ClicknCall frequently asked questions
Our user manual is available for download here.
Q: Do I need any Internet connection to use VoIP?
A: No, you can use our ANI callback service to enjoy the cheap VoIP phone rates.
Q: Shall I disconnect my phone line?
A: Please keep your existing phone line for safety reason.
For example when calling 000 emergency number in a power outage situation you need your existing phone line.
Q: Can I dial premium rate phone number?
A: We currently do not support calling of premium rate phone numbers.
Q: Your registration page doesn't ask me for details, how do I receive my account details after I pay?
A: When your finish your payment we will receive your email, name and phone number from PayPal and these are the details we use to create your account and email you the account details once payment clears.
Q: After I register how long will it take before my new account become active?
A: Normally your account details will be emailed to you within a couple of minutes after payment clears. Please be sure to check your email spam folder in case our emails get caught there.
Q: How can I check my balance?
A: You can either log into our website using your username/password or dial 333 from your sip phone/ATA.
Q: When I topup my account online will the credit be available instantly?
A: In most cases Yes. As soon as your payment clears credit will be available to you.
Q: How much does "callback" cost?
A: ANI callback and web callback involves two legs so there are two call charges if both legs are connected.
For examples:
- If you use callback to connect two Australian landline numbers the total cost will be 10c + 10c = 20c regardless you talk for 1 minute or two hours.
- If you connect an Australian mobile number to an Australia landline number and talk for 5 minutes the cost will be 19c * 5 minutes + 10c = $1.05
- If you connect an Australian mobile number to another Australia mobile number and talk for 5 minutes the cost will be 19c * 5 minutes + 19c * 5 minutes = $1.9
- If you connect an Australian landline number to a number in USA total cost will be 10c + 20c = 30c regardless you talk for 1 minute or two hours.
Q: I have a particular brand of VoIP router, will it work with your service?
A: Our VoIP service should work with most(if not all) types of VoIP router/ATA as long as they are not locked.
We might not have specific instructions for your particular brand of router/ATA but if you send us a screen shot of your VoIP router/ATA configuration page we will do our best to supply you with the relavant details to help you set it up.
Q: What is "ALG" and do it need it on my router?
A: Many comercial routers implement ALG (Application-level gateway), while the intention is good but because of poor implementation it causes a lot of trouble with VOIP.
If you have frequent dropped calls and trouble with in-coming calls here is how to disable ALG on your router.
Q: Can I make multiple calls at the same time using one account?
A: Yes, as long as your calling pattern does not violate our terms and conditions.
By default you have unlimited channels available to call Australian mobiles and landline numbers and 2 channels to call international numbers.
Please email us with your account number if you need to make more than two international calls at any one time.
Q: Can I dial my voip/sip phone from a regular landline in Australia?
A: Yes you can by following these steps:
1) Make sure your sip device is registered with our sip server on port 5060.
2) Contact us to activate this "dial-in" feature on your account.
3) Call one of our conference numbers and at the prompt "Please enter conference pin" simply punch in your 10-digit username and your sip phone shall start to ring.
Q: Can I call other Clickncall members free?
A: Yes. From your sip phone just dial the other member's 10-digit username.
OR, you can also call other CnCn members using the conference facility. Click on "Conference Call" after logged into the CnCn website, click on "Invite" and in the pop up window put in "cncnXXXXXXXXXX" where XXXX... is the CnCn member number.
You can ring your own sip phone using this facility. You will hear music onhold(from the conference room) if you are the first one called.
Q: How much does it cost to ring 13xxxx number?
A: 25 cents untimed. Here's a tip on how to reduce this cost:
Don't call the 13xxxx number directly, rather call the land line number that is linked to that 13 or 1300 number.
You can find out the equivalent land line number from here.
By calling a land line number from our VoIP service you reduce your cost from 25¢ to 10¢.
If you can't locate the land line number from the above link, next time you ring that 13 number just ask the owner/operator what their equivalent land line number is.
Q: What are the difference between different Codecs and which one shall I use with my IP phone or softphone?
A:
When making a call over the Internet, the software (soft-phone) or hardware needs to use a codec to send/receive information in a certain format and convert it to the sounds you hear.
Generally, a codec with a higher bandwidth requirements provides better voice quality (If your Internet connection is fast enough to support the codec). Most VoIP providers/hardware/licensed software will support G.711 and G.729 (However be sure to check this before purchasing hardware, or signing up with a VoIP provider!). The G.711 codec requires a connection almost 3 times faster than that required by the G.729 codec.
If you are using a free soft-phone, then G.729 will not be available to you; however, the GSM codec should be, and will give you similar call quality to that of a mobile phone.
The following table shows bandwidth requirements for many common codecs. For more informatio on codecs please refer to this Codecs Wiki.
Codec.................Bandwidth Usage (Up/Down)
G.711 (64 Kbps).......87.2 Kbps
G.729 (8 Kbps)........31.2 Kbps
G.723.1 (6.3 Kbps)....21.9 Kbps
G.723.1 (5.3 Kbps)....20.8 Kbps
G.726 (32 Kbps).......55.2 Kbps
G.726 (24 Kbps).......47.2 Kbps
G.728 (16 Kbps).......31.5 Kbps
GSM (7 or kbps).......low
ILBC (low)............low
We recommend you to use G.711 (ulaw or alaw) with our VoIP services if you are on an above 512M broadband connection. Using this codec you can have "Hi-Fi" quality sound.
If you are on a slower 64k ADSL connection then the G.729 codec is more suitable (it will lose some sound quality due to compression but will give you an overall acceptable audio).
Q: Does credit expire?
A:
Credit may expire if your account is inactive ("inactive" means not making any out-going calls within any 12 month period).
Q:X-lite takes nearly 1 minute before it dials the number I enter, how to fix that?
A:
This is probably a DNS issue with your PC. Please change the DNS servers to the ones provided by your ISP rather than using the IP address of your router as the DNS servers.
Q: I get X-lite registration error 404, what shall I do?
A:
Please make sure your username and password are correct. Please type them in rather than copy and paste from the email received from us..
A lot of the cases when using "copy and paste" there is an extra space added to your username thus making it incorrect.
Q: How to solve X-lite "Request Timed Out" error or one-way audio problem?
A:
This usually indicates a network connection problem at your end that is blocking some of the VoIP packets.
Please try shutting down any firewall you have on your PC or router and see if it solves the problem. If not, please hang on to your hat and read on ...
Please make sure UDP port 5060 and UDP port range from 10,000 ~ 20,000 are open from your modem/router. You shall do a port forward on your router/modem. For information on how to port forward your particular router/modem please Click Here (opens a new window). Choose your brand of modem/router from the page and then on next page click "skip advertisement" near the top right corner and then on the next page click "X-lite".
Q:It just doesn't work, what should I do?
A:
First step is to gather as much details as you can and contact us providing these information, the more the better as it will help us to help you:
- How are you using the service? using callback, calling card or using your own SIP device?
- If you use your own ATA what is the exact make/model and brand of your modem/router and ATA?
- What is the datetime of the call?
- What is the actual number you have issue calling?
- Is problem with just one number or all numbers?
- Any other information you think is relavant.
