Here, I have listed some of the various ways to make free calls through your CallCentric VoIP account. To make these free calls, you do not require any paid subscriptions. Typically, I only subscribe to pay-per-minute plans and only pay for what I use. But, the methods I describe here allow for totally free calling, even if you don't have a pay-per-minute plan.
There are two primary methods of free calling: VoIP-to-VoIP and PSTN-to-VoIP
- VoIP-to-VoIPCalling through your CallCentric account using an ATA, IP-Phone, or Softphone:
- In-Network calling (between CallCentric users)
- Peer Network calling using SIP Broker (**275* + peer code)
- Call to iNum numbers (+883 global country code)
- Call to SIP URI's in peer networks (firstname.lastname@example.org)
Calling through your CallCentric account using Calling Card Gateways
- Call any In-Network number (to any other CallCentric user number)
- Call any iNum number (world-wide +883 numbers)
- Call Peer Network numbers through SIP Broker Gateway (**275*)
- Call any peered network number, or SIP URI, using your CallCentric Phone Book (*75xx Speed Dial Feature)
Free In-Network Calling is as simple as dialing any other CallCentric 11-digit user account number in the format of 1777xxxxxxx using your ATA, IP-Phone, or softphone. In-network calling can be used anywhere in the world that has high-speed Internet access. For example, if you dial 17771234567, you will connect to the TellMe Information services.
You can also dial any other CallCentric user by first calling a local CallCentric Call Card Gateway access number. Once you have been validated by the Call Card IVR, just dial the CallCentric user account number you wish to call (1777xxxxxxx). For example, if you are in Toronto, Ontario Canada, you would dial the local access number (416-477-2100). At the IVR prompt, enter your CallCerntic 11-digit account number (1777xxxxxxx), at the next prompt, enter your Call Card PIN number as set in your user account, and finally, at the prompt enter the destination number (i.e. 17771234567 for TellMe InfoLine)
Note: Calling Card access requires that you first enable Global Access Numbers in your CallCentric user control panel (Dashboard), in the Preferences section.
Free Peer Network Calling can be achieved by dialing the SIP Broker access code **275*, through your CallCentric connected device, followed by the Peer Code for the network you want to call through, followed by the virtual number of the contact you wish to connect to. For example, dialing **275*747-474-3246 will connect you to the Gizmo5 Echo Test line. (Note: Gizmo5 was recently purchased by Google).
When calling through your ATA (I use Linksys SPA2102-NA), be sure the ATA Dial Plan includes the following entry: |**275*x.| to properly handle the Sip Broker dialing.
Calling (and receiving) any iNum number through your CallCentric connected device is always free. Calling iNums through the CallCentric Calling Card Gateway, as a local call, is also free through our CallCentric account (your only cost is what your PSTN or cell phone provider charges you for local calls to the gateway number). iNums are dialed as +883-5100-xxxx-xxxx. For example, to dial the iNum Echo Test number, dial as follows: +883-5100-0000-0091. When dialing through your CallCentric account, you may need to replace the (+) with the IDD international dialing code 011. Thus, you would dial 011-883-5100-0000-0091.
SIP URI Dialing
You can also call any SIP URI (Uniform Resource Identifier) that terminates on a network that has peering agreements with CallCentric. SIP URI's are typically dialed as: sip:email@example.com . For example iNums can be dialed as: sip:firstname.lastname@example.org. (depending on your VoIP User Agent device, preceding the URI with "sip:" isn't always necessary... you may have to try both ways.
I have provided a few SIP numbers for testing (all free calls):
- sip:email@example.com (Echo Test)
- sip:firstname.lastname@example.org (TellMe Information)
- sip:email@example.com (Time Announcement EST)
Of course, most phones connected to VoIP ATA's can't dial SIP URI's directly. So, the way to get around this is to use CallCentric's phone book Speed Dial or Click2Dial features, or your Cisco Linksys ATA speed dial feature.