Showing posts with label phone. Show all posts
Showing posts with label phone. Show all posts

Wednesday, January 5, 2011

Cisco IP Phone Configuration with Asterisk

Getting the Cisco IP Phone 7970 G to work together with the software PBX Asterisk was something I had my hands on a couple of years back. Here’s how you can get them talking together.

You need a couple of things to get this working:

A functioning DHCP serverA functioning TFTP serverSIP Firmware from Cisco This is just a gzipped and tar’ed file.A functioning asterisk serverA Cisco IP Phone

According to a recent installation, the TFTP server must contain the following files

apps70.1-1-2-26.sbn
cnu70.3-1-2-26.sbn
cvm70sip.8-0-2-25.sbn
dsp70.1-1-2-26.sbn
jar70sip.8-0-2-25.sbn
SIP70.8-0-3S.loads
term70.default.loads
term71.default.loads
SEP.cnf.xml

The file you should pay the most attention to is the SEP.cnf.XML file, this is the configuration file. The configuration file is in XML format. You can find a sample configuration here that should work.


true
SIP




M/D/Y
Greenwich Standard Time


true



ccm-beta-5-1
CallManager 5.0 Beta Pub – 5.0.1.032

2000
5060
5061

2427
2428


ccm-beta-5-1





Disable
Disable
false

2000

2000

2000
IP ADDRESS TO SIP SERVER
5060

5060

5060
false

-1
Default
Default
120



USECALLMANAGER
5060
USECALLMANAGER
5060
USECALLMANAGER
5060
true


true
x-cisco-serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
false
2
true
true
2
2
0
true


6
10
180
3600
5
120
120
5
500
4000
70
true
None

1
false
true
false
false
none
101
3
avt
false
false
3

2
false
15000
10
true
16384
32766


9

USECALLMANAGER
5060
3302
3302

2

3

false
3

4
5
7b452e87-4496-4762-e11f-b26751a1884b

true
false
false
true



5060
184
0

SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml



true
2

SIP70.8-0-3S

falsefalse01000001,707:3010:3001:001

1136931633-57191cee-5ffc-4342-b286-4246b4991890

English_United_States
1
en_US
1.0.0.0-1
iso-8859-1

United_States

United_States
64
1.0.0.0-1

1
0
http://ccm-beta-5-1:8080/ccmcip/authenticate.jsp
http://ccm-beta-5-1:8080/ccmcip/xmldirectory.jsp

http://ccm-beta-5-1:8080/ccmcip/GetTelecasterHelpText.jsp


http://10.86.5.102/CiscoServices/index.xml
96
0
96
4
0


3804
ccm-beta-5-1



false

2
2000
2000

true
true

dialplan.xml

On the Asterisk server, you will have a file named sip.conf and to have the Cisco IP Phone talking to Asterisk you need this

[999999999]
username=999999999
type=friend
secret=password
nat=no
host=dynamic
canreinvite=no
dtmfmode=rfc2833
context=incoming
qualify=yes
disallow=all
allow=ulaw

That should be it, good luck!

Tags: asterisk, cisco 7960, cisco 7970, cisco 7970G, cisco config, cisco ip phone, config files, configuration, ip phone

This entry was posted on Wednesday, May 20th, 2009 at 12:20 am and is filed under Cisco configuration, Tutorials. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.

Sunday, December 19, 2010

Cheap international phone calls using Asterisk/FreePBX and local numbers

One of the easiest ways to make cheap international phone calls is to set up a local access number that automatically forwards to an international number using VOIP.

This is a great way to keep in touch with friends and family in other countries (or to let them keep in touch with you!)

It’s very simple to do in FreePBX in two easy steps.

Create a Misc Destination for the number you want to call -

then create an Inbound Route with your local number as the DID Number -

and set the destination to your Misc Destination -

You can then call the local number and be connected to the international number, only paying inexpensive VOIP rates. The local number can be dialled in the normal way, including from a mobile phone.

Give it a go by ringing this London number – 020 3455 4080 – which will connect you to the Naval speaking clock in Washington DC.

