• Q. I'm in Singapore now (I use Xten softphone on my laptop computer), how should I dial a number to call my friend in UK?
• A. It doesn't matter where you are, always dial country_code-area_code-number. You have to dial 44-XXXXXXXXXX to reach a UK phone number. Don't forget about timezone difference between Singapore and UK:-)

• Q. Do you support 7-digit dialing in the US?
• A. Yes and No. The server uses country_code-area_code-number dial patterns only. However, you can set up the dialplan in your client program or ATA to add 1-area_code to the dialed number if only 7 digits are entered.

• Q. I'm calling USA number 371-555-2222, but your system connects me to a person in Latvia!
• A. Dial 1-371-555-2222.

• Q. My friend's telephone number in France is 01XXXXXXX, I dial 3301XXXXXXX, but can't reach him!
• A. In many countries (including most of European countries) leading 0 is a long distance prefix and should not be dialed when you dial a number starting with the country code. Dial 331XXXXXXX.

• Q. Can I dial 10 digits to reach US phone number?
• A. No, you have to dial 11 digits, 1-NNN-NNNNNNN. You can setup the dialplan in your ATA to automatically prepend 1 to a number if only 10 digits are dialed. But you will not be able to call numbers in other countries which are 10 digits long including country code (Latvia or Iceland for example).

• Q. My friend is your customer too, how can I call her?
• A. Dial her 9-digit long username. The calls between CallWithUs customers are always free.

• Special numbers to dial from SIP/IAX phone:
*67NNNNNNNNNNN - call number NNNNNNNNNNN and do not show your caller ID number to the called party.
*3RNNNNNNNNNNN - call number NNNNNNNNNNN using route R (1-9). If the least cost route does not work properly, try next route to make a call, see the list of available routes and corresponding rates in simulator. *31NNNNNNNNNNN means call the number NNNNNNNNNNN using cheapest route only, do not try another route if the cheapest route failed. Example: *329193NNNNNNNN - call India mobile phone using route number 2. *3R prefix can be used also in DID destination (if you forward the DID to PSTN number) and in speed dial menu.
0-99 - Dial a number assigned to the speed dial code 0-99 (regular calling rates apply).
086 (0VM) or 311 - call voicemail (free).
*0 - check account balance (free).
*1 - redial last dialed number(regular calling rates apply).
3246 (ECHO) - echotest (free).

• Q. I live in Munich, Germany. Can I order US phone number?
• A. Yes. See "Available DIDs".

• Q. If I purchase a DID in some country, the DID includes N free minutes, does it mean I have N free minutes to call telephone numbers in that country?
• A. No. DID is inbound only service to call you from a regular phone. The N free minutes are incoming minutes only. All outgoing calls are charged according to the calling rates.

• Q. What the DID parameter "channels" mean?
• A. Channels is the maximum number of simultaneous incoming calls the DID supports. To implement "call waiting" service the DID have to support at least 2 channels. If the number of simultaneous incoming calls exceeds number of DID channels, the caller will get BUSY signal.

• Q. I have my own VOIP DID, can I redirect it using SIP protocol to you server?
• A. Yes. If you want to redirect the DID to ring your SIP client registered with our server, then set the DID calls forward with your DID provider to sip:username@callwithus.com, where username is your 9 digits username, the prefix sip: or sip/ depends on your DID provider. If you want to redirect the DID to ring your PSTN number using our service or to have additional features provided by our service like caller ID with name, multiple DID destinations, then we need to know the DID number to enter it to our system as a DID with 0 monthly price and $2 setup fee. Set the DID call forwarding with your provider to sip:didnumber@callwithus.com that case.

• Q. How do I buy a phone number?
• A. Login into your account, select "DID" from the menu and click "Add a DID" button. Your account must have enough funds to cover the setup and first month usage charges. Use "Buy Calling Credit" buttons on "Account Info" page in your control panel to add funds to your account.

• Q. How do I specify the DID destination?
• A. If "VOIP call" is set to "Yes", then the destination field should contain VOIP URI prefixed with "SIP/" or "IAX2/" depending on required VOIP protocol. If "VOIP call" is set to "No", then the field should contain a regular telephone number (starting with the country code) to forward the call to (your will be charged a per-minute calling rate this case). You can specify multiple destinations for a DID, the system will try to reach each destination in the order specified by "Priority" field. "Wait time" field sets how long to ring the destination (if no answer) before trying the next one. We suggest to add your PSTN or cell phone as a "last chance" destination with priority 5 (VOIP_call=No), the DID forward to that phone will work as an alarm indicating that something is wrong with your VOIP destination(s) setup.
  The format of VOIP URI is [protocol/]username[[[:password]@your.domain.name]/extension]. Here are the examples:
  • [SIP]|[IAX2]/username - call your SIP/IAX client registered with our setver.
  • [SIP]|[IAX2]/username/extension - call your SIP/IAX client registered with our server. The call will go to the extension "extension" in your dial plan.
  • SIP/12345@fwd.pulver.com - forward DID call to your FWD account.
  • IAX2/myname:mypassword@10.23.11.17/101 - call IAX2 client at IP address 10.23.11.17, extension 101, authenticate the call using username "myname" and password "mypassword".
Please note that when DID call is forwarded to external URI, calls will come to your system from either east.callwithus.com or west.callwithus.com or sip.callwithus.com, your system should be configured to accept incoming calls from these hosts. If you run asterisk or trixbox, create 3 incoming trunks for these hosts. Here is an example:

[sip]
type=friend
host=sip.callwithus.com
context=from-trunk
insecure=invite

[east]
type=friend
host=east.callwithus.com
context=from-trunk
insecure=invite

[west]
type=friend
host=west.callwithus.com
context=from-trunk
insecure=invite

• Q. I have 2 SIP clients connected to your server, is it possible to get both devices ring when my DID is called?
• A. Yes. Assuming your devices have usernames 111111111 and 222222222, set the DID destination to "SIP/111111111&SIP/222222222", note the & sign. "VOIP call" for the destination must be set to "Yes". As soon as one device answers the call, the another one(s) will stop ringing.

• Q. My SIP device supports multiple internal extensions, how do I specify the DID destination to ring a particular extension on incoming DID call?
• A. SIP/extension@your.domain or SIP/username/extension

• Q. Can I setup the DID destination to call my FWD account?
• A. Yes, SIP/yourfwdnumber@fwd.pulver.com.

• Q. How long time it takes to activate the DID after purchase?
• A. The DID activation is instant.

• Q. How can I check if the DID number I am going to purchase works OK?
• A. Dial the number, you should hear recorded message in English "The number you dialed is not in service, check the number and try again later. Message DIDX 1234". Let your potential callers call that number, they should hear the same message.

• Q. Can I transfer my existing US phone number to your service?
• A. No, we do not port numbers. Our service is not a replacement for a home phone line, but an addition to it. We do not support 911 emergency calls. Use your land line or cell phone for emergency calls.

• Q. What kind of telephone adapters can I use with your service?
• A. You can use any telephone adapter which supports SIP or IAX protocol, like Sipura 1001, 2001, 3000, Linksys PAP2, 3102, Grandstream IP phones and adapters, Digium IAXY etc. Note: Skype phones and adapters will not work with our service.

• Q. Which codecs do you support?
• A. We support g711u, g711a, g729 and GSM codecs.

• Q. How to check if my phone adapter is connected to your server?
• A. Login into your account and select "VOIP Accounts" from the menu. You should see the registration status of VOIP client. If the value is "Not registered", double check your configuration. Dial 3246 (echo test) to check the voice quality.

• Q. Can I connect more than 1 SIP device to your server?
• A. Yes. Login into your account, select "VOIP Accounts" from the menu and click "Add" button to create an additional VOIP account for the second device.

• Q. My asterisk server is connected to your server using IAX protocol, I set the DID destination to "IAX/username", but the incoming call do not go through!
• A. The technology specification for IAX is "IAX2", set the DID destination to "IAX2/username".

• Q. My asterisk server supports both SIP and IAX protocols, which one is better?
• A. Use the protocol you are the most comfortable with. We prefer SIP. The rule of thumb is - if your server is on a public IP address, not behind NAT router, then SIP is a better choice, Set "nat=no, canreinvite=nonat" in VOIP account settings on our web site to enable SIP reinvite to get the best sound quality, if your server is located on a private LAN, then IAX may work better for you, it is more firewall-proof.

• Q. Do you support SIP reinvites?
• A. By default SIP reinvite is disabled. If your SIP device is on a public IP or your firewall/router are configured to pass RTP from any IP address, then set "nat=no, canreinvite=nonat" in VOIP account settings to enable SIP reinvite on the VOIP account to get the best voice quality. If your setup can not handle SIP reinvite, but you enabled it in VOIP account settings, you will get one way audio problem.

• Q. Can I enable SIP reinvites if my client is behind NAT?
• A. In most cases yes, if your client supports STUN server or router/firewall are configured properly.

• Q. When somebody calls my DID the phone connected to my PAP2T does not ring and the caller gets busy signal!
• A. Power cycle your ATA, it is a known problem with PAP2T adapters - on incoming call the adapter tries to ring the phone and reboots. Daily power cycling usually helps.

• Q. My Linksys (or Sipura) ATA looses dial tone and I have to reboot it to get it working! How can I fix the problem?
• A. In ATA settings go to Admin/Advanced/SIP tab. In "SIP Timer Values" group change the following parameters:
  • SIP T1 - 1 (default value 0.5)
  • Reg Retry Long Intvl - 120 (default value 1200)

• Q. What is the difference between caller ID setting in "Add caller id" menu and "VOIP accounts" menu?
• A. Caller ID set in "Add caller id" menu has the highest priority and overwrites value set in "VOIP accounts" menu. It is a quick and simple way to set caller ID for outgoing calls. "VOIP accounts" menu alows you to set caller ID individual for each device you use with our service.

