriDex

Work in progress
Reference:
Introducing AsteriDex: Free Web-Based RoboDialer for Asterisk PBX Systems



9.9 AsteriDex II

Work in progress
Reference:
AsteriDex II: Free Web-Based RoboDialer for Asterisk

This is a free software from Nerd Vittles. AsteriDex is a web-based AutoDialer. The AsteriDex autodialer placed calls to all your favorite callees using a web interface. It stores and manage phone numbers in a MySQL database.
When you can click on a contact in a web interface, will AsteriDex initiate a call, AsteriDex will first call you at the number you designated for this contact, and then AsteriDex calls the number you clicked on.

For those lucky enough to have GrandStream's GXP-2000 IP phone with AutoAnswer, you could even configure AsteriDex to automatically activate the speakerphone and then place the call to the contact you selected.

In short, the original worked exactly like Microsoft's TAPI software without the configuration nightmare or your favorite (required) Micro$oft bloatware. Installation and configuration for your Asterisk@Home system was a snap and took less than 10 minutes. It also worked with vanilla Asterisk running the Asterisk Management Panel (AMP) software. AsteriDex was quick to implement and simple to use because it only did one thing, but did it well. The AsteriDex autodialer placed calls to all your favorite callees using a web interface. It's still simple to use, but today we've added two new features. First, you can use AsteriDex to automatically look up CallerID Names in your MySQL database for all your incoming calls. And, second, now you can dial the name of a person in your database by spelling up to five characters of the person's name using any phone in your home or office, and AsteriDex will automatically place the call for you just as if you'd use the web browser interface. Just dial 00 plus the one to five-digit code of the person to call.



9.10 Qmail

install qmail on asterisk@home

any idea if qmailtoaster.com or qmailrocks.org installer can works on a@h ???


9.11 Web Admin Interface Upgrade (Admin-UI v2.0)

A common complaint of the Asterisk@Home distribution is the visual appeal, and functional use of the main web interface/portal. Besides showing all options as text links, it must be manually edited to account for services that have been removed or to accomodate the growing list of 3rd-Party additions built to operate with Asterisk@Home installations.

To that end, the Open Source Projects initiative (http://www.kennonsoft.org/) at Kennon Software Corporation (http://www.kennonsoft.com/) created and continues to improve a drop-in replacement for the main Administration web interface of the Asterisk@Home installation. They have upgraded the visual styling to feel a bit more polished, made it dynamic so as to automatically detect which components are installed/running and adjust accordingly, as well as added support for a few common 3rd Party additions, and lastly added the ability to enable/customize an End-User Menu — while still providing (optionally secured) access to the full Administration portal as needed.

Reference:
Full Article, Files and Instructions by Kennon Software Corporation, Open Source Projects


9.12 NetMrg Network Bandwidth Monitoring

Its a good idea to look at your bandwidth usage to get a sense of how much of your pipe is being used by your server. Especially if you're using a purely IP solution. What you need is a good bandwidth monitor with fancy graphs showing you all the information you need to make good decisions. Netmrg is an excellent solution for this.

A@H 2.7 and earlier does not have NetMgr installed in the ISO so we have to do it manually. Our project manager has agreed to add it to the A@H repetoire for future versions.

Netmgr Home Page http://www.netmrg.net/
Netmgr Screeen Shots http://www.netmrg.net/screenshots.php
Netmgr Users Manual http://wiki.netmrg.net/wiki/Users_Manual




Chapter 9 Software that is not installed with Asterisk@Home

The following software is not installed with Astersik@Home but you definitely may want to consider using it.


9.1 Click-to-Dial using Microsoft Outlook and AstTapi

AstTapi is a Microsoft TAPI to Asterisk bridge that makes it possible to do click-to-dial from Microsoft Outlook and other TAPI compliant applications.

9.1.1 Download AstTapi and install it

Download this software from Sourceforge at http://sourceforge.net/projects/asttapi/.
Be sure that outlook is turned OFF before installing it. When finished, reboot the PC as requested.

9.1.2 Modifying the "Manager_Custom.conf" file in A@H (don't panic! this is easy!)

We need to make a quick edit to a A@H text file. You can do this from inside AMP or you can do this from the CentOS command line.

