In past times, it would have been quite costly to have a "personal" PBX. Using spare hardware, open source software, and low-cost service providers, it can now be done for almost nothing. The key advance is the open source Asterisk IP PBX and the Asterisk@Home package that includes Asterisk and a web-based GUI configuration tool.
You will have to set up three main components: the IP PBX itself, the phones (or softphones) to be used with it, and the gateway service that lets you call other people on the PSTN. I will describe how to set up each of these. I assume you already have a home network and broadband access. If you are behind a NAT firewall, it does not matter – you don't have to do anything special like running the IP PBX outside the NAT. All that matters is that you have enough bandwidth (upstream and downstream) to carry voice traffic.
The IP PBX
You will need a computer to run the IP PBX. While you may already have a Linux server at home, I do not recommend using it to also run your phones. If you want to do that, you will have to figure out how to load and configure the PBX software on your own, although the rest of the instructions here will be helpful.
If you have an old PIII class machine lying around, then use that. If not, you can buy one on eBay for about $40.
I used a Dell OptiPlex GX1, a PIII 450MHz system with 128M RAM. It has a built-in sound and Ethernet, so no additional hardware is required.
You will load Asterisk@Home on this computer. It will take it over – it starts by formatting the hard disk, so make sure there is nothing on the machine that you want to keep. Detailed instructions are given below.
You can buy SIP phones from Grandstream, SNOM, Cisco, even Avaya. Or you can buy an adapter from SNOM or Cisco and use an analog phone. To get started, though, it is easiest to get a softphone and run it on another computer. We will describe how to obtain and install the XLite softphone on a Windows machine to use for a phone. You will also use this Windows machine to administer the IP PBX, through a browser. If you want to use a Linux machine instead, you will need to get an appropriate softphone for it, but you can still use its browser to administer the IP PBX.
I presume you want to communicate with others on the PSTN network, so you need to obtain gateway service. Since part of the call is being carried on the circuit switched network, it costs real money. This means you will have to pay for this part of the system.
I will describe how to set up access to two services, one for outgoing calls and another for incoming calls.
VoipJet is used for outgoing calls. It is priced at 1.3 cents per minute (USA), with no monthly minimum or service charges.
BroadVoice is used for incoming calls. You get a phone number (you can choose the area code and exchange). They have various plans, but I signed up for one that is $5.95 per month, with unlimited incoming minutes. You can also make outgoing calls using this account (it includes 100 minutes per month, and 3.9 cents per minute additional). There is a $9.95 one time activation fee to sign up for BroadVoice.
So if you have the hardware, you can set up your IP PBX for a total of $9.95 and run it for $5.95 per month.
I assume you have a home network, and that you are setting up behind a Gateway Router (otherwise known as a NAT firewall). You will need to pick a static IP address for your IP PBX that is on your home network.
How Much Will This Cost?
I assume you have broadband service, a router, and a Windows machine to run the softphone.
If you already have a spare computer that you can dedicate to this project, there is no cost at all for equipment, unless you need to buy an audio headset for the softphone. If you do not have a spare computer, then you can buy one on eBay for about $60, including shipping. This does not include a monitor, but you don't need a monitor except briefly when you first set it up. I assume you can borrow a monitor (or that you have a KVM switch).
Your only other initial cost will be the $9.95 activation fee to BroadVoice.
Your monthly cost will be $5.95 for incoming service from BroadVoice, and 1.3 cents per minute for outgoing calls to VoipJet. If you only make a few outgoing calls, you could drop VoipJet, and stay within the 100 outgoing minutes that you get from BroadVoice. If, say, you use 100 minutes of VoipJet outgoing calls, it would cost an additional $1.30.
So you can build and experiment with your own IP PBX for an investment of less than $100, and for an ongoing cost of less than $100 per year. This is a lot less than I am now paying for my phone service from AT&T. In fact, a whole year's service would be less than my monthly phone bill. Hmmm…
Step 1 – Sign Up for Service
This section takes you through signing up with VoipJet and BroadVoice. I am using two service because
- it is less costly, if you use it a lot, and
- it illustrates how to set up two different kinds of trunks.
