ClicknCall frequently asked questions
Our user manual is available for download here.
Q: Do I need any Internet connection to use VoIP?
A: Not necessarily, you can use our callback service to enjoy the cheap VoIP call rates.
Q:We are moving to NBN, how do we use your phone service?
A: You can go with any NBN provider of your choice. Most NBN providers will offer you an Internet service plus a phone service, but you don't
have to take up their phone service. Instead you can opt for an Internet only service from your NBN provider and use ClicknCall to make phone calls over your NBN Internet connection.
You also need some VoIP capable modem or device in order to use a regular phone.
You can rent a DID(phone number) from us to receive incoming calls or port your number to us.
Q:How do I call emergency 000 number?
A: Please keep your mobile phone handy for safety reason as our service can't be used in case there is a power outage or interruption to your Internet connection.
Q: Can I get a SIM card from ClicknCall?
A: No, we do not offer sim card. However you can make cheap calls via your existing Internet connection (be it WiFi or 3G/4G). To make Internet calls to normal phone numbers using your smart phone please read here.
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: Your account details will be emailed to you within a minute after payment clears and your account is active straight away.
Please check your email spam folder in case our emails get caught there (this happens a lot with Yahoo and hotmail users).
Q: Can I dial premium rate phone number?
A: We currently do not support calling of premium rate phone numbers.
Q: I can call Australian numbers fine but just get a busy tone when calling certain international numbers, what should I do?
A: By default, not all international destinations are enabled on new account for security reason.
Please email us the international destinations you like to call and your account number if you like to remove this limit on your account.
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 new credit be available instantly?
A: In most cases Yes. As soon as your payment clears credit will be available to you.
Please remember to log into your account first then click on "Recharge account" near the top to recharge.
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 15c * 5 minutes + 10c = $0.85
- If you connect an Australian mobile number to another Australia mobile number and talk for 5 minutes the cost will be 15c * 5 minutes + 15c * 5 minutes = $1.5
- 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 and that your modem router is not blocking any voip traffic.
2) Call one of our conference numbers (you can find the list of numbers after logging into your member area)
and at the prompt "Please enter conference pin" simply punch in 777#
3) At the prompt enter the 10-digit username and your sip phone shall start to ring.
Another option is you can also rent a dedicated number (DID) from us to receive incoming calls.
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 ClicknCall members using the conference facility. Click on "Conference Call" after logged into the ClicknCall website, click on "Invite" and in the pop up window put in "cncnXXXXXXXXXX" where XXXX... is the 10-digit 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: Can I create multiple sip extensions under one account?
A: Yes, you can create up to 10 sip extensions from the web portal (more can be requested).
Multiple extensions enable sip registration from multiple devices (e.g., from your smart phone, PC, other family members' devices).
This is handy when you want to use the same account on multiple sip devices. You can create multiple extensions for free calling between your branches/offices.
You can also check your sip device(s) registration status live and see if your extensions are registered to the Clickncall sip server properly or not.
Q: Can I use your service overseas?
A: Yes, you can. There are multiple ways of doing this including using "webcall" or sip calls (sip calls requires you have a relible Internet connection while oversease and that the ISP is not blocking VoIP in that country).
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. Typically using this codec normally consumes about 1MB per minute of call duration if you use your Internet connection(be it Wifi or mobile data).
If you are on a slower 64k ADSL connection then the GSM or iLBC codecs are 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 remains inactive ("inactive" means not consuming any billable services such as making calls, sending sms etc within a 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 sip registration error, and your website is totally un-responsive, what shall I do?
A:
Please ensure the sip server, your sip username and password you enter are exactly the same as the ones contained in the original Welcome email received when you sign up.
Please also note our firewall might block your IP address after multiple failed login attempts. If you think this is the case please email us your public IP address and we can unblock it from our firewall.
Or simply wait an hour or so and try again with the correct authentication details.
Q: How to resolve one-way audio issue?
A:
One way audio (and sometimes no audio at all) is usually caused by your modern/router blocking some of the voip packets to your sip device
(which usually sits behind a firewall/NAT of your modem/router).
You can do a "port forward" from your modem/router to your sip device. The ports needed to be forwarded are UDP port 5060 and UDP port range 10000~20000.
We also have a "stun" server you can use to help traverse your NAT. Please contact us if you need help on port forwarding and setting a stun server in your sip device.
Q: I have done "everything", it just doesn't work, what should I do?
A:
It would really help us to help you if you can provide more details on this "everything" you have done.
Please gather as much details as you can and contact us with this information, the more details the better:
- How are you using the service? Are you 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?
- If you use a soft phone what is it? Are you using a PC, Mac, iPad/iPhone or Android?
- What is your Internet connection like (ADSL, dialup, 3G/4G or NBN) ?
- What is the datetime of the call? What is the actual number you have issue calling?
- Any recent change from your ISP or your equitments?
- Is it problem with just one number or all numbers?
- Any other information you think is relavant.