If you’re looking for free UK local numbers then try here – http://www.ukddi.com/

Related posts: Cheap international phone calls with AsteriskGetting started with FreePBX – Part 6 Cheap phone calls using DISA and Callbackvoip.ms with Asterisk

Nortel 1535 IP phone with Asterisk and O2 broadband

I recently ordered a Nortel 1535 SIP phone as recommended by Ward Mundy here – http://nerdvittles.com/?p=703


At £35 (price seems to vary up and down) from e-bay they are definitely a bargin. There’s a good thread on the PBX-in-a-Flash forum about setting it up here – http://pbxinaflash.com/forum/showthread.php?t=8273



They come with the default language set to Turkish so that needs changing to English.


I also really struggled with a couple of things setting it up with my O2 router. By default the WiFi would not work and also it would not register with Asterisk.


Wifi not working


I have a O2 Wireless Box II and by default the Wifi security “WPA-PSK Version:” was set to “WPA+WPA2?. The phone would not connect to the WiFi until I set this to just “WPA2? (and entered the passphrase). Even then the WiFi setup on the phone is a bit flakey. Manually assigning the ESSID seems to help too.



Phone would not register with Asterisk


Even more tricky to track down was the fact that the O2 Wireless Box II tries to do something clever with SIP connections to port 5060. I had to telnet to the router (default address – 192.168.1.254) and remove the SIP settings.


To do this -


telnet 192.168.1.254
User = SuperUser password = O2Br0ad64nd
:connection unbind application=SIP port=5060
:saveall


and then reboot the router. This information was taken from here – http://forum.o2.co.uk/viewtopic.php?t=1313


After that the phone would register following the setup directions in the PBX-in-a-Flash forum thread mentioned above.

Friday, December 17, 2010

Buying the right Polycom phone

If you are ready to buy a VoIP phone and have done the research and decided on Polycom, congratulations!  Now you just need to narrow down your selection from the 9 Polycom phone models offered and the 2 variations of each model, so here we go!

Knowing what kind of call volume your business has or will have is imperative to deciding which Polycom phone to install.  Sure, any Polycom phone will perform exceptionally well even if the user is on the phone 24/7, but it won’t necessarily help the user be more effective or satisfy your customers who are calling in.

The IP321, 331, and 335 have 2 SIP lines, 3 soft keys, and a 2.5mm headset port (IP335 has an RJ9 headset port).  If your user will rarely have more than 1 call coming in at a time and will have little need to transfer, this phone will work just fine.  Yes the IP321/331 and IP335 are capable of call transfer, but there is only one hard-key for busy lamp field (BLF) so multiple “favorite extensions” is not possible.  Also with only 3 soft-keys and the basic hold, headset, speaker, menu, and dial hard key, the use of special features like conference, transfer, and redial are a bit more cumbersome.  So the Polycom SoundPoint IP321, IP331, and IP335 are recommended for a desktop where there is very low call volume, or common areas like break rooms, kitchens, copy rooms, and lounges.

Polycom SoundPoint IP335 Polycom SoundPoint IP335

For most deskbound positions with medium to high traffic volume (think 12+ calls a day) the IP450, IP550, or IP560 is a sound, budget conscious choice.  The Polycom SoundPoint IP450 and IP550/IP560 have 3 and 4 SIP lines, respectively, offer HD Voice wideband sound on both the speakerphone and handset, which is awesome, have RJ9 headset ports, and have a large backlit graphical LCD display.  Four context sensitive soft-keys give the user more options at their fingertips.  For those requiring more complex phone functionality like transfer, conference, company directory, and Do Not Disturb, the IP550 or IP560 is recommended over the IP450. The IP550/IP560 has 4 context sensitive soft-keys and a Mute, message, redial, speaker, handset, and hold, like the IP450, but it also has dedicated hard-keys for Directories, Applications, Conference, Transfer, and Do Not Disturb.  I have a Polycom IP550 at my desk and I love it lots. It offers 4 SIP lines, 2 of which I have assigned to my extension and the other two I use as a BLF for my 2 most dialed extensions.

