New York World Phone

Intertional Calling

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Everybody knows using computers to talk with MSN, QQ etc. We’ll nottalk about those methods here. Instead we talk about calls with regularphones on one side or both sides.

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

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Usually, people get phone service from Verizon. They offer individual service and packages like including internet, TV and phone services. Time Warner Cable also offers similar services.

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.

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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.

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I didn’t install voicemail, sounds etc. I have no mp3 files on my asterisk.

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/

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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

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Asterisk on Linksys PAP2v2 and DLink VTA-VR

Broadband reports Link by mazilo.

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Asterisk on WRT54G

openwrt Link

asterisk:

http://zandbelt.dyndns.org/asterisk.html

http://members.home.nl/hans.zandbelt/openwrt/kamikaze/packages/asterisk-1.4/

http://downloads.openwrt.org/whiterussian/packages/

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Asterisk on OpenWRT

Link1

Link2

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Install Asterisk - download the source code and compile it

Link

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Downloading and Compiling from Digium

Link

My experience in installing Asterisk on WRT54GS

http://blog.nyworldphone.com/2008/02/06/my-experience-in-installing-asterisk-on-wrt54gs/

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voip-info Link

openwrt Link

 I followed the post on Asterisk on OpenWRT: Asterisk on OpenWRT part 2. Now, I report my process and problems here.

vi /etc/ipkg.conf (see updated info)

src whiterussian http://downloads.openwrt.org/whiterussian/packages
src acvs http://12.176.248.4/ipkg
dest root /
dest ram /tmp

ipkg update

ipkg install asterisk-cvs

vi /etc/asterisk/modules.conf (slim modules)

[modules]
autoload=yes
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => pbx_dundi.so
noload => app_intercom.so
; load => chan_modem.so
noload => res_musiconhold.so
noload => chan_modem.so
noload => cdr_pgsql.so
noload => cdr_mysql.so
noload => chan_alsa.so
;noload => chan_oss.so
[global]
; chan_modem.so=yes

Set asterisk startup (see update info)

Please see below problems I got in the installationi process.

1. When I run ipkg update, I got the following errors:

root@OpenWrt:/etc# ipkg update
Downloading http://openwrt.org/ipkg/Packages
wget: openwrt.org: Unknown host
Downloading http://12.176.248.4/ipkg/Packages
wget: Unable to connect to remote host (12.176.248.4): Network is unreachable
Downloading http://nthill.free.fr/openwrt/ipkg/stable/Packages
wget: nthill.free.fr: Unknown host
Downloading http://nthill.free.fr/openwrt/ipkg/testing/Packages
wget: nthill.free.fr: Unknown host
Downloading http://www.wildcatwireless.net/wrt54g/Packages
wget: www.wildcatwireless.net: Unknown host
An error ocurred, return value: 5.
Collected errors:
ipkg_download: ERROR: Command failed with return value 1: `wget –passive-ftp    -q -P /tmp/ipkg-mWFahm http://openwrt.org/ipkg/Packages’
ipkg_download: ERROR: Command failed with return value 1: `wget –passive-ftp    -q -P /tmp/ipkg-mWFahm http://12.176.248.4/ipkg/Packages’
ipkg_download: ERROR: Command failed with return value 1: `wget –passive-ftp    -q -P /tmp/ipkg-mWFahm http://nthill.free.fr/openwrt/ipkg/stable/Packages’
ipkg_download: ERROR: Command failed with return value 1: `wget –passive-ftp    -q -P /tmp/ipkg-mWFahm http://nthill.free.fr/openwrt/ipkg/testing/Packages’
ipkg_download: ERROR: Command failed with return value 1: `wget –passive-ftp    -q -P /tmp/ipkg-mWFahm http://www.wildcatwireless.net/wrt54g/Packages’
 

Run route -n, I saw that gateway was 0.0.0.0.  I’m actually using it behind my dlink router which has LAN ip 192.168.0.1. So I changed WRT ip to 192.168.0.100 with gateway 192.168.0.1. My computer is connected to one LAN port on WRT. Another cable connect another LAN port on WRT to dlink LAN port.

You could do it manually using the following commands or in the GUI:

nvram set lan_ipaddr=192.168.0.100
nvram set lan_gateway=192.168.0.1
nvram set lan_dns=192.168.0.1
where 192.168.0.1 is your gateway address
if you use WAN connection, use the same command with  wan_gateway and wan_dns

Then when I run “ipkg install asterisk-cvs” to install asterisk, I got an error “src openwrt http://openwrt.org/ipkg is unreachable”. But it seems asterisk was installed properly.

2. When I run asterisk, I got the same problem as in the comment of the post:

asterisk: can’t load library ‘libgcc_s.so.1′

and

asterisk: can’t load library ‘libssl.so.0.9.7′

ln -s /usr/lib/libssl.so.0.9.8 /usr/lib/libssl.so.0.9.7

and

ln -s /lib/libc.so.0 /lib/libgcc_s.so.1

This solved the problems.

