时间：2021-07-01 10:21:17 帮助过：22人阅读
1.0 - Hardware platform used in the creation of this document Installing on a Dell Dimension 9150 Intel 630 CPU (3.0GHz) 1GB RAM Zaptel card: Digium Developers kit (TDM 400P with 1 FXO module, 1 FXS module) Vonage line connected to the FXO
1.0 - Hardware platform used in the creation of this document
Installing on a Dell Dimension 9150
Intel 630 CPU (3.0GHz)
1GB RAM
Zaptel card:
Digium Developer’s kit (TDM 400P with 1 FXO module, 1 FXS module)
Vonage line connected to the FXO module
Standard cordless telephone connected to the FXS module
2.0 - Download and install Trixbox v1.2
Download Trixbox v1.2 from SourceForge:
http://sourceforge.net/project/showfiles.php?group_id=123387&package_id=192286
Burn the ISO to a CD and boot to it
*** WARNING ***
This CD will completely destroy whatever data is on the computer you are booting to. Make sure this is what you want to do before proceeding.
When Trixbox splash screen opens, hit Enter
*** NOTE: Trixbox does not seem to work if you choose the i586 option...instead, just hit Enter and take the default mode.
For keyboard type, take default (US)
Select appropriate time zone and hit OK
Type in your root password twice and hit OK
System will now format and install Trixbox v1.2
When the install finishes, the CD tray will open. Remove the CD (or else it will start over). System will reboot 2-3 times, and then you will end up at the login prompt.
*** NOTE: If the system does reboot with the CD still in, you can simply remove the CD and press CTRL+ALT+DEL at the Trixbox splash screen...it will continue normally upon the next reboot.
*** NOTE: Sometimes the setup seems to freeze on 'munin-install.' It is NOT actually frozen, so do not reboot or else the Trixbox install has not completed. Go grab a cup of coffee because sometimes it takes 7-10 minutes for this part to complete successfully.
Once the install has finished, log in as root with the root password you entered above.
3.0 - Setting up your server
3.1 - Network configuration
Set up IP address information by typing netconfig.
Select Yes when prompted about setting up networking.
Select to use DHCP if you want (not recommended) otherwise, enter in IP information for your Trixbox box.
I will use the following information for my home setup:
IP: 192.168.200.16
Netmask: 255.255.255.0
Gateway: 192.168.200.254
Primary nameserver: 4.2.2.2 (This is one of AT&T's public name servers. If you have your own DNS server, enter it in this field).
Click OK and restart the network with ‘service network restart.’
Secondary nameserver – if you would like to add a secondary nameserver for backup DNS purposes, nano /etc/resolv.conf and add a line ‘nameserver (ip address)’ underneath the primary nameserver information. My resolv.conf now looks like this:
nameserver 4.2.2.2
nameserver 4.2.2.1
To exit nano, hit 'CTRL+X' and then 'Y' when asked if you want to save.
3.2 - Install Webmin
Webmin is a valuable tool used for the configuration of Linux-based servers. I install Webmin by default on all of my Linux boxes simply due to it's ease of use. Webmin installs an HTTP-based GUI which you can get to by using port 10000 from a browser.
To install Webmin, first you need to download the RPM file. There are two ways to do this. First, you can go to www.webmin.com, download the latest RPM, and then get it to your Trixbox box somehow (typically I would use FTP to do this). Alternatively, you can skip the middleman and download the Webmin RPM direct from the linux CLI using wget. I typically download installs to /usr/local.
The wget command looks like this:
cd /usr/local
wget superb-east.dl.sourceforge.net/sourceforge/webadmin/webmin-1.290-1.noarch.rpm
Once you have the RPM file downloaded, install it by running:
rpm -ivh webmin-1.290-1.noarch.rpm
You can now get to your Webmin console by putting the following into a browser that exists on your LAN:
http://192.168.200.16:10000 (obviously, replace my IP with your Trixbox IP)
4.0 - Zaptel Driver
Setting up your Zaptel driver has typically been a problem with Trixbox v1.1 and earlier, however I did not run into any issues with v1.2.
From command line do ‘ztcfg –vv’ – if there are no errors, you should get an output that says:
Zaptel Configuration
======================
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
2 channels configured.
Or similar based on your Zaptel hardware configuration. The above output is from Digium's standard developer's kit TDM400 (mentioned in chapter 1).