First we have to make a login for AstTapi. Click Maintenance under "AMP" and then click "Config Edit" and then click manager_custom.conf. In this file there already 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 your A@H server is on. This is NOT an IP address! It is the NETWORK address.

Now there is a chance that your phone may not even BE on the same network as the asterisk server. If this is true, you'll have to use 0.0.0.0/0.0.0.0 (which means any IP address from any subnet that has access to port 5038 can login). If this is true, you better use a better password then what is already there. Remember this login and password. Save the file. Then reload Asterisk.

9.1.3 Configuring AstTapi in outlook

Now we need to configure AstTapi inside of outlook.

Start outlook and select and click on a contact. There is a phone icon on the bar above the contact. Click that phone icon. A small window will appear with the following (use your imagination):

Number to dial
Contact: (Contact Name Field) (Open Contact Button)
Number: (Phone Number Field) (Dialing Properties Button)
(CHECK BOX Create new Journal Entry when starting new call)

(Start Call Button) (End Call Button) (Dialing Options Button) (Close Button)

Now that you can see the name and number you want to dial, click the "Dialing Options Button". In the "Connect Using Line" field, arrow down to "Asterisk" and click the "Line Properties" button right next to it.

Use the following entries in the next window:

Asterisk Server:
Host: ip of Asterisk server (you can use an IP address here or even a DNS address)
Port: 5038

User Information:

User: AstTapi (this might be case sensitive)

Password: AstTapi (or the password which you have chosen in the "manager_custom.conf" file)

User Channel: sip/200 (This is your extension. This is the number that will ring, requiring you to pick it up and get connected to the contact's number)

Context:

Select "Dial by 'Context'"

Context: outbound-allroutes (note, this is in every guide I can find, but didn't work for me on a default install. I had to use "Caller ID" and use my extension's CID and also check off "Attempt to set outgoing ID") (your milage may vary). (If neither work, try empty "Contect - Dial by Contect" or "Dial - Dial out by using the Dial application" fields)

Click Apply and OK and OK again

Now Click "Start Call" in the "New Call" box to begin the call.

Your extension will ring and once you pick it up, asterisk will connect you to the number of the contact you've chosen.


9.2 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 file to you Asterisk@Home server using WinSCP. 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/


9.3 Webmin - Web Based Linux Management

Webmin in a great package for managing a Linux box from the web. Webmin make it easy to manage all types of different service in a linux box (file manager, change passwords, etc). To install Webmin download the latest RPM from their web site and install it.

http://www.webmin.com/ - Webmin Homepage
http://sourceforge.net/projects/webadmin - Sourceforge Page

From the CentOS command line type the following:
(please note, check to see what is the LATEST version and file name of the webmin "noarch.rpm" file. If it is different from the example, be sure to use the latest file name. At the time of this edit, the file name was webmin-1.260-1.noarch.rpm)

wget http://heanet.dl.sourceforge.net/sourceforge/webadmin/webmin-1.270-1.noarch.rpm

rpm -Uvh webmin-1.270-1.noarch.rpm

Once it is installed, you can use the following to connect to the web console.

HTTP://PutYourAsterisk@HomeIpaddressHere:10000

Remember, Webmin uses TCP port 10,000 and the SIP RTP uses UDP port 10,000 so there is no chance for conflict

9.4 How to use Shorewall Firewall to protect your A@H Server

As you probably already know, SIP and NAT don't play well together. If you're like me, you don't even want to deal with that mess and would rather just place the asterisk directly on the net to avoid any NAT issues. If you decide to go that route you better use a firewall on your asterisk server. I know that CentOS is a very secure operating system. However, you must still use a firewall on the server itself to have some peace of mind. Shorewall is a robust solution for our firewall needs on the A@H server.

I would like to thank Samy Antoun for his input. His tutorial on shorewall http://samyantoun.50webs.com/asterisk/firewall/firewall.htm helped me write this section.


9.4.1 What is Shorewall

According to the Shorewall Site: http://www.shorewall.net/ The Shoreline Firewall, more commonly known as "Shorewall", is a high-level tool for configuring Netfilter. You describe your firewall/gateway requirements using entries in a set of configuration files. Shorewall reads those configuration files and with the help of the iptables utility, Shorewall configures Netfilter to match your requirements. Shorewall can be used on a dedicated firewall system, a multi-function gateway/router/server or on a standalone GNU/Linux system. Shorewall does not use Netfilter's ipchains compatibility mode and can thus take advantage of Netfilter's connection state tracking capabilities.