Phone users who would be considered “super users” would most definitely need an IP650 or IP670. These super users, such as receptionists, operators, and call center managers need more than a few Busy Lamp Fields for transferring to multiple extensions and monitoring others’ phone status. The IP650 and IP670 offer 6 SIP lines that can be used as BLF extensions right out of the box, however, you can add up to 3 sidecar modules to these bad boys to give you an additional 12 BLF extensions each.  The Polycom sidecars are nicer than most because instead of using paper inserts, the labels are an LCD screen (monochrome for the IP650 and color for the IP670).  This offers flexibility for the administrator to change extensions quickly and for the user to see the changes in real time without having to print out and insert a new paper label.

Consider carefully where you will be placing your Polycom SoundPoint IP phones before you decide on a model.  First question you should ask yourself is will these phones be attached to a PoE (Power over Ethernet) switch?  If so, all Polycom SoundPoint IP Phones are PoE, so you are cool there, but don’t spend more than you need to.  Most Polycom SoundPoint models come in two variations, with (part number ends in -001) and without an AC Power adapter (ends in -025). If your network is already PoE, make sure you buy a Polycom model that ends in -025. Why pay for a power adapter you don’t need?

Secondly, will the desktop or location you are installing the phone have 1 or 2 RJ45 jacks?  If there is only 1 and there will be a PC there as well, you need to choose a Polycom phone that has a dual RJ45 switch integrated into it. This will allow you to plug the phone into the single wall jack, and then the PC into the phone, passing connectivity through to both appliances without having to install expensive additional wiring.  Polycom IP Phones that offer the dual RJ45 jacks are the IP331, IP335, IP450, IP550, IP560, IP650 and IP670.  So if you only have one network jack and need a phone and PC there, stay away from the IP321.

Another consideration here is speed. If you are using your Polycom IP Phone to pass data through to your PC (dual RJ45 jacks) you should consider the speed of your server switch.   If you have a 10/100/1000 network switch and want to offer gigabit speed to your PC, you need a Polycom phone not only with dual RJ45 ports, but with dual 10/100/1000 RJ45 ports.  That narrows down your selection quite a bit more.  If you are looking for a gigabit Polycom phone, your choices are the IP560 or IP670.

The Polycom IP650 and IP670 are virtually the same phone with 2 major differences. First, the IP670 has gigabit support (boring).  But even cooler is that the IP670 and its sidecar support a full color LCD, so you can choose color schemes or set pictures as your background. In addition, the IP670 has a USB port for local call recording. That’s pretty cool, and necessary if you are in the legal or financial industry where you may need record of phone calls on the fly.

Polycom IP670 Polycom IP670

You should also consider Polycom’s HD Voice.  Polycom HD Voice equipped phones not only support the G.722 wideband codec, but they also build up their hardware in the handset and speakerphone to enhance the HD voice sound.  It is really better than actually being there.  Today, HD Voice is really a cool factor, the sound of it will just amaze you, but you only really experience the rich sounds when both you and the other party are on HD Voice phones, the call is completely IP, and all PBX’s involved support HD Voice.  For today that may only involve internal calls and outbound calls with a few clients. However buying a Polycom phone that is equipped with HD Voice will be future proofing your infrastructure for the next 5 years. More and more phones are coming standard with HD voice support, and as the world continues to migrate to IP PBX’s, most already support HD voice and if they don’t, it is in their upcoming upgrades.

Because you asked, and even if you didn’t I am going to tell you anyway, I recommend the Polycom IP450 for most office applications.  I am a moderate phone user and I have the IP550. The extra hard keys for Directory, Applications, Conference, Transfer, and redial are nice, but I don’t really use them.  My IT guy has provisioned some of these options into the context sensitive soft-keys company-wide for some of the users who don’t have the IP550 and I use those for transfer and redial.  Yeah, the IP550 is pretty cool, but if you are on a limited budget, the IP450 should still do you well.  Of course, for high volume users like secretaries and receptionists, you would need the IP650 and the IP650 sidecar.  IP670 is a must in legal or financial scenarios where local call recording is required.