*** NOTE: If there are no channels showing up when doing 'ztcfg -vv', try running 'genzaptelconf' at the Linux CLI to see if it will detect your card.
*** NOTE: If you are still having Zaptel driver issues, please follow the steps in my v1.1 Setup Guide, Chapter 4: http://www.sureteq.com/asterisk/trixbox.htm#4.0_-_Zaptel_Issues
5.0 - General Trixbox Setup/Security
Now it is time to configure Asterisk/Trixbox
Using a web browser, connect to your new setup by typing in the IP address. For my box, I will use http://192.168.200.16.
To enable SSL web browsing to the Trixbox web console, run the following commands from the Linux CLI:
yum -y install mod_ssl
service httpd restart
You can now connect to https://192.168.200.16.
You should see the Trixbox web console. If you get ‘Page Can Not Be Displayed,’ go back and verify your IP settings.
Click on ‘System Administration.’ Enter the following info:
Username: maint
Password: password
First, let’s secure Trixbox. On the command prompt, change the ‘admin’ password by typing ‘passwd admin’ and then put in a new password twice.
Update maint password by typing ‘passwd-maint’ at the command line. Enter the password twice.
Update AMP password by typing 'passwd-amp' at the command line. Enter the password twice.
Update meetme password by typing 'passwd-meetme' at the command line. Enter the password twice.
Update all Cent-OS packages by typing 'yum -y update' at the command line.
Now, let's secure SSH. This is an ***optional*** step, but I like to allow access via SSH2 only.
cd /etc/ssh
nano sshd_config
Find this div (should be just below the opening comments...about 12 lines down):
#Port 22
#Protocol 2,1
#ListenAddress 0.0.0.0
#ListenAddress ::
Uncomment the first three lines and change them to this:
Port 22
Protocol 2
ListenAddress 192.168.200.16 #(your asterisk IP address)
#ListenAddress :: (leave this commented out...unless you are using IPv6...which you are probably not).
Type 'CTRL+X' to exit nano and 'Y' to save changes.
From the Linux CLI, do:
service sshd restart
You can now connect to your Trixbox with SSH2 by using an SSH client such as PuTTY: http://www.chiark.greenend.org.uk/~sgtatham/putty/
5.1 - Install Trixbox Dynamic UI v2.1
This is an ***optional*** step. The Trixbox Dynamic UI v2.1 is a nicer looking interface than the stock index.php that comes with Trixbox. It also allows you to customize what options you have from both a client and administrative standpoint. It is developed by Kennon Software and can be found at www.kennonsoft.org.
First, download and install the UI. Open up a CLI and do the following:
cd /var/www/html
mv index.php index-old.php (backup your original index.php file)
wget http://www.kennonsoft.org/projects/trixbox/admin-ui-21.tgz
tar -xzf admin-ui-21.tgz
rm admin-ui-21.tgz
chmod 777 welcome/index.dat
Your Dynamic UI is now installed. Next, we will want to configure it.
Open up https://(your asterisk IP) in a browser. You should see the following:
Click on 'Administration...' in the bottom right-hand corner.
Click on Menu Config.
You can choose to uncheck 'Enable End-user Menu' at the top of the screen...this saves you a step by going straight to the administration menu when you browse to the Dynamic UI home page.
Select all of the modules and click 'Update' followed by 'Done.' You should now see this when you browse to your Asterisk box:
6.0 - Module Install
Now to get into Asterisk and install some modules. Modules are different software packages used by Asterisk for different applications. For instance, if you want voice mail, you would install the Voicemail module. This allows you to pick and choose your Asterisk features.
Click on Asterisk Mgmt (FreePBX). Once FreePBX has opened, click on ‘Tools’ in the upper right, and then ‘Module Admin’ on the left.
Lets start with some basic modules (you can add and remove modules at any time).
Check the boxes next to Core, Feature Code Admin, Time Conditions, Voicemail, On Hold Music, IVR, Queues, Recordings and Backup & Restore.
Make sure ‘Enable Selected’ is in the drop-down box and click Submit.
See the red bar that just popped up at the top of the page? Click on the red bar anytime you make changes. This reloads the Asterisk configuration.
You can now click on ‘Setup’ in the upper right to get to the modules you have just activated.
7.0 - Dial Plan
Before we start configuring, we need to come up with a dial plan. Basically, you want to map out exactly what you want your Trixbox to do prior to configuring it. This will help guide you to the appropriate configuration. Keep in mind that a Dial Plan is not something you configure in Asterisk...step away from the computer and pull out a piece of paper to design your Dial Plan.