Browse to http://www.voipjet.com
. Sign up for service.
Then log on and follow the line describing how to set up Asterisk. You will need to copy down your "VoipJet account number (username)," your "Authorization code (password)," and your server IP address (depending on your location). Ignore the rest of the setup instructions.
Browse to http://broadvoice.com
. Sign up. Say "I want to use my own SIP device". When it asks what type of device, select "Not Listed (Generic SIP). When it asks for details, just say "Asterisk". Next, pick your phone number. Next, pick your plan. I recommend "BroadVoice BYOD Lite." Once you have finished the signup process, log in, click on "Account" and follow the "Show Settings" link in the "Your Devices" section. This section gives your phone number (in case you forget) and your password (you will need this later).
You should also click on the "Support Center" link on the left and bring up the "Step By Step Installation Guide" for "Bring Your Own Devices." Select "Asterisk." You must follow the instructions in Section 3 to find the "right proxy" to use. This involves pinging their proxies to find which has the lowest latency. You will use this later.
2. Pick the one with the lowest latency. In my case, it was proxy.dca.broadvoice.com.
Step 2 – Set Up Home Network
Pick an IP address for the IP PBX. You will need to find an unused address on your subnet outside the range assigned by your DHCP server. I picked 192.168.0.40. If you pick a different IP address, you will have to adjust the instructions accordingly.
Step 3 – Set Up SIP Softphone
Run XLite. Click on the "Menu" icon to configure it. Click on "System Settings", then "SIP Proxy", then "Default".
Fill in the following fields:
User name: 200
Authorization user: 200
SIP Proxy: 192.168.0.40
It should look something like this.
The phone will try to register, but for now it will fail.
Step 4 – Set Up Asterisk@Home
The rest of the instructions explain how to install and configure Asterisk@Home. From here on, I will refer to it as AAH.
AAH is a package consisting of several major components. These were developed and supported relatively independently. The "Asterisk" part is the core IP PBX, and the "@Home" part consists of applications, a provisioning system, an installer, and an operating system that, together, make a complete package. The major components that make up AAH are
- Asterisk, the core PBX
- Sugar, a CRM system
- Flash Operator Panel, a screen-based operator's console
- Web Meet Me Control, a meet me conferencing control application
- Asterisk Management Portal (AMP), a web-based provisioning tool for Asterisk
- A report system, part of AMP, which provides CDR reporting tools
- A Maintenance system, also part of AMP, which provides low level interfaces to some components and real time system information
- CentOS, a version of Linux related to from Red Hat Enterprise Linux (but without Red Hat branding and support).
Get the Software
This is going to reformat your hard disk and load everything from the operating system on up, so make sure there is nothing on the hard drive that you want to save.
Boot your machine from the CD. When it prompts, type ENTER. Then wait as everything loads and compiles. This could take 30 minutes or more, depending on how fast your computer, hard drive, and CDROM are. At the end, it ejects the CD and reboots from the hard drive.
The initial login is:
Set Up Networking
The machine probably got an IP address from DHCP, but it is not what you want. Log in as root and run:
It will display setup information. You should enter the following:
IP address: 192.168.0.40
DNS Server: 192.168.0.1
OK these changes, then reboot the machine to make them take effect.
After reboot, long in once more. You need to add the BroadVoice server IP address to /etc/hosts.
Edit /etc/hosts, and add the following line at the end:
Here the IP address was the one obtained when signing up with BroadVoice.
Once this is done, you can do the rest of the configuration through the web.
Browse To AMP
Select "Asterisk Management Portal." Log in as follows:
Now you should be at the AMP main screen.
Click on "Setup" to bring you to the main setup screen.
Set Up BroadVoice Trunk
Click on "Trunks" and then "Add SIP Trunk." You see a blank SIP trunk form.
You will need to fill out the main items on the SIP/Trunk screens. I am using BroadVoice for incoming service only, so I will not administer outgoing trunk information.
