<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;"><p><a rel="nofollow" target="_blank" href="http://www.callcentric.com/?aid=134925"><span class="yshortcuts" id="lw_1305145030_0">Callcentric</span></a>
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 (<span class="yshortcuts" id="lw_1305145030_1">Origination</span>
/ DID / DDI ) within the USA, Canada, and other countries. Callcentric
supports softphones, VoIP ATA's, VoIP Phones, and IP PBX equipments such
as <a rel="nofollow" target="_blank" href="http://www.asterisk.org"><span class="yshortcuts" id="lw_1305145030_2">ASTERISK</span></a> .</p>
<p>With the Online <span class="yshortcuts" id="lw_1305145030_3">Calling Card feature</span> you can use your <a rel="nofollow" target="_blank" href="http://www.callcentric.com/?aid=134925">Callcentric</a>
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.</p>
<p>Here is a basic configuration.</p>
<p>
</p><table border="0" cellpadding="5">
<tbody>
<tr>
<td class="yiv256432349dev_step"><font size="2">1</font></td>
<td class="yiv256432349small8pt"><b><font size="2">Edit file sip.conf:</font></b></td></tr>
<tr>
<td><br></td>
<td>
<ul><li><font size="2">Add/change [general] section to indicate the following parameters:<br></font><span class="yiv256432349dev_mono"><pre><font size="2"><font color="#ff0000">[general]
dtmfmode = rfc2833
context=from-callcentric
srvlookup=yes
register => 1777MYCCID:<span style="border-bottom: 2px dotted rgb(54, 99, 136); cursor: pointer;" class="yshortcuts" id="lw_1305145030_4">SUPERSECRET@callcentric.com</span>/1777MYCCID
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas</font>
</font></pre></span><br>
</li><li><font size="2">Add the following section to handle calls to/from callcentric:<br></font><span class="yiv256432349dev_mono"><pre><font size="2"><font color="#ff0000">[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
defaultuser=1777MYCCID
secret=SUPERSECRET
fromuser=1777MYCCID
fromdomain=callcentric.com
insecure=port,invite</font>
</font></pre></span><br>
</li><li><font size="2">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:<br></font><span class="yiv256432349dev_mono"><pre><font size="2"><font color="#ff0000">[123]
context=to-callcentric
type=friend
username=123
secret=PASSWORD
host=dynamic</font>
</font></pre></span></li></ul></td></tr>
<tr>
<td class="yiv256432349dev_step"><font size="2">2</font></td>
<td class="yiv256432349small8pt"><b><font size="2">Edit the file extensions.conf:</font></b></td></tr>
<tr>
<td><br></td>
<td>
<ul><li><font size="2">Add the following section to route calls FROM callcentric TO your SIP phone with extension 123:<br></font><span class="yiv256432349dev_mono"><pre><font size="2"><font color="#ff0000">[from-callcentric]
exten => s,1,Dial(SIP/123)</font>
</font></pre></span><br>
</li><li><font size="2">Add the following section to route calls FROM your SIP phone TO callcentric:<br></font><span class="yiv256432349dev_mono"><pre><font size="2"><font color="#ff0000">[to-callcentric]
exten => _X.,1,Dial(SIP/${EXTEN}@callcentric)</font>
</font></pre></span></li></ul></td></tr>
<tr>
<td class="yiv256432349dev_step"><font size="2">3</font></td>
<td class="yiv256432349small8pt"><b><font size="2">Verify Asterisk operations</font></b></td></tr>
<tr>
<td><br></td>
<td>
<ul><li><font size="2">Connect to asterisk console by running:<br><font color="#ff0000"><span class="yiv256432349dev_mono">asterisk -r </span><br></font><br></font>
</li><li><font size="2">Verify that Asterisk is registered to callcentric with console command 'sip show registry'<br></font><span class="yiv256432349dev_mono"><pre><font size="2"><font color="#ff0000">*CLI> sip show registry
Host Username Refresh State
callcentric.com:5060 1777MYPHONE 17 Registered</font>
</font></pre></span><br>
</li><li><font size="2">Verify that your SIP phone is registered to Asterisk with console command 'sip show peers'<br></font><span class="yiv256432349dev_mono"><pre><font size="2"><font color="#ff0000">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</font>
</font></pre></span><br><font size="2">If you see Host as "(Unspecified)" and Port as "0", then your SIP phone is not configured correctly.<br><br></font>
</li><li><font size="2">Disconnect from Asterisk by typing "exit". </font></li></ul></td></tr>
<tr>
<td class="yiv256432349dev_step"><font size="2">4</font></td>
<td class="yiv256432349small8pt"><b><font size="2">Placing Test Calls</font></b></td></tr>
<tr>
<td><br></td>
<td><font size="2">You can make a test call to <span class="yshortcuts" id="lw_1305145030_5">17771234567</span>,
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:<br><span class="yiv256432349dev_enter">1 + the area code and number for calls to the US</span><br><b>Or</b><br><span class="yiv256432349dev_enter">011 + the country code, area code, and number for calls worldwide</span> (you may also use 00 instead of 011). <br><br></font></td></tr></tbody></table>
<p>Read more on <a rel="nofollow" target="_blank" href="http://www.callcentric.com/?aid=134925"><span class="yshortcuts" id="lw_1305145030_6">CallCentric's</span></a> website ...</p>
<p> </p><br></td></tr></table>