I have 1 inbound Vonage line, and three extensions ready to be configured. I like to use 1xx extensions internally, and for inbound, I want to set up some hours of operation so that I’m not bothered by clients after 7pm. With this simple setup, my dial plan will look like this:
Inbound:
8:00am – 7:00pm – Go to main greeting (Thanks.wav). Greeting states that:
2 gets me (forwards to my cell phone)
3 gets my business partner (forwards to his cell phone)
4 to reach first available representative (round robin between our two cell phones)
7:00pm – 8:00am – Go to closed greeting (Closed.wav). Greeting states that our office is closed, and to please call back between 8am and 7pm PST.
Outbound:
All calls should be routed out the FXO port of my Digium TDM400 (my Vonage line). – I may change this later to include my IAX line if the Vonage line is busy, but let’s keep it simple for now.
Internal extensions:
101 – Grandstream Budge Tone 100 SIP phone
102 – ExpressTalk softphone on my Windows computer
111 – Cordless telephone that is connected to the FXS port of my Digium TDM400
That’s it for my dial plan…very simple.
8.0 - Configure Internal Extensions
8.1 - Zaptel extensions
I will start by setting up my extensions. In FreePBX, click on ‘Setup’ and then ‘Extensions.’
We’ll start with my cordless phone (extension 111 – connected to my Digium TDM400).
Click on ‘ZAP’
I used these settings:
Extension number: 111
Display name: Cordless
Direct DID:
DID Alert Info:
Outbound CID:
Emergency CID:
Record Incoming: On Demand
Record Outgoing: On Demand
Channel: 1
Voicemail & Directory: Enabled
Voicemail password: 111 (same as extension number…keeping it simple)
Email address: (my email address)
Pager email address:
Email attachment: Yes (since I like getting the messages in my email)
Play CID: No
Play Envelope: No
Delete Vmail: Yes (this way, the voice mails are delivered to my email inbox only…if set to no, once I delete the emails, I also have to go into the voicemail and delete the voicemail manually)
Vm options:
Vm context: default
Hit ‘Submit’
Click the red bar to apply the options.
I now get a dial tone on my cordless phone.
8.2 - Soft phone extension
Next, we’ll set up a soft phone so that I can dial between internal extensions.
I use the X-Lite soft phone which can be downloaded from http://www.xten.com/index.php?menu=download.
In FreePBX, click on ‘Setup’ and then ‘Extensions.’
Click on ‘SIP.’
I used these settings:
Extension number: 102
Display name: Soft phone
Direct DID:
DID Alert Info:
Outbound CID:
Emergency CID:
Record Incoming: On Demand
Record Outgoing: On Demand
Secret: 12345 (can be whatever you want)
Dtmfmode: rfc2833
Voicemail & Directory: Enabled
Voicemail password: 102
Email address: (my email address)
Pager email address:
Email attachment: Yes (since I like getting the messages in my email)
Play CID: No
Play Envelope: No
Delete Vmail: Yes (this way, the voice mails are delivered to my email inbox only…if set to no, once I delete the emails, I also have to go into the voicemail and delete the voicemail manually)
Vm options:
Vm context: default
Hit ‘Submit’
Click the red bar to apply the options.
Now to configure the X-Lite soft phone.
Upon first installing, the SIP configuration pops up automatically, otherwise, you can click on the down arrow at the top of the phone and choose ‘SIP Account Settings.’
Click ‘Add.’
Display Name: Soft Phone
User Name: 102
Password: 12345 (my ‘secret’ from above)
Authorization user name: 102
Domain: 192.168.200.16 (your Trixbox IP address)
Domain Proxy
Register with domain and receive incoming calls (checked)
Target domain selected
Click OK.
Click Close on the SIP Accounts window.
I now try dialing my cordless phone x111 and it works! From the cordless, I dial my Soft Phone x102 and it works as well.
8.3 - Endpoint Manager
Trixbox v1.2 now comes with an endpoint manager which is really slick.
I have a GrandStream GXP-2000 SIP phone on my desk and plugged into the network. It has already been configured as IP 192.168.200.107. I now need to create an extension for my GXP-2000. In FreePBX, click on ‘Setup’ and then ‘Extensions.’
Click on ‘SIP.’