Outbound caller ID: 7237570239
Max channels: 1
The outgoing settings can be left as is, except to fill in the trunk name as BroadVoice.
In Incoming Settings, fill in the following:
User Context: 7327570239
The format of the Register String is: username:password:phone_number@provider_domain. For BroadVoice this is what you will want to use (replacing with your BroadVoice phone number & password):
Once all of this information is entered, you can click the Submit button. Once you do, a red line will appear at the top of the page. To "apply" your changes you must click this red line (this will cause AAH to tell the Asterisk engine to reload its config files).
Set Up VoipJet Trunk
Click on "Trunks" and then "Add IAX2 Trunk." You see a blank IAX2 trunk form.
Fill in the fields as follows:
Outbound Caller ID: 7327570239
Maximum channels: 1
The outgoing dial rules control how numbers are processed before they are sent to the trunk. In the case of VoipJet, phone numbers must be in the form "1+Area Code+Local Number". If the number appears to have an area code but not a "1", then it is added. If it does not have an area code, then "1732" is added (I am in area code 732). So the dial rules are:
This time we will fill in outgoing settings.
Trunk name:: voipjet
secret=*********** (get this from voipjet config info)
host=220.127.116.11 (get this from voipjet config info)
username=***** (get this from voipjet config info)
You can leave the "Incoming Settings" and the "Register String" blank.
When you are done, the form appears as follows.
Set Up Extensions
Now it is time to define the extensions. I will set up two extensions, number 200 in the basement and number 201 in the study. I have already shown how to configure XLite extension 200, and you can configure another extension like it for 201. Or you can set up a SIP phone instead.
Click on "Extensions" on the left, and "Add an Extension." You will see the extensions page.
The information you will add is the following:
Full name: basement
Voice Mail Password: 1234
Email Address: <your email address>
When you're finished adding the extension, you can see the details of the extension by clicking on its name on the right-hand side of the AMP interface (as shown above).
Add extensions 201 similarly. Remember to click on the red bar, to make Asterisk take not of your changes.
Check for Phone Registration
At this point, the XLite application should have registered with AAH. If not, exit XLite (you have to stop it from the tray icon) and restart. It should say "Logged in". If not, see the troubleshooting section below. To start with, make sure you have assigned the same password in the extension form and in the phone.
On XLite, dial *23 to test audio input and output levels.
Set Up Ring Group
Now we set up a ring group, so that we can ring all the extensions at once. This is useful, for instance, so that incoming calls can alert at all extensions.
On the left, click "Ring Groups" and on the right "Add Ring Group".
Fill out the form as follows:
Group Number: 1
Ring Time: 18
Destination if no answer: Voicemail basement <200>
Submit the changes, and click the red bar to make them take effect.
Set Up Digital Receptionist
A digital receptionist is an application that answers incoming calls, interacts with callers, and forwards their call on. It allows callers to dial an internal extension, to access the company directory, or to reach selected destinations according to predefined dial patterns. We will set it up to dial by extension, to give the company directory, and to ring all extensions if the caller presses "1".
The digital receptionist set up involves a series of screens and actions, not all of which will be illustrated. Part of the setup includes recording a message to be played to callers. You can record this independently, and download the MP3 file, or you can use XLite to dial in and make the recording. The latter method will be illustrated. In general, the setup instructions are clear and explicit, so this will only give the essentials.
Click "Digital Receptionist" on the left. Enter 200 as the current extension number. On XLite, dial *77 and record a message. It goes something like this:
Hello. You have reached Charles Hayden's experimental Asterisk phone system. You can dial a three digit extension. Dial 1 to ring all extensions . Or dial pound for a directory.
Hang up and dial *99 to hear the message. Rerecord if necessary.
Name the recording "top greeting" and add a description.
On the next page, for "Number of options for Menu top greeting enter 1.
On the next screen, choose "Ring Group #1" as the action. This means that when the caller enters "1" it will go to Ring Group #1, which rings all extensions.
Set Up Outbound Routing
Click on the "Outbound Routing" link and fill in the following:
Route Name: outgoingVoipJet
Dial Patterns: 011.
