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Dialing a number
Phone numbers (DIDs)
Hardware
FAX
Caller ID
Voice mail
VPN
API
Billing
Troubleshooter

Dialing a number
  • 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. Set "Dial Rule" to "North America" and specify your area code.
  • 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 or set "Dial rule" to "North America".
  • Q. My friend’s telephone number in Paris, 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. Yes if you set "Dial rule" to "North America".
  • Q. My friend is your customer too, how can I call her?
  • A. Dial 000 followed by her 9-digit long username. Calls between CallWithUs customers are always free (but account balance must be positive). STUN support must be enabled in both SIP clients or you will get audio problems. To verify if STUN support is working properly in your SIP client, check "VoIP accounts" menu, registration status of your SIP client must be marked as "no NAT".
  • Q. What is the difference between standard, premuim and PSTN routes?
  • A. The main difference is in quality and reliability. PSTN routes are direct to service providers and deliver CLI to all supported destinations, but are not A-Z type, some destinations may be unreachable using PSTN routes. Standard plan uses all available routes regardless of the quality rating. Premium routes are a good A-Z compromise. You can switch your default tariff plan in "user settings" menu on our web site and check call rates in simulator. Standard routes have the lowest call rates, rates on premium and PSTN routes are usually higher.
  • Special numbers to dial from SIP phone (require dialplan in SIP client to be configured to accept numbers which begin with *):
    • *30NNNNNNNNNNN - use PSTN route.
    • *31NNNNNNNNNNN - use Standard route.
    • *32NNNNNNNNNNN - use Premium route.
    • *4TTTNNNNNNNNNNN - call number NNNNNNNNNNN using call termination trunk TTT, the trunk numbers and call termination rates are shown in simulator when you logged into your account. If the trunk number has 2 digits only, prepend a 0 to the number. If the default route to the called number does not work properly, try the next trunk to make a call. Example - *40709193NNNNNNNN - call India mobile number using call termination trunk 70. The 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 or 225 (BAL) - check account balance (free).
    • *1 - redial last dialed number(regular calling rates apply).
    • 3246 (ECHO) - echo test (free).
  • What are the meanings of the "terminate cause" column values in the call history?
    • ANSWER - the call has been answered by the called party.
    • CANCEL - the caller did hang up before the called party answered the call.
    • CONGESTION - the call can't be delivered to the called party. Usually means call to an invalid or not in service number
    • NOANSWER - the call was not answered and timed out.
    • BUSY - the called party responded with SIP BUSY message.
    • OVER_MAX_RATE - call rate to the dialed number is higher than "max rate" parameter value in "user settings" menu in customer portal.
    • CHANUNAVAIL - incoming call failed because your SIP client is not registered with our server.
    • NO_ROUTES - we have no routes to the dialed number.
    • INVALID CID - you present an invalid (or disallowed) caller id number.
    • Invalid number - you're trying to call an invalid phone number.
    • Blocked by ASR - the ASR of your traffic is too low, we require the ASR to be more than 50%. Your account is temporarily suspended for 24 hours.
    • No Dest - we received incoming call to a DID, but the DID destination is not set.

Phone numbers (DIDs)
  • Q. I live in Munich, Germany. Can I order a UK phone number?
  • A. Yes. See "Available DIDs".
  • Q. If I purchase a DID in some country and the DID includes N free minutes, does that 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 does the DID parameter "channels" mean?
  • A. The channels parameter is the maximum number of simultaneous incoming calls the DID supports. To implement "call waiting" service, the DID has 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 to you server via SIP?
  • A. Yes. If you want to redirect the DID to ring your SIP client registered with our server, set the DID calls forward with your DID provider to username@did.callwithus.com, where username is your 9 digits username. Your SIP client must have STUN enabled or be on a public IP address, we do not handle media and NAT for DIDs from a third party. 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 or handle NAT issues by our server, we need to know the DID number to enter it to our system as a DID with $0.5 monthly price and $2 setup fee. Set the DID call forwarding with your provider to didnumber@did.callwithus.com that case.
    Your account balance must be positive to receive incoming calls.
  • Q. How do I specify the DID destination?
  • A. If the destination field should contain SIP URI, prefix it with "SIP/". If the SIP/ prefix is missed, 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 in 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, the DID forward to that phone will work as an alarm indicating that something is wrong with your VoIP destination(s) setup.
    Here are examples of DID destination:
    • SIP/username - call your SIP client registered with our setver.
    • SIP/username/extension - call your SIP client registered with our server. The call will go to the extension "extension" in your dial plan.
    • SIP/extension@your_ip_address - call "extension" on SIP server at specifier IP address. No SIP registration is required to receive incoming calls.
    • SIP/12345@fwd.pulver.com - forward DID call to external SIP URI (your FWD account).
    • Please note that when DID call is forwarded to external SIP URI, calls will come to your system from either east.callwithus.com or west.callwithus.com, your system should be configured to accept incoming calls from these hosts. If you run asterisk or trixbox, create 2 incoming trunks for these hosts. Here is an example: 
    