• Q. What "activated" Caller ID in "Add Caller ID" menu means?
• A. Caller ID numbers in "Add caller id" menu work for authentication if you use our access number/callback features and to set caller id number when you make outgoing call. All numbers entered work for authentication, but only activated number will be set as caller id on outgoing call.

• Q. I run asterisk server, can I set caller ID in my server dialplan?
• A. Yes. Remove all caller ID numbers in "Add caller ID" menu, in "VOIP accounts" menu set caller ID to blank. The caller ID set by your asterisk server will be forwarded as is after that. Please note, your asterisk must be registered with our server this case.

• Q. Will the caller id name I entered in ATA settings be shown to the called party?
• A. Yes if you make a VOIP call, and no if you call PSTN number. PSTN (at least in North America) does not transmit caller id name, the local phone carrier of the called party does CNAM database lookup to find caller id name by caller id number.

• Q. How do I access my voice mail box?
• A. Dial 086 (0VM) from your SIP or IAX phone or softphone. To access voice mail box from a PSTN or cell phone dial your DID number, wait for the voice mail system prompt and press the star key on the phone. Use the following chart to navigate voicemail menu system.

• Q. What is my voice mail password?
• A. Login into your account and select "Voice Mail" menu to see the current password and/or change it.

• Q. How to activate the Voice Mail box?
• A. Login into your account, select "Voice mail" menu and click "Setup Voice Mail..." button. You account will be charged $1 setup fee and $1 on 1st of each month thereafter. Remove the VM box at any time to stop monthly charges (all messages and recorded greetings will be removed).

• Q. Can I use my ATA with VPN connection or I'm limited to soft phones only?
• A. Yes, you can use ATA. This requires some manual configuration of your network. Assuming your router has internal IP address 192.168.1.1 and gives network clients IP addresses in the 192.168.1.100-255 range (check your router configuration for details), you need to configure static IP addresses on the PC which is running OpenVPN and ATA.
  On PC
  • Set IP address to 192.168.1.70, netmask 255.255.255.0, DNS server address to the same value which is set if you use "automatic" IP address configuration on your PC (the values are shown with "ipconfig /all" command).
  • Enable Internet Connection Sharing on the OpenVPN network interface.
  On ATA
  • Set IP address to 192.168.1.71, netmask 255.255.255.0, default gateway 192.168.1.70 (PC IP address), SIP server address 10.39.0.1.
  In short, the PC has to share internet connection on OpenVPN interface and ATA have to be configured to route IP packets to 10.39.0.1 to the PC IP which runs the OpenVPN client. If you have no clue what the above is about, hire a network guru to do the configuration for you.

• Q. What is a billing unit to call a PSTN number?
• A. 60 seconds.

• Q. Is there the service cancellation fee?
• A. No. The initial 30 cents credit is not refundable. We charge 10% refund fee if you made a payment but can't configure your VOIP equipment to work with our service. Get your equipment working before making a payment.

• Q. Do you offer calling credit?
• A. No. Our service is prepaid only.

• Q. Which payment methods do you accept?
• A. We accept credit card payments with Google Checkout, Paypal and bank wire transfer.

• Q. I made a payment one hour ago, but my account balance does not reflect the payment!
• A. It could take up to 24 hours to apply the payment to your account, we delay the account funding to prevent the service fraud. Your account will be set up to immediate payment processing after 2 months.

• Q. Can I setup an automatic refill of my account?
• A. No. We have no access to your credit card or PayPal account, we can't initiate the charge, only you can do it.

• Q. What is the best way to not miss a monthly payment for my DID (and don't loose the number!)?
• A. Set the "Alert Balance" in "User settings" menu to be higher than the monthly DID price. Our server will send you daily notification emails if your account balance falls below the treshold you set.

• Q. How can I find out the calling rate to a particular phone number?
• A. Login into your account, select "Simulator" from the menu and eneter the number you wish to call.

• Q. Why does the simulator show more than one result with different rates?
• A. The server automatically selects the least cost route to reach the number you dialed. But if this route is not currently available (which is very unlikely), then the next route will be chosen. We do the best to complete your call using the least cost route.

• Q. How to prevent accidental calls to destinations with a high call price?
• A. Set "Max Rate" parameter in "User Settings" menu to the desired value. Please note that you will not be able to make calls to some destinations if the value is too low.

• Q. I made a call, the call was not answered, but my account was charged for the call, how come?
• A. Our server got "answer" signal from the remote end, that's why you were charged (and we were charged by call termination vendor too). We are not responsible for remote end equipment malfunction and do not refund charges for failed calls. Use another service (or another route with *3R prefix) to call that phone number. We appreciate if you report us this kind of problems and we'll remove the ill behaving route from the call path. Please let us know if the "false answer" problem is persistent or sporadical, copy/paste the call info from your call history to the message when contacting us. Getting rid of bad routes have the highest priority to us.