Trunk Sequence: IAX2/voipjet
The dial patterns given here will allow you to dial international calls, long distance with 1+area code, and calls within the 732 area code with only the area code but without the 1. This is how it works here in the 732 area. If you can dial locally without the area code, then you could add "NXXXXXX" as well. As we have seen, the trunks cannot necessarily accept all these dialing combinations, so they have their own rules to screen and transform them.
If you've made it this far you should be able to dial "7777" on your SIP phone - this will simulate an incoming phone call - and you should hear your greeting. At this point the system should have enough functionality to dial out through VoipJet.
Go ahead and call a number and it should ring at the other end. Remember that you need to dial 1 followed by a 10 digit number. If it does then pat yourself on the back! Only a few more steps are needed to get incoming calls to work (you can try to call your BroadVoice number, but it won't work yet).
Set Up Incoming Calls
Now we will set up incoming calls to go to the digital receptionist. Choose the "Incoming Calls" link, and click the "Digital Receptionist" "top greeting".
So far we've done everything through the AMP web GUI. Now we have to drop into editing text files. This is obviously not desirable, but this is how it is right now, so you have to go through this. You can edit extensions.conf through the AMP web GUI. Click the Maintenance link at the top of the AMP page and then click the Config Edit link in the left-hand menu. Then click on extensions.conf.
Look on the left for from-sip-external and click it. By default, AAH sends all incoming SIP calls to a congestion status. This means that all incoming calls will ring fast-busy to the caller and they will never make it into our AAH system.
There are four lines under the [from-sip-external] section. One is already commented out with a semi-colon (;). Comment out the remaining three lines by adding semi-colons. Now we need to add two new "exten" lines.
The first is:
which tells AAH to wait for 1 second once a call has been detected from an external SIP device. This makes sure we don't clip off the first part of our greeting message. The second line we need to add is:
exten => _.,2,Goto(from-pstn,s,1).
When we're done the section should look like this:
;give external sip users congestion and hangup
;exten => _.,1,AbsoluteTimeout(15)
;exten => _.,2,Congestion
;exten => _.,3,Hangup
exten => _.,1,Wait(1)
exten => _.,2,Goto(from-pstn,s,1)
When you're finished and you've saved your changes, you need to click the link at the top of the config edit screen labled Re-Read Configs. This will cause Asterisk to reload its configuration files and make any changes take effect.
Test Incoming Calls
Now is the moment of truth. If everything went smoothly, you should now be ready to test an inbound call to your AAH system. From a phone (POTS or cell, either will do) dial your BroadVoice number. You should hear your greeting! At this point you'll probably be giddy that it's all working. Go ahead and listen to your greeting a few times (dial 9 to repeat it). Then either dial "1" or your extension and your SIP phone should start ringing. You now have Asterisk@Home configured and working with your BroadVoice account.
Set up Passwords
You need to change the default passwords, so that people do not break into your system. There are a variety of passwords, used for different things.
AMP allows you to assign different login accounts to access the Maintenance section separately from the other sections (Setup, Reports, and Panel). If you log into AMP with the maintenance account, you will be able to access everything, but if you log in with the admin account, you will have to log in also to get to the maintenance section.
Command to Change Password
log in to console
log into AMP GUI, access Setup, Reports, Panel
log into AMP Maintenance section
log into Web Meetme Control
for checking system mail (Maintenance / Web Mail login)
To change these passwords, you must log into the console as root, either on the physical console or using a ssh
client such as putty (which can be obtained from http://www.chiark.greenend.org.uk/~sgtatham/putty/
) from a windows platform or using ssh from a Linux platform.
If things did not work the first time, you can track down the trouble using the Asterisk console. This can be used to display SIP messages coming through the system, as well as steps executed in interpreting the dial plan instructions.