9.4.2 How do I download and Install Shorewall?

We will download shorewall to the tmp directory and install it from there. Please be sure to check the shorewall web site for the latest RPM file and edit the following commands according to the lastest versions they offer.

At the CentOS Command Line type in the following commands to download and install Shorewall.

cd /tmp

wget http://www.invoca.ch/pub/packages/shorewall/2.2/shorewall-2.2.5/shorewall-2.2.5-1.noarch.rpm

rpm -ivh shorewall-2.2.5-1.noarch.rpm


9.4.3 How do I configure Shorewall

There are several files that need to be edited to setup our newly installed firewall. You can use nano at the command line to edit the files. They consist of the following files:

/etc/shorewall/interfaces
/etc/shorewall/masq
/etc/shorewall/policy
/etc/shorewall/routestopped
/etc/shorewall/rules
/etc/shorewall/shorewall.conf
/etc/shorewall/start
/etc/shorewall/zones

Each file is a text file with a pretty good description of what options you have and examples of how to use those options. I will not be cut and pasting the descriptions here because the wiki would become pretty large. I will however cut and paste a simple network setup and how it would be configured.

Our example will be a simple internet connection (it doesn't really matter if it's DSL, Cable, T1 or whatever). We will have a Static Internet IP address of Eth0 1.1.1.1 255.255.255.248. Remember, we don't.....


9.4.3.1 The Interfaces File

You must add an entry in this file for each network interface on your firewall system.

Our Example:
  1. ZONE INTERFACE BROADCAST OPTIONS
net eth0 detect routefilter,norfc1918,tcpflags
loc eth1 detect tcpflags
  1. LAST LINE — ADD YOUR ENTRIES BEFORE THIS ONE — DO NOT REMOVE

9.4.3.2 The Masq File

Use this file to define dynamic NAT (Masquerading) and to define Source NAT (SNAT).
(not sure why eth0 and eth1 need to be listed that way since there is no NAT occuring).

Our Example:
  1. INTERFACE SUBNET ADDRESS PROTO PORT(S) IPSEC
eth0 eth1
  1. LAST LINE — ADD YOUR ENTRIES ABOVE THIS LINE — DO NOT REMOVE


9.5 How to use IPCOP firewall to protect the A@H Server









9.6 The definitive guide to Sound Card Installation A@H 2.0-2.1

I would like to thank Tracy Carlton for an amazing well done guide and code.

Step 1: Edit the channels makefile to allow the compiling of chan_oss.so
Open/Edit /usr/src/asterisk/channels/makefile
On or about line 16 find:
CHANNEL_LIBS=chan_sip.so chan_agent.so chan_mgcp.so chan_iax2.so chan_local.so chan_skinny.so chan_features.so

Following chan_features.so add {space} chan_oss.so

Save and exit the file.

Step 2: Recompile Asterisk to build chan_oss.so
From the linux command line (via SSH/Putty or console) change to the Asterisk source code directory:

cd /usr/src/asterisk

Execute the rebuild by typing the following:

make clean {enter}

make {enter}

make install {enter}

Step 3: Checking on the chan_oss.so file
After the recompile completes verify that a chan_oss.so file exists in the correct directory: /usr/lib/asterisk/modules

Step 4: Edit the Asterisk/AMP startup script to allow Asterisk access to the soundcard device.
Open/Edit /usr/sbin/amportal
On or about line 34 find:
chown -R asterisk:asterisk /dev/zap
chown asterisk /dev/tty9

Insert the following line between these two lines:

chown -R asterisk:asterisk /dev/dsp

After which the lines should exactly match this:

chown -R asterisk:asterisk /dev/zap

chown -R asterisk:asterisk /dev/dsp

chown asterisk /dev/tty9

Now save and exit the file

Step 5: Create the Asterisk configuration file for chan_oss.so: oss.conf
Copy and paste exactly the text below:

;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console?  Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=from-internal
;
; Set overridecontext to yes if you want the context specified above
; to override what someone places on the command line.
;
overridecontext=yes
;
; Default extension to call
;
extension=s
;
; Default language
;
language=en
;
; Silence suppression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary.  Use the echo test to evaluate the best setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; On half-duplex cards, the driver attempts to switch back and forth between
; read and write modes.  Unfortunately, this fails sometimes on older hardware.
; To prevent the driver from switching (ie. only play files on your speakers),
; then set the playbackonly option to yes.  Default is no.  Note this option has
; no effect on full-duplex cards.
playbackonly=yes
;


Save this file to the Asterisk configuration directory via WinSCP or a Samba file share as: oss.conf in /etc/asterisk

After saving the file change its permissions to allow Asterisk to access it. From the linux command line (via Putty or direct) type the following:

cd /etc/asterisk {enter}

chmod 0777 oss.conf {enter}

chown asterisk:asterisk oss.conf {enter}

Step 6: Enable Asterisk to automatically load chan_oss.so upon startup.
From a web browser enter the following URL to directly access the “Config Edit� web editing utility:

http://xxx.xxx.xxx.xxx/maint/phpconfig/phpconfig.php

Substituting your Asterisk server’s IP Address for xxx.xxx.xxx.xxx.
Login as maint as you would normally.
Find the file modules.conf and click to open it.
On or about line 30 find:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
noload => chan_oss.so
;


Change the line containing the reference to chan_oss.so from noload to load.

load => chan_oss.so

Verifiy that the line containing the reference to chan_alsa.so is set to noload as shown below:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
load => chan_oss.so
;


Click the “Update� button but DO NOT click on the “Re-Read Configs� link yet.

Step 7: Add a custom dial plan script to access the soundcard as a paging port.
While still in the “Config Edit� web interface find and click to open: extensions_custom.conf
On or about line 16 find:


exten => *60,1,Answer
exten => *60,2,Playback(at-tone-time-exactly)
exten => *60,3,SayUnixTime(,,IMp)
exten => *60,4,Playback(beep)
exten => *60,5,Hangup


Insert the following lines directly ABOVE this block:

exten => *52,1,Dial(console/dsp)
exten => *52,2,Playback(custom/bosun)
exten => *52,2,Hangup()

Allow one blank line between each block.

The custom/bosun.gsm file is the bosun’s whistle sound effect used as an intercom/paging alert tone aboard the USS Enterprise from Classic Star Trek. Any short “alert� type sound such as the default beep.gsm can be used.

Click on the “Update� button and then click on the “Re-Read Configs� link.

Step 8: Hookup and Test the paging function.
If not already connected, connect the soundcard’s primary speaker output (if it has more than one) to the inputs of the paging amplifier or for testing use common computer speakers or headphones.
IMPORTANT NOTE: The soundcard may not produce ANY sound output unless there is some sort of resistance (load) being put on the output jack. For this reason always attach speakers, headphones, or an amplifier to the sound card and reboot before testing.

Reboot the Asterisk server.
After rebooting test the paging by dialing the “Paging Code� *52 from any handy telephone. If after adjusting volume levels on the speakers/headphones/amplifier you still have no sound at all skip to Step 10. Otherwise continue on to Step 9.

Step 9: Finish the Installation.
If not already connected to the final device do so now and adjust volume levels to suit.
YOU ARE DONE! Congratulations!!!

Step 10: Correcting Centos’ volume level issues.
Centos 4.x appears to set the soundcard’s outputs to muted and also a zero volume level requiring a mixing/tuning utility to correct these issues.
From the linux command line (via Putty or direct) execute the following command:

yum list | grep alsa {enter}

NOTE: This REQUIRES access to the internet.

Due to the fluid nature of linux the list returned may not match exactly the list below:

alsa-lib.i386
alsa-lib-devel.i386
alsa-utils.i386

Execute the following command:

yum –y install {item 1} {item 2} {item 3} {item x} {enter}

For example:

yum –y install alsa-lib.i386 alsa-lib-devel.i386 alsa-utils.i386 {enter}

After the download and installation completes, reboot the Asterisk server.