Thursday, November 18, 2010

New Grandstream GXV 3140 IP Video Phone is Skype Certified

 
New Grandstream GXV 3140 IP Video Phone is Skype Certified Grandstream Networks, manufacturer of IP voice/video telephony and video surveillance solutions, announced that its award-winning GXV3140 IP multimedia phone.    The GVX 3140 uses mega-pixel camera and H.264 codec to deliver razor sharp video calling over Internet, is officially certified by Skype and incorporates native Skype embedded software to allow Skype video calling without a PC. Skype’s community of users worldwide, who already have the capability to send and receive free video calls to and from their computers, can now do so with Grandstream’s IP multimedia phone. Designed to strengthen relationships with business colleagues and customers, as well as keep in touch with family and friends, the GXV3140 offers FREE two-way or three-way video calling out-of-the-box along with real-time instant access to a variety of Web and popular social multimedia applications.

“Skype certification on the GXV3140 IP multimedia phone creates a seamless video-enabled communications experience from PC to desk phone, giving Skype users the utmost flexibility and convenience for video calling,” said David Li, CEO of Grandstream Networks. “Grandstream's GXV3140 bridges the gap between convenient-to-use dedicated voice/video end-points and the increasingly popular world of PC-based Internet multimedia communication offerings like Skype."

The Skype-enabled Grandstream GXV3140 is the industry’s first desktop IP multimedia phones compatible with both Skype and SIP open standards for voice & video calling. No service contract or additional monthly fee is required to use Skype for video calling.

With a 4.3” digital color LCD, the ‘always-on’ GXV3140 has an integrated Web browser, giving users convenient, one-touch access to personalized feeds of real-time online information services (such as news video or RSS feeds, weather, IM, stock/currency updates, etc), thousands of Internet radio stations, major Web photo sites (Yahoo Flickr, Photobucket, Phanfare), popular social networking sites (Facebook, Twitter) and Internet streaming entertainment sites like YouTube and Last.fm.

Earlier this year, Grandstream also announced successful Skype Connect™ certification for its IPPBX and gateways. Skype Connect interoperability allows business users to receive calls through their Grandstream IPPBX from Skype users around the world. Companies can now allow their customers and remote employees to make free inbound calls to their corporate PBX system using Skype from anywhere in the world.

“We are pleased to deliver this new video device category that works with Skype and are honored to be selected for our innovation, rich features, and affordability. The GXV3140 phone joins Grandstream’s other Skype-enabled solutions including our IPPBX and gateway products previously certified by Skype,” Li added. “We value our strategic relationship with Skype and look forward to bringing more devices with Skype integration to market in the future.”

Pricing and Availability

The GXV3140 is available through Skype Store and Grandstream’s worldwide distribution partners for $229.00.

Friday, November 12, 2010

Windows VOIP Business Phone System


Windows based, ISDN and VOIP Ready. Suitable for companies with 1 to 200 employees.
Supports Traditional, USB, SIP Phones, Wireless Headsets, Wifi Phones and Softphones.

The features are comprehensive and includes:

VOIP Ready – Internet VoIP calls over SIP Carriers. ISDN Ready – Connect in existing ISDN lines. VOICEMAIL – by phone, or by email for every extension and service groups. AUTOMATED CALL ATTENDANTS (IVR) Voice/Music Greetings to Callers providing Optional Menus by Touch Tone Dialling. PRESENCE indicators for every extension showing extension on-call, free, away or offline. Outlook Integration – pop up relevant contact details and notes for incoming caller. Call Recording – Automatic or manual instant recording. WAN – Integrate branch offices and benefit from free inter office calls, and improve office communication efficiency. Centralised Phone Book – company wide telephone book showing names, numbers and business information. Fax Server – company wide fax server allowing employees to fax straight from desktop PCs, receive incoming faxes, and notifications of successful/unsuccessful messages. Call Pickup/Conferencing/Divert/hold/Forward Music on Hold – configurable, accepts any .wav file Short Text Messaging – short message any extension on same system or in another branch office. SIP Phones, USB Phones, Softphones and Traditional RJ11 Phones – wide ranging support of hard phones and softphones. Least Cost Routing – automatic routing with calls between VOIP and ISDN carriers to minimize outages and call charges. Day/Night/Holiday Modes – system programmable to offer different messages or call routes depending on day, time and holidays.

Tuesday, November 2, 2010

Cisco 30 Millionth VoIP Phone is deployed by HSBC bank

Cisco celebrated a major milestone with the distribution of its 30 millionth VoIP phone to one of the world's most prominent banking and financial services enterprise HSBC. As a part of a standardization concept to implement the uniform global standard, HSBC has deployed over 100,000 Cisco Unified VoIP Phones throughout 87 different locations around the globe.