If you are more comfortable using Ethereal, you can use that instead (or also) to see the SIP messages. I will not describe how to download or configure it, other than to point out that it can be obtained fromhttp://www.ethereal.com
To use the Asterisk console, go to the Linux console or enter through ssh. Remember the username is root and the password is password. Give the command:
This will attach to the console with verbose mode set. Give the command
to enable SIP debugging. You can turn off SIP debugging using the command
Softphone will not register
XLite registers by sending a REGISTER message, which is challenged. It then sends another REGISTER with credentials, which is accepted. If you do not see these REGISTER messages, then XLite proxy setting is probably not right. If you see the challenge but it still rejects the REGISTER with credentials, the passwords are probably mismatched. The password abc123 should be used in both places. Also, be sure the phone "Authorization user" is set to 200.
If you are having trouble making outgoing calls, check the username and secret in the IAX2 trunk. Make sure you can ping the host address listed there. The username is a 4 digit number that you find from your account settings on VoipJet.
On a message trace, you should see an INVITE going out, followed by receipt of TRYING and OK (when the far end answers). If there is no response, then perhaps you have the wrong IP address. If the INVITE is immediately followed by a 4XX or 5XX error, maybe the username or password is wrong.
If the incoming BroadVoice trunk is set up properly, Asterisk will register successfully, and will renew the registration every 10 seconds. Look for an outgoing REGISTER, a challenge, another outgoing REGISTER with credentials, and an incoming 200 OK. If you get no response, or either REGISTER yields an error reply, then look at the host, secret, and user fields from the SIP trunk.
If the REGISTER succeeds, then when you call (from a POTS phone) you should see an INVITE being received. If you have not properly changed extensions.conf, then the system will reply with an error rather than with TRYING. Look at the debug output to see how the INVITE is being processed. You should see the Wait and the GoTo that you entered into extensions.conf. If you see these, and the call is still not answered, you might want to change Incoming Calls to route the call to a specific phone, for instance to "basement <200>". This will bypass the Digital Receptionist and the Ring Group for now. If this fixes things, look at those two screens. If not, make sure the "context=from-pstn" is present in the incoming trunk user details. If this still does not fix things, you will have to follow the trace as it steps through the contexts and steps in extensions.conf.
This section describes some things you might want to do to further customize your system. I am experimenting myself, and presenting the results here. These instructions might not be the most effective way to achieve the desired results. If you find a better way, please let me know. The instructions in this section are briefer, reflecting your increased level of experience with AAH.
Weather for external callers
You can modify the digital attendant so that it can read the weather to callers. Remember to modify your digital attendant announcement recording if desired ("press 2 for weather").
- Modify the Digital Receptionist. For "Number of options" give "2". On the next screen, for option 2, give "Custom App" and enter
- Go into "Maintenance", "Config Edit", "extensions_custom.conf". At the end, add:
exten => s,1, Goto(from-internal-custom,*61,1)
You can customize the weather for your own city be
. You need to edit $custpath
. Don't forget to modify the announcement a few lines down. You can ftp to weather.noaa.gov
to see the state/city combinations that are available.
You can set up the BroadVoice trunk to carry outgoing calls. Since you get 100 "free" minutes per month, you might want to use these up first, before switching over to VoipJet. It would be great if the system could do this for you, but for now there is no easy way to do this. So instead, let's up a dial-9 trunk for BroadVoice.
- Open the BroadVoice trunk. Add "Peer Settings" of:
- Go to "Outbound Routing". Add a new route. Name it "outgoingBroadVoice." Add the following dial patterns. These mimic the local dialing rules, which permit "1" followed by a ten digit number, or "732" followed by a seven digit number. An initial "9" selects the broadvoice trunk, but it is stripped off before the number is dialed.
Set the "Trunk sequence" to SIP/broadvoice."
Set Up Voice Mail
By default, when you set up an extension, and add an email address in the "email address" field, then when the system records a message, it sends email containing a link that will let the recipient listen to the voice message on their computer (through the browser). For this to work, you must open up your Asterisk machine to browser traffic from outside the router's firewall. I leave it to the reader to figure out this one.
Once you do, you still need to change the email, which contains a link to the mail reading web page.
- Go to "Maintenance" and then "Config Edit" and choose "vm_email.inc".