Step 11: Un-mute and tune the soundcard.
It is suggested that a pair of headphone be used for this step for best results.
After the reboot enter the following command from the linux command line (via Putty or direct) - For this step a keyboard and color monitor connected directly to the Asterisk server is highly recommended:

alsamixer {enter}

This SHOULD launch the alsamixer utility application. Alsamixer is a text based soundcard mixer/tuner. It is functionally identical and visually similar to the MS Windows sound card mixer/tuner. Each “channel� and/or “feature� is listed with it own control. Because of the large number of soundcard makes and models the exact number, labeling, and order of these channels will vary. However, there are only TWO channels that need to be adjusted and they are a constant: The MASTER channel and the PCM channel.

NOTE: It is recommended that prior to adjusting the levels that the paging code be dialed and an active paging session be underway. This will allow for “on-the-fly� real-time volume level setting to be done.
On-Screen, each channel consists of a “Label Box� at the base of a “Slider Bar� with a “Mute Status� indicator at the top. The channel currently being edited will have its name in the label box listed in red as opposed to white for all others. For switching or scrolling through channels use the left and right arrow keys.

To adjust the level of a channel use the up and down arrows.

To mute or un-mute a channel use the mute toggle key the “M� key.

To exit and save you level adjustments press the “ESC� key.

The levels of both the MASTER channel and the PCM channel must be adjusted for satisfactory volume levels and both channels MUST be UN-MUTED for correct functioning.

Starting with the far left or first channel, this should be the MASTER channel. Use the “M� key to un-mute the channel if it is muted (It most likely is muted.). Adjust the level to 55. This is a good “middle of the road� value to start testing with.

Using the left and right arrow keys scroll through the channels until you find the PCM channel and, if necessary, un-mute the channel then adjust its level to 55 as well.

If headphones are being used, test this level as being of “average� loudness and not to loud and/or distorted. If the level is to high adjust each channel down slightly until a “comfortable� level is reached. If the sound level is too quiet then adjust up the level in the same fashion.

After the levels have been adjusted to a satisfactory level press the “ESC� key to save the level settings and exit. These values are stored in a file that is parsed during startup making them, in effect, permanent. Once set, no changes should be required unless a hardware change occurs. Return to Step 9.


9.7 The definitive guide to Sound Card Installation, configuration and usage with A@H 2.2+

I would like to thank Tracy Carlton for writing this exceptional guide.

Step 1: Edit the Asterisk/AMP startup script to allow Asterisk access to the soundcard device.
Open/Edit /usr/sbin/amportal
On or about line 34 find:
chown -R asterisk:asterisk /dev/zap
chown asterisk /dev/tty9

Insert the following line between these two lines:

chown -R asterisk:asterisk /dev/dsp

After which the lines should exactly match this:

chown -R asterisk:asterisk /dev/zap
chown -R asterisk:asterisk /dev/dsp
chown asterisk /dev/tty9

Now save and exit the file.

Step 2: Create the Asterisk configuration file for chan_oss.so: oss.conf
Copy and paste exactly the text below:


;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console?  Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=from-internal
;
; Set overridecontext to yes if you want the context specified above
; to override what someone places on the command line.
;
overridecontext=yes
;
; Default extension to call
;
extension=s
;
; Default language
;
language=en
;
; Silence suppression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary.  Use the echo test to evaluate the best setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; On half-duplex cards, the driver attempts to switch back and forth between
; read and write modes.  Unfortunately, this fails sometimes on older hardware.
; To prevent the driver from switching (ie. only play files on your speakers),
; then set the playbackonly option to yes.  Default is no.  Note this option has
; no effect on full-duplex cards.
playbackonly=yes
;


Save this file to the Asterisk configuration directory via WinSCP or a Samba file share as: oss.conf in /etc/asterisk

After saving the file change its permissions to allow Asterisk to access it. From the linux command line (via Putty or direct) type the following:


cd /etc/asterisk {enter}
chmod 0777 oss.conf  {enter}
chown asterisk:asterisk oss.conf {enter}


Step 3: Enable Asterisk to automatically load chan_oss.so upon startup.
From a web browser enter the following URL to directly access the “Config Edit� web editing utility:

http://xxx.xxx.xxx.xxx/maint/phpconfig/phpconfig.php

Substituting your Asterisk server’s IP Address for xxx.xxx.xxx.xxx.
Login as maint as you would normally.
Find the file modules.conf and click to open it.
On or about line 30 find:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
noload => chan_oss.so
;


Change the line containing the reference to chan_oss.so from noload to load.

load => chan_oss.so

Verifiy that the line containing the reference to chan_alsa.so is set to noload as shown below:


; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
load => chan_oss.so
;


Click the “Update� button but DO NOT click on the “Re-Read Configs� link yet.