I used these settings:
Extension number: 101
Display name: GXP-2000
Direct DID:
DID Alert Info:
Outbound CID:
Emergency CID:
Record Incoming: On Demand
Record Outgoing: On Demand
Secret: 12345 (can be whatever you want)
Dtmfmode: rfc2833
Voicemail & Directory: Enabled
Voicemail password: 101
Email address: (my email address)
Pager email address:
Email attachment: Yes (since I like getting the messages in my email)
Play CID: No
Play Envelope: No
Delete Vmail: Yes (this way, the voice mails are delivered to my email inbox only…if set to no, once I delete the emails, I also have to go into the voicemail and delete the voicemail manually)
Vm options:
Vm context: default
Hit ‘Submit’
Click the red bar to apply the options.
Close out of FreePBX and open up your Asterisk Mgmt. screen. Click on 'Endpoint Manager' to access it.
I see that the 'Map Devices' div has already filled in my local network, so I click 'Go.'
After about 20 seconds, it has found my GXP-2000:
I click on the MAC Address of my GXP-2000, and I am now taken to the configuration screen:
I select the Budge Tone SIP extension I created above (101) and hit 'Submit.' My GrandStream phone is now configured.
9.0 - Outbound Routes
9.1 - Outbound route with default Zap trunk
Now that we have set up our internal extensions, let’s focus on getting calls out. To do this, in FreePBX, click on 'Setup Outbound Routes.'
By default, Trixbox has already created a trunk out of my FXO port in the Digium TDM400 card (trunk Zap/g0), and has already created a route which makes the user dial 9 to get an outside line (0 9_outside). Lets click on the ‘0 9_outside’ route on the right hand side of the screen and add a few more dial patterns for outbound dialing.
The only dial pattern so far should be ‘9|.’ Under Dial Patterns, you should see ‘Insert.’ This allows you to insert pre-defined dial patterns for commonly dialed numbers. I picked the following patterns:
Local 7/10 digit
Toll Free
Information
Emergency
Click ‘Submit Changes’
Click the red bar at the top of the screen to apply the changes.
My Dial Pattern for route 0 9_outside now looks like this:
311
411
911
1800NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
9|.
NXXNXXXXXX
NXXXXXX
I try dialing my cell phone with my cordless x111, and it works!
9.2 - Setting up an IAX2 trunk using VoipJet
Since I only have a single Vonage line out to the Internet, I'd like to use a VoIP provider to provide some redundancy. To accomplish this, I use an IAX VoipJet line (available from www.voipjet.com). This is a pretty solid service, and is easy to set up. Plus, it allows you to just put in as much money as you need, and is not a monthly-charge type of setup.
First, I click on 'Trunks' and then click 'Add IAX2 trunk.' I used these settings:
Outbound caller ID: "SureTeq"
Maximum channels: (leave blank)
Dial Rules: 1NXXNXXXXXX
1818+NXXXXXX
(These dial rules are 'as is' for a call outside my area code (10 digit), but for calls within 818, or 7 digit dialed numbers, it adds the 1818 to the front)
Outgoing Settings
Trunk Name: (userid)@voipjet (userid is a 4 digit number)
PEER Details:
auth=md5
context=from-internal
host=test.voipjet.com
secret=(my secret)
type=peer
Leave Incoming settings and the rest of the page blank.
Click Submit and then the red bar at the top of the screen to apply these settings to Asterisk.
Here is what it should look like once completed:
One more thing we need to do is to nano /etc/asterisk/iax_custom.conf and add the following:
[voipjet]
type=peer
host=test.voipjet.com
username= (your username)
secret= (your secret)
auth=md5
context=from-internal
To test whether or not this worked, click on 'Outbound Routes', and then '0 9_outside.' Under 'Trunk Sequence,' change the drop-down box from 'Zap/g0' to 'IAX2/(username)@voipjet.' Click Submit and then click on the red bar at the top to apply the changes. Try making an outbound call to see if your IAX trunk works properly.
9.3 - Setting the IAX trunk as a second Outbound route
Now that we've successfully set up our IAX trunk, we would like to use it as a backup for our Zap trunk. Click on 'Outbound Routes', and then '0 9_outside.' Under 'Trunk Sequence,' set the first drop-down box to 'Zap/g0' and the second drop-down box to 'IAX2/(username)@voipjet.' Click Submit and then click on the red bar at the top to apply the changes. You should now be able to make multiple outbound calls.