- Replace the string "192.168.1.101" with the external name or IP address of your Asterisk system. In other words, you need to form a URL that can be used to browse to the Asterisk system from where you will be reading the mail, which is probably outside your NAT router.
- After you click "Update" do not forget to select "Re-Read Configs".
- Test it by calling extension 200, leaving voice mail, receiving the email, and following the link.
- You will be prompted to log in. Use extension number 200, and its password "1234". You can then retrieve and listen to the email.
You may notice a few things wrong with the voice mail reader application. I guess this means that it is not really finished, although it appears to basically work. I noted that the initial login does not fill in your extension, as it should. Also the image links are broken. Maybe the next version will clean up these details.
Add Free World Dialup IAX Trunk
Free World Dialup (FWD, at http://www.freeworlddialup.com/
) is a free VoIP service provider. Using it, you can call other FWD users, and through cooperative arrangements with other service providers, subscribers on many other VoIP networks. In addition, you can use their gateways to call to the PSTN (but only toll free numbers, since someone still has to pay for other kinds of calls).
You first need to sign up for a FWD "phone number," which you can do at the web site listed above. Be sure to enable IAX service while you are there. My number is 679263. I will use this in the examples below, but you should replace it with your own number and password, as appropriate.
- Open the Trunks page, select "Add Trunk" and select "Add IAX2 Trunk". Enter the following information. I have chosen to use a prefix of "393" (FWD) to select the FWD trunk. You can pick a different trunk prefix if you want.
Outbound Caller ID: Charles Hayden <679263>
Maximum Chanels: 1
Trunk Name: fwd
secret=<your FWD password>
USER Context: iaxfwd
secret=<your FWD password>
- Open the "DID Routes" page, and select "Ad Route". The following information:
DID Number: 679263
Ring Group: #1
- Open the "Outbound Routing" page and select "Add Route". Enter the following information:
Route Name: outgoingFWD
Dial Patterns: 393|.
Trunk Sequence: IAX2/fwd
After you submit and apply these changes, you can go to the "Maintenance/Asterisk Info" page and check the IAX2 Sip Registry and the IAX2 Peers. These should both show the trunk you set up with StateRegistered and Status OK. You can test FWD by dialing "393612" to hear the time. You can make an incoming call to your Asterisk system using the FWD page that used to sign up for IAX service.
There are many choices for phones that you can use with the Asterisk system. Among the possibilities are IAX phones, SIP phones, SIP softphones, and analog phones with a SIP/analog converter. Such a converter should provide a FXO interface. Sources of phones and converters are listed below.
I purchased some ArtDio
phones, since they were the least expensive SIP phones I could find. Adding new extensions for the phones is straightforward – just like the others. Administering the phone itself was a little more difficult.
The procedure for setting up the phone consisted of two steps.
- Plug the phone in, log on to it, and set its IP address, net mask, and default gateway parameters.
- Browse to the phone and set up its SIP gateway, its name and password, and other SIP parameters. You will know these are right when the phone can register. In the case of the ArtDio phone the critical step was knowing that there were two passwords, and that only the "super user" password ("12345678") would allow access to the key SIP configuration parameters.
- The fields that I had to change on the ArtDio are as follows (assuming the extension is 201 and the IP address of the phone is 192.168.0.60):
Use service: checked
Service type: common
Service addr: 192.168.0.40
Service id: 192.168.0.40
Phone number: 201
Call type: advanced
If you do not see the screen below when you browse to the phone's IP address, but instead see a simpler screen, then it means that you have logged in with the phones user password, not its superuser password.
Once I had a few phones set up, I realized that there are places in my home where I wanted a phone, but where I did not have Ethernet wiring. There are several ways to address this:
- rewire my house, putting Ethernet jacks all over
- use an analog phone adapter and set up cordless phones as extensions
- get WiFi SIP phones, used in conjunction with my existing wireless infrastructure
- use an analog phone adapter and existing phone wiring and phones
The simplest, least expensive, and least disruptive choice was to use an analog phone adapter with a cordless hone. The Sipura adapter is widely available on eBay. It was distributed by various IP telephony service providers. I guess people who tried it and did not like it are selling their used adapters. Just make sure you get one that is "unlocked". Apparently Vonage adapters are set up so that you cannot modify the critical SIP parameters, and are not usable with Asterisk. Other service providers such as BroadVoice, do not lock their devices.