HSBC is operating with Cisco's Unified VoIP phones and Cisco Unified Communications Manager as they have figured out that standardization, unification and simplification massively improves overall performance, increases organizational versatility and cuts the expenses on supporting infrastructure. Additionally, savvy call routing, a single directory and template for a dial-plan provide much greater employee interoperability in order to develop customers’ satisfaction strategies. By implementing device identity, digital signed images, call tracing, and call recording features, Cisco's IP phones provide HSBC with highest level security to contain conversations of a intimate nature secured, as well as providing regular support for various device functions and applications directly on the handsets.
Having begun the installation and deployment of the Cisco Unified IP Phones over 18 months ago, HSBC is almost halfway on the three-year global program that will be finished by installed base of 354,000 Cisco devices by the end of 2012. With investments in communications and collaboration technologies such as Cisco TelePresence, Cisco Unified IP phones and Cisco WebEx, HSBC is targeting to considerably lower its carbon dioxide emissions by reduced business travel in 2012. Cisco made the first Voice over IP phone call in 1997 and sold its first Cisco Unified IP Phone in 1998. In the last 13 years, Cisco Unified IP Phones have evolved beyond voice to connect people with video technology, and act as interfaces to networked applications, services and content. They are part of a new collaboration experience that is rich, integrated, mobile, and often virtualized.

Sunday, September 26, 2010

The phone system that lets you change your mind


 


Switchvox resellers already know how powerful the product’s features are, so I thought I would stress how powerful the sales process can be from both the reseller and corporate customer perspectives.


Here’s one of the messages you should understand about Switchvox:


“You can always change your mind.”


A phone system upgrade or replacement can seem overwhelming to small businesses. That’s why SMBs are justifiably worried they may be buying the wrong solution, they may need a feature and forget to mention it, etc.  Of course price is a major item, but so is fear of the unknown. Remember, it’s a significant expenditure and organizations want to be sure they are getting the right product. When considering a Switchvox purchase, customers need not worry because if their business changes or they need something new, they can always change their mind with Switchvox.


Here’s something we often discuss with customers.  During the needs discovery for automatic call distributor (ACD), call recording, find me follow me, etc., prospects will typically just throw out some of these options as they don’t need them today. However, it’s important to know that “You can always change your mind” when you go with Switchvox.  This helps dispel some of the fear about buying the wrong thing.


The analogy we use is some people have a small car when they get married and that works great until they have a kid, then they get a dog, bikes, etc.  Now that small, fun, two-door car no longer fits their needs. It was great when they got it, but now things are different and they need something like a minivan.  Well, it’s time to go trade in the small car and get the minivan to suit the new family needs. This means getting rid of one piece of hardware to go through the buying process again hoping they get it right this time.


It’s the same with any organization — needs change as a company will invariably decide to alter its business, hire more people, acquire a competitor, etc.  What was not important when initially buying a phone system — like the awesome and free Switchvox Mobile smartphone applications — is now critical. For example, imagine a top-performing new salesperson who is accustomed to having access to all the great features of her office phone system right on her smartphone, which she can’t live without it. A company would realize that her value goes up when she has the correct tools to make her productive. The organization would want to be able to make that addition on the fly.


With other systems, prospects may need to buy an add-on for more cost, or worse, it may not even be available on the product they bought. With Switchvox, they can add call recording, ACD, etc. at any time.  This way, organizations “change their mind” and Switchvox allows them to grow and accommodate as needs increase and change.


Switchvox has a multitude of features built in and can adapt to future needs as it is based on open standards. With this in mind, prospects can rest assured that as they change and grow, they can change their minds as to the types of features required to meet the needs of their organizations. They’ll have a great platform that changes with them and a partner delivering the best VoIP phone system that exists.


 

How to Lower Monthly Phone Expenses

I’ve been in the IT industry a long time, and I’ve seen lots of changes. When the economy was going strong, many clients wanted the latest and greatest – and they didn’t mind paying extra for every little bell and whistle. This worked well for a lot of VARs for a lot of years.