Step 4: Add a custom dial plan script to access the soundcard as a paging port.
While still in the “Config Edit� web interface find and click to open: extensions_custom.conf
On or about line 16 find:

exten => *60,1,Answer
exten => *60,2,Playback(at-tone-time-exactly)
exten => *60,3,SayUnixTime(,,IMp)
exten => *60,4,Playback(beep)
exten => *60,5,Hangup

Insert the following lines directly ABOVE this block:


exten => *52,1,Dial(console/dsp)
exten => *52,2,Playback(custom/bosun)
exten => *52,3,Hangup()

Allow one blank line between each block.

The custom/bosun.gsm file is the bosun’s whistle sound effect used as an intercom/paging alert tone aboard the USS Enterprise from Classic Star Trek. Any short “alert� type sound such as the default beep.gsm can be used.

Click on the “Update� button and then click on the “Re-Read Configs� link.

Step 5: Hookup and Test the paging function.
If not already connected, connect the soundcard’s primary speaker output (if it has more than one) to the inputs of the paging amplifier or for testing use common computer speakers or headphones.
IMPORTANT NOTE: The soundcard may not produce ANY sound output unless there is some sort of resistance (load) being put on the output jack. For this reason always attach speakers, headphones, or an amplifier to the sound card and reboot before testing.

Reboot the Asterisk server.
After rebooting test the paging by dialing the “Paging Code� *52 from any handy telephone. If after adjusting volume levels on the speakers/headphones/amplifier you still have no sound at all skip to Step 10. Otherwise continue on to Step 9.

Step 6: Finish the Installation.
If not already connected to the final device do so now and adjust volume levels to suit.
YOU ARE DONE! Congratulations!!!

Step 7: Correcting Centos’ volume level issues.
Centos 4.x appears to set the soundcard’s outputs to muted and also a zero volume level requiring a mixing/tuning utility to correct these issues.
From the linux command line (via Putty or direct) execute the following command:

yum list | grep alsa {enter}

NOTE: This REQUIRES access to the internet.

Due to the fluid nature of linux the list returned may not match exactly the list below:

alsa-lib.i386
alsa-lib-devel.i386
alsa-utils.i386

Execute the following command:

yum –y install {item 1} {item 2} {item 3} {item x} {enter}

For example:

yum –y install alsa-lib.i386 alsa-lib-devel.i386 alsa-utils.i386 {enter}

After the download and installation completes reboot the Asterisk server.

Step 8: Un-mute and tune the soundcard.
It is suggested that a pair of headphone be used for this step for best results.
After the reboot enter the following command from the linux command line (via Putty or direct) - For this step a keyboard and color monitor connected directly to the Asterisk server is highly recommended:

alsamixer {enter}

This SHOULD launch the alsamixer utility application. Alsamixer is a text based soundcard mixer/tuner. It is functionally identical and visually similar to the MS Windows sound card mixer/tuner. Each “channel� and/or “feature� is listed with it own control. Because of the large number of soundcard makes and models the exact number, labeling, and order of these channels will vary. However, there are only TWO channels that need to be adjusted and they are a constant: The MASTER channel and the PCM channel.

NOTE: It is recommended that prior to adjusting the levels that the paging code be dialed and an active paging session be underway. This will allow for “on-the-fly� real-time volume level setting to be done.
On-Screen, each channel consists of a “Label Box� at the base of a “Slider Bar� with a “Mute Status� indicator at the top. The channel currently being edited will have its name in the label box listed in red as opposed to white for all others. For switching or scrolling through channels use the left and right arrow keys.

To adjust the level of a channel use the up and down arrows.

To mute or un-mute a channel us the mute toggle key the “M� key.

To exit and save you level adjustments press the “ESC� key.

The levels of both the MASTER channel and the PCM channel must be adjusted for satisfactory volume levels and both channels MUST be UN-MUTED for correct functioning.

Starting with the far left or first channel, this should be the MASTER channel. Use the “M� key to un-mute the channel if it is muted (It most likely is muted.). Adjust the level to 55. This is a good “middle of the road� value to start testing with.