I bought a used Sipura SPA-2002. I have also tried this with a Sipura 1000 and it works about the same. Setting it up was pretty easy.
1. Set up an extension, as before, in the AMP Setup/Extensions screen.
2. Plug it into Ethernet and connect a phone. When you apply power, it gets an IP address from DHCP.
3. You need to get the IP address. Pick up the phone, dial "****" and then dial "110#". It will speak the IP address.
5. If you have purchased a used unit, it would be a good idea to clear out any old settings first. On Sipura devices, you can do a factory reset by:
a. Dial "****" and verify that you hear a voice prompt.
b. Dial "RESET#" (73738#).
c. Dial "1" to confirm.
On the "System" tab, enter the fixed IP address you want to assign the device.
When you are done, submit the changes.
Using the new IP address, browse to the advanced admin screen again. This time, select the "Line1" tab. Enter the proxy information (the IP address of the Asterisk system) and the subscriber information (the extension number and password you set up in step 1).
You should be able to use the phone at this point. Go to AMP's Maintenance/Asterisk Info screen to check that the new extension is registered.
Once you have your system set up and are using it for phone service, you may notice that the sound quality is terrible when you are doing other things on your broadband connection, such as large FTP transfers. This depends a lot on your connection speed. In my case, which I imagine is pretty common these days, I have only 128K bits per second uplink speed. For comparison, with the standard G711 coder, one phone call takes 64K bits, half the bandwidth. Unless you do something, FTP will take enough bandwidth to make telephone calls impossible. What can you do about this ?
Your router may be able to help, by giving traffic coming to and from the Asterisk box priority. I have a Linksys
WRT54G, a widely available combo router, 4 port switch, and wireless access point. It has a way to assign priority to different traffic in various ways. I set it up to give priority to a given switch port, where I have the Asterisk system plugged in.
To set this up, browse to your router, log in, and go to "Applications and Gaming" and then to "QoS". I plugged my Asterisk system into port two, so I set up:
Port 2: Priority High
The AAH people have provided you with a ton of pre-configured features. You can start reading through the main configuration files to see some of the cool things you can do with AAH and your BroadVoice and VoipJet accounts. You might want to set up the BroadVoice trunk to do outgoing calls, using a prefix. It would be nice if it could use BroadVoice for outgoing calls until the 100 minutes are used up, then switch to VoipJet. If you figure out how to do this, let me know.
You can make a backup of your configuration files by backing up the files in the /etc/asterisk directory of your AAH server. The AMP "Backup" command on the left can be used to set up periodic backups.
If you are interested in making international calls, add some more patterns in "Outbound Routing". Commonly used dial patterns are preconfigured, so you only have to select them.
Read through the extensions.conf and see what all is already configured for you to test and work with.
Here are some places to go to find out more about AAH, Asterisk, and the components that make it up.
Asterisk@Home and Asterisk:
AMP (Asterisk Management Portal)
FOP (Flash Operator Panel)
A preliminary version of this set of instructions was produced by Jacob Cazell (http://www.cazz.org/
). It helped immensely in getting my own system set up properly the first time, and inspired this document. A few of the sections here incorporate material that first appeared there.
I will be adding to these instructions from time to time as I experiment with additional features, service providers, and subsequent releases. If you have suggestions, corrections, or comments on these instructions, I welcome your feedback. Please send commands to email@example.com
Appendix 1 – Feature Codes
Schedule wakeup call
Festival test (your extension is XXX)
Last Called ID
Activate Call Waiting (deactivated by default)
Deactivate Call Waiting
Call Forwarding System
Disable Call Forwarding
Call Forward on Busy
Disable Call Forward on Busy
Message Center (does no ask for extension)
Enter Message Center
Playback IVR Recording
Simulate incoming call
Appendix 2 – Asterisk CLI commands
This is a selected summary of the Asterisk CLI commands. These commands can be entered on the Asterisk console, which you access by running:
!<command>: Executes a given shell command
abort halt: Cancel a running halt
add extension: Add new extension into context
add ignorepat: Add new ignore pattern
add indication: Add the given indication to the country
debug channel: Enable debugging on a channel
dont include: Remove a specified include from context
help: Display help list, or specific help on a command
include context: Include context in other context
load: Load a dynamic module by name
logger reload: Reopen log files. Use after rotating the log files.