9.4 - Setting up a BroadVoice trunk
(Chapter 9.4 submitted by Thomas Cirillo)
I have a BroadVoice account (www.broadvoice.com) which provides me with a local area code to call my system. The FreePBX GUI is used to configure the trunk and no editing of sip.conf, extensions.conf or other conf files is necessary.
Outbound Caller ID: “Caller name” <#######>
Maximum Channels: (I left this blank)
Outgoing dial rules were left blank also.
Trunk Name: Broadvoice
PEER Details:
authname=
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=
host=sip.broadvoice.com
insecure=very
secret=password *(read below on where to find this)
type=peer
user=phone
username=
(*)The password used as the secret in peer details is not the same as the account password with the service provider. When you log into www.broadvoice.com with your account name and select the “Account” tab; To the right of the “My Devices” div there is an HTML tag for “Show Settings”; Select this and note the “auth_id: & auth_password:”
10.0 - Incoming calls/IVR setup
Now, I need to tell my Trixbox what to do with incoming calls.
10.1 - System recordings
But first, let’s take care of the necessary recordings for my Digital Receptionist.
I want two different messages…one for during business hours, and one for when we are closed. The scripts will look something like this:
Thanks.wav – “Thank you for calling Schw00d’s Asterisk emporium. Please press 2 for Schw00d, 3 for Jim, and 4 for the next available associate. Thank you and have a wonderful day.”
Closed.wav – “Thank you for calling Schw00d’s Asterisk emporium. Our office is now closed. Please call back between the hours of 8am and 7pm Pacific Standard time. Thank you.” (click).
To get these recorded first clear your throat and put on your best announcer voice.
Click on System Recordings.
You can either record your greetings as .WAV files in an external application (8-bit mono recordings work best over the phone lines), or straight into an extension. We’ll record using our extension.
Put your extension number into the extension field and click ‘Go.’
Pick up the extension that you put into the extension field and dial *77. You will hear a beep…start your recording. When you are done, hang up the extension. Type in a name for your recording and click ‘Save.’
10.2 - Digital Receptionist
Click on Digital Receptionist and then click Add IVR.
I used these settings for my initial IVR:
Change Name: BusinessHours
Timeout: 10
Enable Directory: Checked
Directory Context: default
Enable Direct Dial: checked
Announcement: Thanks (my pre-recorded Thanks.wav file)
For my options, I will have 3, so I leave the Increase/Decrease default, but if you have more or less options, feel free to add/remove them.
Option 1:
2 – Core – Cordless <111>
3 – Core – Soft phone <102>
4 – Core – Budge Tone <101>
Save
Click on the red bar at the top of the screen.
(Note: I know I said that I was going to have options 2 and 3 forward to cell phones…I’m getting there, but I just want to get this set up and working with local extensions first).
Click on ‘Add IVR’
Change Name: AfterHours
Timeout: 10
Enable Directory: Checked
Directory Context: default
Enable Direct Dial: checked
Announcement: Closed (my pre-recorded Closed.wav file)
Option 1:
2 – Core – Cordless <111>
3 – Core – Soft phone <102>
4 – Core – Budge Tone <101>
(Note: The options are exactly the same as my BusinessHours IVR except for the name and the announcement…why the heck did I do that? Well…because my Thanks.wav file lists the possible extensions (2,3,4), and my Closed.wav does not…however I still want specific people to be able to call my extension after hours, so even though the after hours recording does not list the options…the options are still available).
10.3 - Time Conditions
Setting up time conditions will be the next step. Click on Time Conditions. I used these settings:
Time Condition Name: Incoming
Time to match:
Time to start: 08:00
Time to finish: 19:00
Week Day Start: Monday
Week Day Finish: Sunday (we’re open 7 days!)
(Leave the rest blank)
Destination if time matches:
IVR: BusinessHours (my daytime-created IVR)
Destination if time does not match:
IVR: AfterHours (my nighttime-created IVR)
Click ‘Submit Changes’
Click the red bar at the top of the screen to apply the settings.
10.4 - Inbound Route
Next, we want to add our incoming route. Click on ‘Inbound Routes.’ You can route based on DID channel, Caller ID Number, Zaptel channel, or if all of those are left blank, it will route all un-matched (by DID, CID or Zap channel) calls to your settings.