When the economy turned sour, companies that had made their mark by making fast, product-based (rather than service-based) sales had a much harder time staying in business. In fact, 40,000 of the 250,000 VARS in the United States went out of business in 2009.


One of the reasons that my business continues to flourish is because I always knew that I should focus on cost savings and quick ROI for my clients. Everyone loves the bells and whistles – but can’t we have it all? (The answer is “yes”).


When it comes to phone systems, Digium Switchvox is a great solution for helping customers save on their telecom expenses while improving employee productivity. Switchvox is the right answer for any business that wants amazing features, rapid and effective return-on-investment, and a great price.


Case in point:


One of my clients, a local Dallas Fort Worth-based CFO outsourcing company with about 40 employees, wanted to lower its telecom expenses and improve employee productivity.


At the time, the company utilized Covad’s Hosted VOIP service and 3Mb/s of Internet service. However, their monthly telecom expenses topped $3,500 per month – and we knew we could help them realize savings and even better functionality.


My company, The Fulcrum Group, proposed a completely managed VOIP solution that would mirror their current Covad Hosted VOIP service. We proposed that the company utilize:

XO 3Mb/s IP Flex Voice/Internet service (2 T-1s, up to 32 SIP trunks, dynamically allocated between voice and Internet), 3 year contract;Digium Switchvox SMB AA355 appliance with 50 Silver subscriptions (we re-used their existing Cisco switches and phones), 3 year lease; andThe Fulcrum Group’s SPOT Managed VOIP Services offering, a fixed price, “all you can eat” support offering that covers all of the maintenance, support, moves/adds/changes, backups, and end training for the Switchvox VOIP system.

Bottom line:  We were able to lower the customer’s monthly telecom expenses by approximately $2,000 per month, and they now have a more flexible, higher functionality phone system that is improving employee productivity.


The takeaway?  Digium Switchvox is a great product to be selling in these tough economic times. But it’s going to continue to be a product that will make companies like ours look great – no matter what the economy is doing.

Monday, August 30, 2010

Snom releases the 821 IP Phone that supports gigabit ethernet.

Snom technology AG, a leading developer and manufacturer of advanced voice over IP phones for enterprise and residential markets, announced today the launch of the snom 821, a new desktop Internet protocol (IP) phone with all the advanced functionality and interoperability of other snom 8xx series phones, as well as a new integrated gigabit switch.  In addition to the technological advancements to the phone, the snom 821 also comes in both light and dark colors, just like its big brother the snom 870. The snom 821 is expected to be fully available in the U.S. by the end of the second quarter.
"As desktop phones continue to migrate from being voice-centric devices to acting as complete dashboards for unified communications, the speed of application execution is becoming more and more critical," said Dr. Michael Knieling, Executive Vice President of Marketing and Sales for snom technology AG. "The addition of a gigabit switch came from customer requests, and we are very pleased that we were able to add innovation while keeping the same price point."

The snom 821 offers a large high-resolution TFT color display for brilliant depiction of call lists, phone directories, and caller information via the integrated XML browser. In addition, the phone's wideband technology captures more than double the frequency spectrum that a standard handset does, providing crystal clear sound quality.

As a business communications endpoint, the snom 821 features snom's industry-renowned firmware, allowing for robust security supported through secure integrated open VPN, TLS and SRTP, and high sound quality using the handset or the speaker function. The snom 821 also supports all of the leading SIP-based IP PBX and unified communications environments, including Microsoft Office Communications Server 2007 R2.

As with all of snom's desktop phones, the snom 821 also offers the necessary functionality for any business user, including features such as 5-way conferencing, multiple ringtones, and up to twelve different SIP identities.

The snom 821 and 870, as well as associated accessories such as the PA1 public address system, the MeetingPoint conference module, and the snom Vision sidecar, are designed to take the desktop business phone beyond its traditional position as a person-to-person voice communications device and make it a truly interactive and intelligent business communication device. Armed with full-color and touchscreen interfaces, web programmable ports, and a wide array of interoperable unified communications elements, the entire suite of next generation snom devices represents the future of desktop interaction.

 Source:  Snom Technologies AG

Thursday, January 7, 2010

Asterisk PBX Hardware

The Components

Your 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.
Phones

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.
SIP Gateway

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.
Home Network

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.