1. Of course, traditional way calling internationally is using calling card.
I went to Chinatown to buy calling cards when I first came here. I bought it online. We can get calling card with rates about or less than 1 cent per minute calling China.

2. Make international calls with local calling rates (or free): Talkster.com

http://www.talkster.com/

No registration, no credit cards, no catches, no limits!

With talkster, you get a local access for yourself, and an access number for your friend.
Say you want to call your friend, you call your access number which rings your friend’s phone. He/She answers the phone, then hangs up. You stays online. Your friend calls his/her access number. Then you  two will be connected, basically having a conference call.

The good thing about this method is that your friends, families in China call also you by calling their access numbers first. You answer the phone, hang up, ask them to stay online, and then calls your access number.

3. Skype Unlimited World $9.95/month

http://www.skype.com/allfeatures/subscr … nadaworld/

In particular, unlimited calls areas include:
China landlines and mobiles
USA landlines and mobiles

Make Unlimited Skype™ Calls without a Computer with Belkin’s New Desktop Internet Phone for Skype: $99

Of course you don’t need pay anything if both sides use skype.
Get a skypein phone number, you basically have a lanline phone. It costs €50 one year.

4. There’s another way which is a little bit complicated. But it’s free calling between two sides. You need pay to get sip devices like this: Linksys PAP2T-NA for both sides and connect regular phones to the devices.

Then get sip accounts for both sides, like voipuser.com, voxilla.com or freedigits.com etc. Set it on the Sip device.
Reference:

http://blog.nyworldphone.com/2008/02/06 … cked-pap2/

.

In particular, Freedigits’ account is a US phone number. Basically, you and your friend in China get a free US phone number. Calling between you is free! Of course, other people in US can call you.

If you want to get local numbers, go to grandcentral.com, get numbers and forward it to your freedigits number(or GizmoProject sip account).

If you want to call other numbers in US or China, register accounts and buy credit from your favorite VOIP companies like voipdiscount.com etc, and register it on the sip device.

Basically, you get a home phone with a free US phone number, free calls between you and other people with the same setup and low rate calling to other people.

5. Another convenient way to call China is use Packet8 Mobile Talk with cell phones for 1 cent per minute to China.

6. I guess a lot of poople know iTalkBB. They offer unlimited calls to north America and China with a monthly fee $25.

7. The complicated way is to set up a free PBX asterisk which will provide the most features that you want. You can get a WRT54GS, load OpenWRT firmware and install asterisk. Then you can set up free US phone numbers (or even China number which you usually need pay) to receive calls. And you can set up calling services to call China and US.

Domestic Calling

1. Skype Unlimited US & Canada $2.95/month

Usually, you need a computer to make skype calls. Get a phone above, you’ll make calls like regular.
Make Unlimited Skype™ Calls without a Computer with Belkin’s New Desktop Internet Phone for Skype: $99

2. People who have tmobile service, even the lowest package, want to have a cheap home phone, Tmobile-at-Home is good choice.
$10 a month for unlimited calls in north America:

http://www.t-mobileathome.com/

3. Same as international calling #4 which may be the cheapest way. You can get Linksys PAP2-NA for about $30 on ebay.
4. Same as international calling #7.

If you got an Aim Phonline free DID, set up a secure SIP device password in their Dashboard at https://dashboard.voice.aol.com/settings/sip.

In your application like xlite or sip devices, set as follows.

Domain = sip.aol.com
Username = screenname@aim.com
Password = “password set under the AIM Call Out dashboard SIP Clients settings page”
Caller ID name = screenname

If you need stun server, use turn.oscar.aol.com .

For Asterisk, register as follows.

register => MyScreenName@aim.com:SIP-Password@sip.aol.com

For more details, check here: http://dev.aol.com/api/aimcall.

Client list: voice.aol.com/SIP-Client-List.

It shows that it has some unlimited calls, like to China.

Link: http://www.therichcom.com/carddetails.asp?Brand=2562

Link: http://www.tigerphonecard.com/product/sti_nj_(800_and_nj_local_access)phone_card.html

According to the description,

ABSOLUTELY NO REFUND OR EXCHANGE !!WHEN USING A TOLL FREE NUMBER FROM A PAY PHONE A $0.90 PER CALL SURCHARGE WILL APPLY. A WEEKLY MAINTENANCE FEE OF $0.79 WILL BE ASSESSED AFTER THE FIRST CALL.

If only $0.79 is charged per week for unlimited calling, it’s still a good deal. I wonder how reliable it is.

With Packet8 MobileTalk, people can make international calls directly through cell phone via Packet8’s IP network. The rate is not bad for some countries, like 1 cent for China. It’s a good way to make calls. And no wifi/data plan required. But people do need install their application on their phones. Here is a list of Supported Mobile Phones.

I wonder what the quality is. If anybody has tried it, you’re welcome to comment.

Jan 1 00:44:48 NOTICE[1254]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?!
Jan 1 00:44:48 WARNING[1254]: res_musiconhold.c:336 spawn_mp3: /var/lib/asterisk/mohmp3 is not a valid directory
Jan 1 00:44:48 WARNING[1254]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player

Source link at voip-info.org.

Shoutcast Music On Hold

You can have asterisk use a streaming source for on-hold music.

