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Supported devices
ATA Adaptors
IP Phones
Cordless DECT IP Phones
IP Conference Phones
IP PBX Software/Hardware
Desktop Softphones
Mobile Softphones

Security considerations for Callcentric customers

3CX
3CX Version 6
3CX Version 12
3CX Version 12.5
3CX Version 15
3CXPhone (for Windows)
3CXPhone (Mobile)
Aastra 6753i
Acrobits/Groundwire for iPhone
Android SIP Client
Apivio MWP1100
Asterisk 1.4 and 1.2
Asterisk 1.6
Asterisk 1.6.2, 1.8, and 10
Asterisk Admin GUI
Bria Desktop
Bria Mobile
Callcentric Android App
Callcentric iPhone App
Callcentric Softphone
Cisco ATA 186/188
Cisco SPA112/SPA122
CloudTC Glass 1000
CSipSimple
D-Link DVG-1402S
Ekiga
Elastix
Gigaset A510 IP
Gigaset C610A IP
Gigaset DX800A
Grandstream DP715/710
Grandstream GXP1450
Grandstream GXP1620
Grandstream GXP1625
Grandstream GXP2110
Grandstream GXP2160
Grandstream GXP2200
Grandstream GXV3140
Grandstream GXV3240
Grandstream GXV3275
Grandstream HandyTone 286
Grandstream HandyTone 486
Grandstream HandyTone 702
Grandstream Wave
HTek UC803
HTek UC926
Innomedia SIP MTA-6328
Jitsi
Linksys PAP2
Linksys SPA3102
LinPhone Desktop
LinPhone Mobile
NCH Axon PBX
NCH Express Talk
Nokia E90
Obihai
OBi100/110
OBi200/202
OBi1032
pbx-in-a-flash
pbxnsip
pbxnsip Version 3
PhonerLite
Polycom SoundPoint IP 601
Polycom SoundStation IP 5000
snom 1xx/2xx/3xx
snom 820
snom M9
snom ONE
SFLphone
SJphone
Telco AC-211
trixbox / Elastix / pbx-in-a-flash
Twinkle
Windows Messenger
UTStarcom F3000
Vodia PBX
VTech VCS754
VTech VSP600
VTech VSP725
VTech VSP726
VTech VSP735
VTech VSP736
X-Lite / X-Pro / eyeBeam
Yealink T32G
Yealink W52P
Yealink T42G
Yealink T46G
Yeastar MyPBX U100
ZoIPer
ZoIPer 3.2
ZoIPer Mobile
Zoom 5801
Other Linksys/Sipura products
Generic / Other Device

DID-Based Routing with Asterisk
DID-Based Routing with trixbox / Asterisk Admin GUI / Elastix / PBX-in-a-Flash
UTStarcom F3000

GENERAL INFORMATION
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.

Many users are stuck with ATAs (Analog Telephony Adapters) and softphones. This essentially limits you to your local environment and requires many changes, or otherwise cumbersome adjustments, once you decide to become mobile.

Luckily with the introduction of WiFi SIP phones to the market users now have more options for mobility and more comfortable VoIP experiences. Once such WiFi phone is the UTStarcom F3000 which is a clamshell factor device. This device features all of the features you would expect from your standard SIP phone plus a small size which makes it easily portable.


Configuring F3000
Here we will configure the F3000 through the web interface in order to register to Callcentric to allow you to place and receive calls. The information presented in this guide is based on Version 5.90st of the F3000 firmware. If you are running a different firmware version some of menu options and settings may be different. This guide assumes that you have connected your F3000 to a wireless gateway, have a valid IP address and that you have administrative access to the administration web interface.

NOTE: We have tested the F3000 and found that the method it uses for call transfer is not supported by Callcentric, in either blind or consult mode. You can however use call holding properly with our servers.

We recommend that you read each step through in its entirety before performing the action indicated in the step.
STEP 1 Logging into your device
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:



STEP 2 Configuring your Callcentric account
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:

FieldSetting
SIP Terminal Use Outbound Proxy:Yes
SIP Terminal Use Register:Yes
SIP Profile Name:Callcentric
SIP Proxy Domain Name:callcentric.com
SIP Proxy IP Address:0.0.0.0
SIP Proxy Port:5060
SIP Register Domain Name:callcentric.com
SIP Register IP Address:0.0.0.0
SIP Register Port:5060
SIP Authentication String:This is either the default extension 1777MYCCID OR 1777MYCCIDEXT, where 1777MYCCID is the 1777 number assigned to you by Callcentric and EXT is the three digit extension you are trying to register this UA to.

For example: 17770001234101 would register to extension 101 on account 17770001234.

You cannot register to your account using only the extension number.
SIP Terminal User Name:This is either the default extension 1777MYCCID OR 1777MYCCIDEXT, where 1777MYCCID is the 1777 number assigned to you by Callcentric and EXT is the three digit extension you are trying to register this UA to.

For example: 17770001234101 would register to extension 101 on account 17770001234.

You cannot register to your account using only the extension number.
SIP Terminal Password:Enter your extension SIP Password here. Your extension SIP password is the password you created for the extension you are trying to use. You may edit the SIP password you wish to use in by logging into your My Callcentric account and clicking on the Extension menu link and then modifying the appropriate extension.
SIP Terminal Port:5060
SIP Terminal Use Null Packet:No
SIP Terminal Use DNS:Both Register And Proxy Servers Use DNS


Make sure to click submit to save your changes.



STEP 3 Codec and DTMF
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.



STEP 4 DNS type
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].



STEP 5 That's it! You can now make a phone call.
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).