[asterisk-biz] Unlimited VoIP calls worldwide with Asterisk

[toto tata]

Callcentric 
 is a service that provides VoIP based Broadband Phone service using the
 SIP protocol for personal / residential and business users. Services 
include outbound calling (termination), inbound calling (Origination
 / DID / DDI ) within the USA, Canada, and other countries. Callcentric 
supports softphones, VoIP ATA's, VoIP Phones, and IP PBX equipments such
 as ASTERISK .
With the Online Calling Card feature you can use your Callcentric 
 account to place calls from a regular phone such as a Cell/Mobile phone
 while on the road. Calls placed using the Calling Card feature are 
billed at Callcentric's low Pay Per Call rate plan calling rates, 
regardless of which rate plan you have on your account.
Here is a basic configuration.




1
Edit file sip.conf:




Add/change [general] section to indicate the following parameters:
[general]
dtmfmode = rfc2833
context=from-callcentric
srvlookup=yes
register => 1777MYCCID:SUPERSECRET at callcentric.com/1777MYCCID
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas


Add the following section to handle calls to/from callcentric:
[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
defaultuser=1777MYCCID
secret=SUPERSECRET
fromuser=1777MYCCID
fromdomain=callcentric.com
insecure=port,invite


Add a section to handle calls to/from your SIP 
phone. This is just a sample. Refer to Asterisk documentation and your 
SIP phone documentation for details. 123 is the extension of your phone:
[123]
context=to-callcentric
type=friend
username=123
secret=PASSWORD
host=dynamic


2
Edit the file extensions.conf:




Add the following section to route calls FROM callcentric TO your SIP phone with extension 123:
[from-callcentric]
exten => s,1,Dial(SIP/123)


Add the following section to route calls FROM your SIP phone TO callcentric:
[to-callcentric]
exten => _X.,1,Dial(SIP/${EXTEN}@callcentric)


3
Verify Asterisk operations




Connect to asterisk console by running:
asterisk -r 


Verify that Asterisk is registered to callcentric with console command 'sip show registry'
*CLI> sip show registry

Host                    Username    Refresh State
callcentric.com:5060	1777MYPHONE	17      Registered


Verify that your SIP phone is registered to Asterisk with console command 'sip show peers'
pbx*CLI> sip show peers

Name/username   123/123
Host            10.11.22.33
Dyn Nat ACL     D
Mask            255.255.255.255
Port            5060
Status          Unmonitored

If you see Host as "(Unspecified)" and Port as "0", then your SIP phone is not configured correctly.


Disconnect from Asterisk by typing "exit". 

4
Placing Test Calls



You can make a test call to 17771234567,
 or if you are signed up for one of Callcentric's rate plans you can 
place a call to a traditional landline or mobile phone by dialing 
either:
1 + the area code and number for calls to the US
Or
011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011). 


Read more on CallCentric's  website ...
 
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