DID Number:
Caller ID Number:
Zaptel Channel:
Fax Extension: disabled (for now)
(Leave the rest of the fax stuff default)
Privacy Manager: No
Alert Info:
Set Destination:
Time Conditions: Incoming (the one we just created)
Click Submit
Click the red bar at the top of the screen to apply changes.
You will receive a warning stating that Leaving the DID and Caller ID Number empty will match all incoming calls received not routed using any other defined incoming Route. Are you sure? Click OK.
Click the red bar at the top of the screen to apply the settings.
At this point, we can simulate an incoming call by dialing 7777 from one of our extensions. You should hear one of the two recordings we did for the Digital Receptionist.
11.0 - On Hold Music
Next, we will set up our hold music.
Click on ‘On Hold Music.’ I typically delete all of the included mp3’s and choose to upload my own. You have to always have 1 mp3 available for hold music, so you’ll have to upload at least one song prior to deleting all of the included music.
Browse for a .wav or .mp3 file that you would like to be played as your hold music, and click ‘Upload.’
Click on the red bar at the top of the screen to apply the settings.
You can repeat this for as many songs as you’d like to upload.
One nice thing about Asterisk is that it only plays hold music while someone is actually on hold. So, if someone hears 30 seconds of an on-hold song, and then is taken off of hold, the song will pick up where it left off the next time someone is put on hold.
12.0 - Forward calls to a cell phone
Since I only have a single Vonage line out to the Internet, I need to use VoIP to forward an incoming call through the Internet to my cell phone. For this, we will use the IAX2 trunk that we set up in chapter 9.
Click on 'Extensions' and then click 'Custom.' Enter the following information:
Extension number: 112
Display Name: My cell phone
(leave all of the rest blank until you get to 'dial.')
dial: IAX2/(username)@voipjet/18185551212 (my cell number on the end there)
Voicemail & Directory: Disabled
Click 'Submit' and then click the red bar at the top to apply the changes to Asterisk.
Now, we go back to our Digital Receptionist and change the extension we originally set up to the custom VoipJet number.
Click on Digital Receptionist. Under ‘2’ which is currently going to Core:Cordless<111>, we want to change that to:
Core: My cell phone <112>
Now, when a caller calls in and presses '2,' it will automatically forward that call to my cell phone. In addition, if an internal user dials extension 112, it will also forward to my cell phone.
13.0 Simple Queue setup
13.1 Forward to cell phones queue
I would now like to set up a new queue which routes inbound callers to the cell phone extensions I created in chapter 12. I had set up extension 112, however lets say that I also have extension 113. This is the exact same concept for internal extension queues.
Click on Queues on the left hand side of the FreePBX screen.
I used the following information to set this up:
Queue number (extension for the queue): 200
Queue name: CellRoundRobin
Queue password:
CID name prefix:
Static agents: 112
113
Agent announcement:
Hold Music Category: Default (an interesting note...you can set up different hold music contexts for different applications such as standard on-hold, or separate queues...so for instance, if you would like callers to hear perhaps a song followed by an advertisement for your company, you would set that up as a different hold music category...for our purposes however, I will leave it default).
Max wait time: Unlimited
Max callers: 0 (unlimited)
Join empty: Yes
Leave when empty: No
Ring Strategy: roundrobin
Agent timeout: 15 seconds
Retry: 5 seconds
Wrap-up-time: 0 seconds
Call recording: No
Caller announcements: 0 seconds
Announce position: No
Announce Hold Time: No
Voice Menu: None
Join Announcement: None
Fail Over Destination:
Core: Voicemail box 111 (my cordless...since this ACD transfers calls to cell phones, this is kind of a moot point for the application).
Click Submit Changes
Click the red bar at the top of the screen to submit the changes.
You should now be able to pop into Panel and see your new Queue. If you dial 200 from an internal extension, the call will go to extension 112 (my cell phone). If you dial it again, it will go to 113, and so forth.
14.0 - Panel configuration
Now that we have some good extensions, trunks, and queues in our Trixbox box, we should take a look at the FOP (Flash Operator Panel) or simply Panel. It shows up next to 'Reports' on the main selection screen. When you click panel, you will see your extensions, queues, and trunks all laid out nicely. If you double-click on any of the green buttons, you will be prompted for a password.
First, lets set that password to something we can use. (The default password is: passw0rd).