3. When I try to register user 3000 on asterisk server, I got an error message: Username/auth name mismatch.

In sip.conf, host = dynamic instead of host = xx.xx.xx.xx solved the problem.

4. Unable to connect to remote asterisk, can’t access to CLI with asterisk -r

Receiving the following error message:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

/var/run/asterisk.ctl does exist and asterisk is running.

 The problem was:

 then add these lines to it or just uncomment them if already exist.

 [files]
astctlpermissions = 0660
astctlowner = asterisk
astctlgroup = pbx
astctl = asterisk.ctl

5. For the voicestick registering problem, I followed the post: http://www.asteriskextras.com/index.php?option=com_content&task=view&id=13&Itemid=2I will copy the post later.Comment by alen — February 4, 2007 @ 1:38 am | Edit This

Problems continued:

6. pap2 couldn’t register
nat = yes
problem solved.
5. 503: service unavailabe
I have sip.conf:[3000]
type = friend
context = default
username = 3000
host = dynamic
mailbox = 3000
dtmfmode = rfc2833
nat = yes

 

and

[3001]
type = friend
context = default
username = 3001
host = dynamic
mailbox = 3001
dtmfmode = rfc2833
nat = yes

extension.conf:

[internal]
exten => 3000,1,Dial(SIP/3000,30,Ttm)
exten => 3000,2,Playback(invalid)
exten => 3000,3,Hangup
exten => 3001,1,Dial(SIP/3001,30,Ttm)
exten => 3001,2,Playback(invalid)
exten => 3001,3,Hangup

Result:
3000 is fine.
3001, I got 503: service unavailable

Don’t know why. 

Update:

http://zandbelt.dyndns.org/asterisk.html

asterisk for whiterussian has been taken down. Asterisk for OpenWRT is now obsolete.

Kamikaze is now the official OpenWRT stable version. But it’m not sure if Kamikaze has a version for wrt54gs v1.0 v2.0 2.1.

In the links in http://lestblood.imagodirt.net/archives/106-Asterisk-on-OpenWRT-part-2.html, only http://12.176.248.4/ipkg is still available.

I put the following in the /etc/ipkg.conf:

src libncurses http://download2.berlios.de/pub/xwrt/packages
src acvs http://12.176.248.4/ipkg
dest root /
dest ram /tmp

It seems it works. Or use the openwrt download links.

src whiterussian http://downloads.openwrt.org/whiterussian/packages
src non-free http://downloads.openwrt.org/whiterussian/packages/non-free
src acvs http://12.176.248.4/ipkg
dest root /
dest ram /tmp

Start Asterisk automatically at startup

Add a simulink /etc/init.d/S60asterisk to asterisk

ln -s asterisk S60asterisk 

Then  put the following content in /etc/default/asterisk.

vi /etc/default/asterisk:

## startup options for /etc/init.d/asterisk

ENABLE_ASTERISK=”yes”
OPTIONS=”"

I checked later. It seems that if I make a simulink S99asterisk, it’ll work without vi /etc/default/asterisk: ln -s asterisk S60asterisk .

For authenticate methods, please check here: http://blog.nyworldphone.com/2008/02/06/how-to-authenticate-users-by-callerid/.

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[voipagain]
type=friend
username=xxxxxxxxxxxx
secret=xxxxxx
port=5060
host=66.150.247.189
fromuser=xxxxxxxxxxxx
context=voipagain
canreinvite=no
insecure=invite
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
qualify=yes

I checked all posts. Finally I got it work. One thing it seems to be that the order matters.

[i2telecom.com]
allow=ulaw
canreinvite=no
context=voicestick
disallow=all
dtmfmode=rfc2833
fromdomain=i2telecom.com
fromuser=1xxxxxxxxxx
host=i2telecom.com
insecure=invite
nat=yes
outboundproxy=206.165.50.116
port=5060
secret=xxxxxx
type=peer
username= 1xxxxxxxxxx

When I used this, it didn’t work.
When I call the number, it gave me error message:
NOTICE[18415]: chan_sip.c:3588 process_sdp: No compatible codecs!

Then I tried another one, it worked.

[i2telecom.com]
type=peer
username=1xxxxxxxxxx
secret=xxxxxx
port=5060
host=i2telecom.com
fromuser=1xxxxxxxxxx
fromdomain=i2telecom.com
context=voicestick
outboundproxy=206.165.50.116
canreinvite=no
insecure=invite
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
qualify=yes

Both of these two are from some posts and are almost the same except the order.

We have to change

allow=ualw
disallow=all

to

disallow=all
allow=ulaw

for it to work.
In the first case, all codecs are disallowed I think. 2nd one, ulaw is allowed.

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OP