Make a directory and put a 0 size file ending in .mp3.
I called my directory: /var/lib/asterisk/mohmp3-empty

in musiconhold.conf, add a line such as:
default => mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/

Alternative method –

The initial method seems to not work in 1.2 & +. You can use the method below, doing a custom application and calling mpg123, and specifying your shoutcast source.

default
mode=custom
dir=/var/lib/asterisk/mohmp3-empty
application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -s –mono http://213.186.41.160:8000/

In June of last year, T-Mobile introduced their Hotspot@Home UMA service, which lets users “seamlessly” (depending who you ask) transfer calls between their cellular network and Wi-Fi. It cost $19.99 per month for a single line. Now it’s $10.

Tmobile@Home

T-mobile just launched “Tmobile@Home” plan, which provides users with a $50 WRTU54G, allowing users to plug in a landline in order to get unlimited local and domestic calling for just $10 a month. T-Mobile will transfer your number to their system, and they’re also offering a $60 VTech wireless phone. The service was initially only offered in Dallas and Seattle, but as of July 2 will be available nationwide.

T-Mobile CEO talks with the Seattle Times about the company’s shift in direction.

Free areas: http://www.talkster.com/Countries.aspx 

Call with regular phone: Just get your free Talkster numbers on the homepage: http://www.talkster.com/Default.aspx.

Call with skype: http://www.talkster.com/Skype/default.htm.

Call with googletalk: http://www.gtalk2voip.com/talkster.shtml.

Free Calls and Conference Calls with all your facebook friends: http://apps.facebook.com/talkster/

It works like a conference call. You call your local number, the other person calls his/her local number. Both are charged local fees if the phone company charges.

To make a call, call the talkster number, and tell your friend to handup and call in his local talkster number. In the meantime, you wait online until your friend calls in.

A new york based company called pccall.com offers a plan called “Lucky 7″, with which people could calls 7 countries with monthly fee $7.

The 7 countries are

As explained here: https://www.pccall.com/lucky7.asp, “monthly Fee of $7 for your first full calendar month”, which sounds tricky.

It seems people have to call from computers. There’s no sip proxy introduced so that people could call from sip device or asterisk.

Check out with the company if you are interested. Welcome to comment on it if you get any experience.

I’m using WRT54GS + Openwrt + asterisk at home. It is good because we have all controls. But admin password will be lost from time to time, then I’ll have to reset it: http://blog.nyworldphone.com/2008/02/06/openwrt-on-wrt54gs-reset-the-root-password/. And, for some providers, I can’t make it work with my asterisk on wrt54gs.

Then I tried pbxes.org. It is still free and works perfect. I can do everything I’m doing right now with my asterisk.  It is a good option for personal use which cost nothing. Here I would like to explain some configurations that I did.

1. Receiving calls on Sip phone/device.

A) Internal between extensions.

• Extensions-> Add Extension
• Display name(e.g. 3000)
• Password(e.g. 3000)
• Outbound CID(e.g. 3000)
• Voicemail(optional, disabled by default)
• Submit

You could do the same thing above to add more extensions (e.g. 6000 etc).

When you configure it on your sip phone/device, usename is “pbxes_username”-exten, e.g. username-3000, password is your pbxes extension password and domain/sip proxy is pbxes.org. Now you can dial extensions from one to another, e.g. dial 6000.

B) Through a DID number

a. Add extensions as A).

b. Add trunks

• Trunks->Add Trunk->(e.g. Add SIP Trunk)
• Trunk Name (e.g. freedigits)
• username (e.g. freedigits number)
• password (e.g. freedigits password)
• Sip server(e.g. freedigits.net)
• Submit changes

 c. Add Inbound Routing

• Inbound Routing->Add Incoming Routing
• Trunk: (e.g. freedigits)
• Caller ID Number (optional, specify allowed incoming CID)
• Set Destination: mark Extension, and choose one from the list, e.g. 3000.
• Submit

Now when people call the freedigits DID, calls will be routed to the extension 3000.

2. Make calls

A) Add extensions as 1.A.

B) Add trunk as 1.B.b with your favorite voip provider, e.g. voipbuster.

C) Set outbound routing

• Outbound Routing->Add Route
• Route name: (e.g. sip out)
• Trunk sequence: (e.g. voipbuster)
• Set Destination (optional)
• Save changes

Now you can make calls usign your favorite vsp from your sip phone/device.

3. Call Forwarding/Receiving calls on PSTN phones (landline or cell phones).

A) Add trunks as 1.B.b.

B) Add ring groups

• Ring Groups->Add Ring Group
• Group number: (e.g. 1)
• extension list: landline or cell phone number plus #. e.g. 12126668888# (it may differ according to your provider)
• ring time, e.g. 20
• Destination if no answer (optional)
• Submit changes

C) Add inbound routing

Same as 1.B.c, except in “Set Destination” mark Ring Group(e.g. #1).

D) Add outbound routing as 2.C.

Now if people call your freedigits DID, the call will be forwarded to your regular phone using through your vsp.

4. Call back

Start with callthru by adding an inbound route with the number of your phone entered as Caller ID. Leave Trunk Name empty. If it doesn’t work check your call monitor for the right format of your Caller ID. When dialing the callthru destination you may always press the * key to redial and # to end digits. After the called party hangs up you will hear another dialtone for your next call. To hangup by yourself transfer the call to an invalid destination by dialing *2.

- alen