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| STEP 1 |
Logging into your device |
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Log into the F3000 administrative interface and click on the USER MENU option from the left side of the page. A menu with the various options you will need to edit will be displayed as below:
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| STEP 2 |
Configuring your Callcentric account |
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Next click on the SIP Config option from the left side of the page. You will now be given the ability to edit your SIP server settings.
Here we will configure your F3000 to register to the Callcentric servers. Please use the information below to assist you:
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| SIP Terminal Use Outbound Proxy: | Yes |
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| SIP Terminal Use Register: | Yes |
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| SIP Profile Name: | Callcentric |
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| SIP Proxy Domain Name: | callcentric.com |
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| SIP Proxy IP Address: | 0.0.0.0 |
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| SIP Proxy Port: | 5060 |
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| SIP Register Domain Name: | callcentric.com |
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| SIP Register IP Address: | 0.0.0.0 |
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| SIP Register Port: | 5060 |
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| SIP Authentication String: | 1777MYCCID |
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| SIP Terminal User Name: | 1777MYCCID |
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| SIP Terminal Password: | Your SIP password |
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| SIP Terminal Port: | 5060 |
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| SIP Terminal Use Null Packet: | No |
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| SIP Terminal Use DNS: | Both Register And Proxy Servers Use DNS |
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Make sure to click submit to save your changes.
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| STEP 3 |
Codec and DTMF |
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Now we will configure the codecs you will use with the Callcentric services as well as the DTMF transport you will use. From the left side of the page
select Codec and DTMF. Now set the priorities according to the information below:
G711u: 5
G711a: 3
G729a: 4
G726: 0
And set your DTMF Transfer Mode to RFC2833.
Once done click on submit to save your settings.
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| STEP 4 |
DNS type |
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Before you reboot your phone you will want to make sure that your DNS Query Type is set to SRV on
your RTP and STUN page. You can also verify that Use STUN is set to do not use STUN.
Once done click on [submit].
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| STEP 5 |
That's it! You can now make a phone call. |
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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).
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Among many of the draw points for VoIP is the ability to use your service from pretty much anywhere you have a stable internet connection. SIP further expands this by offering a common platform for interoperability.