Go to your CLI and type the following:
cd /var/www/html/panel
nano op_server.cfg
About 42 lines down, you'll see security_code=passw0rd. Change the 'passw0rd' to whatever password you would like, and hit CTRL+X and then Y and ENTER to exit nano.
Now, restart the panel with amportal restart from the CLI, and you're good to go.
There are a few useful things you can do with the FOP. For instance, if a call is in session, you can click on the RED button to disconnect the call. If you would like to call an extension, you can drag the phone icon located to the right of the extension number to another extension. This initiates the call to both phones. To transfer a call, drag the phone icon from the extension you want to disconnect to the new extension, and the call is transferred. For more information on the FOP, please see http://www.aussievoip.com/wiki/TB-FOP.
15.0 - HUD Lite configuration
HUD Lite and HUD Server are products that are now bundled with Trixbox. They are another tool for visually controlling what your extensions are doing.
For my Trixbox v1.2 installation, Hudlite-server wasn't installed by default, but I did find a file which did the trick.
I ran:
/usr/local/sbin/install-hudlite
This ran through a quick install routine and started the hudlite service. I'm not sure if my installation was weird, or if this is how to get it installed with v1.2. If anyone knows for sure, please email me.
Next, lets configure some extensions in the HUD Manager. Click on HUD Manager and then click on New Device.
Let's add one of our extensions:
Device: SIP102
Ext: 102
Name: Soft Phone
Username: username (this is the HUD username only...can be anything)
Password: password (this is the HUD password only...can be anything)
Email/IM/Cell Phone are optional
Click 'Save.'
Repeat this for each extension you want monitored.
Time to download and install HUD-Lite. You can get HUD-Lite at www.hudlite.org. Download the latest version and install onto your Windows PC using all defaults.
Once inside HUD-Lite, choose File --> Settings. Click the '+' next to Advanced Settings and enter the following information:
Username: (the username you used for one of your extensions above)
Password: (the password you used for one of your extensions above)
Server name: (the IP address of your Trixbox server)
Password: (the HUD-Server password - by default it is 'password')
Server port: 6600
Login timeout: 30
Park extension: 9000
Click 'Apply' and 'Close.'
HUD-Lite should now connect to your Trixbox and show you the extensions you set up:
HUD-Lite tells you a few things when calls are in progress with colors:
Green - extension is on an outside call
Purple - extension is on an intra-office call
Orange - extension is on a queue call
Gray - extension is unavailable
You can drag and drop your extension (upper-left corner extension 101) to another extension to call that extension (will dial your phone first and then call the 2nd extension once you have picked up).
You can also drag and drop your call to the 'on hold' div in the upper left to put a call on hold (although this didn't seem to work too well for me). You can also drag and drop a call into the envelope icon of another extension to send that call directly to the voicemail of that extension.
HUD-Lite also allows you to Barge/Monitor a call in progress by clicking the 'B' next to a call that is in progress. This will ring your extension and when you pick it up, you will hear the conversation that is going on between the two other extensions, or to the outside world. Muahahaha...big brother in the house.
16.0 - SugarCRM configuration
SugarCRM is a bundled contact management software very similar to ACT!, but with Asterisk/Trixbox integration. With a few simple steps, you can get this to work with Trixbox.
First, open the SugarCRM application and put in the default username/password of Admin / password.
The first thing we want to do is change our Admin password for security, so click on 'My Account' in the upper right-hand corner. Then click on the 'Change Password' button underneath 'Users: Administrator (Admin) in the center-left of the screen. Enter and confirm your new password and click 'Save.'
Now it's time to set up your extension. Click on 'My Account' again and then click the 'Edit' button. Change 'Asterisk Phone Extension' to your Asterisk extension. I am using extension 102 (my Windows XP computer's soft phone). Click 'Save' to save that information.
Let's add a couple more contacts. Click on the contacts tab and then select 'Create Contact' from the left hand Shortcuts menu.
I start by adding my other extensions as contacts:
Last name: Cordless
Office phone: 111
Save
Add another:
Last name: GrandStream (my Grandstream SIP phone)
Office phone: 101
Save
Add another:
First name: Chris
Last name: Sherwood
Office phone: (my cellular number)
At this point, I try calling one of my extensions by clicking on contacts, and then clicking on the little phone icon next to their name. It rings my soft phone x102 which I pick up, then dials extension 111. It works!
I won't go much into SugarCRM, but as you can see, it can be a very powerful tool. You can add multiple users who will each have their own settings/contacts/etc.