no debug channel: Disable debugging on a channel
pri debug span: Enables PRI debugging on a span
pri intense debug span: Enables REALLY INTENSE PRI debugging
pri no debug span: Disables PRI debugging on a span
remove extension: Remove a specified extension
remove ignorepat: Remove ignore pattern from context
remove indication: Remove the given indication from the country
save dialplan: Overwrites your current extensions.conf file with an exported version based on the current state of the dialplan. A backup copy of your old extensions.conf is not saved. The initial values of global variables defined in the [globals] category retain their previous initial values; the current values of global variables are not written into the new extensions.conf. Using "save dialplan" will result in losing any comments in your current extensions.conf.
set verbose: Set level of verboseness
show agents: Show status of agents
show applications: Shows registered applications
show application: Describe a specific application
show channel: Display information on a specific channel
show channels: Display information on channels
show codecs: Display information on codecs
show conferences: Show status of conferences
show dialplan: Show dialplan
show image formats: Displays image formats
show indications: Show a list of all country/indications
show locals: Show status of local channels
show manager command: Show manager commands
show manager connect: Show connected manager users
show parkedcalls: Lists parked calls
show queues: Show status of queues
show switches: Show alternative switches
show translation: Display translation matrix
soft hangup: Request a hangup on a given channel
show voicemail users: List defined voicemail boxes
show voicemail zones: List zone message formats
restart gracefully: Restart Asterisk gracefully
restart now: Restart Asterisk immediately
restart when convenient: Restart Asterisk at empty call volume
reload: Reload configuration
stop gracefully: Gracefully shut down Asterisk
stop now: Shut down Asterisk immediately
stop when convenient: Shut down Asterisk at empty call volume
extensions reload?: Reload extensions and only extensions
unload: Unload a dynamic module by name
show modules: List modules and info about them
show uptime: Show uptime information
show version: Display Asterisk version info
show agi: Show AGI commands or specific help
dump agihtml: Dumps a list of agi command in html format
Database handling commands
database del: Removes database key/value
database deltree: Removes database keytree/values
database get: Gets database value
database put: Adds/updates database value
database show: Shows database contents
IAX Channel commands
iax2 debug: Enable IAX debugging
iax2 no debug: Disable IAX debugging
iax2 set jitter: Sets IAX jitter buffer
iax2 show cache: Display IAX cached dialplan
iax2 show channels: Show active IAX channels
iax2 show peers: Show defined IAX peers
iax2 show registry: Show IAX registration status
iax2 show stats: Display IAX statistics
iax2 show users: Show defined IAX users
iax2 trunk debug: Request IAX trunk debug
iax debug: Enable IAX debugging
iax no debug: Disable IAX debugging
iax set jitter: Sets IAX jitter buffer
iax show cache: Display IAX cached dialplan
iax show channels: Show active IAX channels
iax show peers: Show defined IAX peers
iax show registry: Show IAX registration status
iax show stats: Display IAX statistics
iax show users: Show defined IAX users
init keys: Initialize RSA key passcodes
show keys: Displays RSA key information
SIP channel commands
sip debug: Enable SIP debugging
sip no debug: Disable SIP debugging
sip reload: Reload sip.conf
sip show channels: Show active SIP channels
sip show channel: Show detailed SIP channel info
sip show inuse: List all inuse/limit
sip show peers: Show defined SIP peers (clients that register to your Asterisk server)
sip show registry: Show SIP registration status (when Asterisk registers as a client to a SIP Proxy)
sip show users: Show defined SIP users