Using the left and right arrow keys scroll through the channels until you find the PCM channel and, if necessary, un-mute the channel then adjust its level to 55 as well.

If headphones are being used, test this level as being of “average� loudness and not to loud and/or distorted. If the level is to high adjust each channel down slightly until a “comfortable� level is reached. If the sound level is too quiet then adjust up the level in the same fashion.

After the levels have been adjusted to a satisfactory level press the “ESC� key to save the level settings and exit. These values are stored in a file that is parsed during startup making them, in effect, permanent. Once set, no changes should be required unless a hardware change occurs. Return to Step 6.


9.8 AsteriDex

Work in progress
Reference:
Introducing AsteriDex: Free Web-Based RoboDialer for Asterisk PBX Systems



9.9 AsteriDex II

Work in progress
Reference:
AsteriDex II: Free Web-Based RoboDialer for Asterisk

This is a free software from Nerd Vittles. AsteriDex is a web-based AutoDialer. The AsteriDex autodialer placed calls to all your favorite callees using a web interface. It stores and manage phone numbers in a MySQL database.
When you can click on a contact in a web interface, will AsteriDex initiate a call, AsteriDex will first call you at the number you designated for this contact, and then AsteriDex calls the number you clicked on.

For those lucky enough to have GrandStream's GXP-2000 IP phone with AutoAnswer, you could even configure AsteriDex to automatically activate the speakerphone and then place the call to the contact you selected.

In short, the original worked exactly like Microsoft's TAPI software without the configuration nightmare or your favorite (required) Micro$oft bloatware. Installation and configuration for your Asterisk@Home system was a snap and took less than 10 minutes. It also worked with vanilla Asterisk running the Asterisk Management Panel (AMP) software. AsteriDex was quick to implement and simple to use because it only did one thing, but did it well. The AsteriDex autodialer placed calls to all your favorite callees using a web interface. It's still simple to use, but today we've added two new features. First, you can use AsteriDex to automatically look up CallerID Names in your MySQL database for all your incoming calls. And, second, now you can dial the name of a person in your database by spelling up to five characters of the person's name using any phone in your home or office, and AsteriDex will automatically place the call for you just as if you'd use the web browser interface. Just dial 00 plus the one to five-digit code of the person to call.



9.10 Qmail

install qmail on asterisk@home

any idea if qmailtoaster.com or qmailrocks.org installer can works on a@h ???


9.11 Web Admin Interface Upgrade (Admin-UI v2.0)

A common complaint of the Asterisk@Home distribution is the visual appeal, and functional use of the main web interface/portal. Besides showing all options as text links, it must be manually edited to account for services that have been removed or to accomodate the growing list of 3rd-Party additions built to operate with Asterisk@Home installations.

To that end, the Open Source Projects initiative (http://www.kennonsoft.org/) at Kennon Software Corporation (http://www.kennonsoft.com/) created and continues to improve a drop-in replacement for the main Administration web interface of the Asterisk@Home installation. They have upgraded the visual styling to feel a bit more polished, made it dynamic so as to automatically detect which components are installed/running and adjust accordingly, as well as added support for a few common 3rd Party additions, and lastly added the ability to enable/customize an End-User Menu — while still providing (optionally secured) access to the full Administration portal as needed.

Reference:
Full Article, Files and Instructions by Kennon Software Corporation, Open Source Projects


9.12 NetMrg Network Bandwidth Monitoring

Its a good idea to look at your bandwidth usage to get a sense of how much of your pipe is being used by your server. Especially if you're using a purely IP solution. What you need is a good bandwidth monitor with fancy graphs showing you all the information you need to make good decisions. Netmrg is an excellent solution for this.

A@H 2.7 and earlier does not have NetMgr installed in the ISO so we have to do it manually. Our project manager has agreed to add it to the A@H repetoire for future versions.

Netmgr Home Page http://www.netmrg.net/
Netmgr Screeen Shots http://www.netmrg.net/screenshots.php
Netmgr Users Manual http://wiki.netmrg.net/wiki/Users_Manual

Created by: GinelLipan, Last modification: Tue 20 of Sep, 2011 (23:46 UTC) by admin
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