[east]
type=peer
host=east.callwithus.com
context=from-trunk
insecure=invite,port

[west]
type=peer
host=west.callwithus.com
context=from-trunk
insecure=invite,port
    Please note - if you forward a DID to a SIP URI, we assume that your SIP server is not behind a NAT router and can handle direct media. Our server does not perform NAT handling this case.
  • 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. As soon as one device answers the call, the another one(s) will stop ringing.
  • Q. Is it possible to ring my SIP device and my cell phone on incoming DID call?
  • A. Yes, set the DID destination to SIP/111111111&1XXXXXXXXXXX, where 1XXXXXXXXXXX is your cell phone number, note the & sign. 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. Where the DID server is located?
  • A. By default DID calls are processed by a server in North America. If you're located in EMEA region and your SIP client is behind a NAT router, we suggest to contact us to request routing DID calls through our European server. No configuration changes are needed on your end. You can request the routing change for all your DIDs or for individual DIDs only.
  • Q. How can I check if the DID number I am going to purchase works OK?
  • A. Dial the number from your land line or cell/mobile phone, you should hear recorded message in English "The number you dialed is currently not in service, please check the number and dial again". 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.

Hardware
  • Q. What kind of telephone adapters can I use with your service?
  • A. You can use any telephone adapter which supports SIP protocol, like Obihai OBi100, OBi110, Sipura 1001, 2002, 3000, Linksys PAP2, 3102, Grandstream IP phones and adapters. Note: Skype phones and adapters will not work with our service.
  • Q. Which codecs do you support?
  • A. We support g711u, g729 and GSM codecs.
  • Q. How can I check whether 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 the 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. You can use the same username/password configured in multiple devices, or 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. Do you proxy media?
  • A. CWU server automatically detects if your SIP device is on a public IP address or NAT traversal solution like STUN is employed in your SIP client, and sets direct audio path with call termination gateways, otherwise audio will be proxied by one of our servers closest to your geographical location.
  • Q. My Linksys (or Sipura) ATA loses its dial tone and I have to reboot it to get it working again! How can I fix this problem?
  • A. In the 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. Do you support SIP/TCP?
  • A. Yes, sip.callwithus.com supports SIP over TCP and TLS.
  • Q. Do you support IPv6?
  • A. Yes, sip.callwithus.com provides experimental support for IPv6 protocol.
  • Q. Do you provide SIP trunking?
  • A1(Marketing BS is off). What is SIP trunking?
  • A2(Marketing BS is on). Yes, we do provide SIP trunking and virtual PRI/DC*/T1/E1/J1 circuits etc.
  • Q. Is it possible to authenticate calls by IP address instead of SIP username and password?
  • A. Yes, in VoIP account settings in customer portal set "Host" field to your host or network IP address to accept calls from. It could take up to 15 minutes to activate the setting update.

FAX
  • Q. Do you provide fax over VoIP?
  • A. Fax doesn‘t work reliably over VoIP (including T.38), it‘s a hit or miss. It may work but may not. Give it a try, but we will not provide fax-related support.

Caller ID
  • 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 does "activated" Caller ID in the "Add Caller ID" menu mean?
  • A. Caller ID numbers in "Add caller id" menu work for authentication if you use our access number/callback features and to set the Caller ID number when you make outgoing calls. 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.
  • 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. I set caller id number in VoIP account settings, but called party see my CWU username instead of the number.
  • A. If you have Linksys/Sipura ATA, then set in Admin/Advanced/Line X tab "SIP Remote-Party-ID" to "no". For OBi devices uncheck "X_InsertRemotePartyID" in ITSP profile X/SIP.

Voice mail
  • Q. How do I access my voice mail box?
  • A. Dial 086 (0VM) or 311 from your SIP 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 the 1st of every month thereafter. Remove the VM box at any time to stop monthly charges (all voice mail messages and recorded greetings will be removed).

VPN
  • Q. Can I use my ATA with VPN connection or am I 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 on ATA.
      On the 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 (the 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.

API
  • Q. I run asterisk, can I access LRN and CNAM API from asterisk dialplan?
  • A. Sure, here is an example of CNAM lookup from asterisk dialplan:

    exten => _X.,1,Set(CALLERID(name)=${CURL(http://lrn.callwithus.com/api/cnam/index.php,key=apikey&number=1${CALLERID(num)})})

Billing
  • 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 a new account, we delay the account funding to prevent service fraud. Your account will be changed to immediate payment processing after 2 months.
  • Q. What is a billing unit to call a PSTN number?
  • A. 60 seconds.
  • Q. Is there a service cancellation fee?
  • A. No. We will refund your remaining account balance in full on your request. But 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. If you made the last payment more than 6 months ago, the only refund option is sending money to your paypal account.
  • Q. Do you offer calling credit?
  • A. No. Our service is prepaid only.
  • Q. Which payment methods do you accept?
  • A. We accept the following payment methods:
    • Paypal (from bank account, credit/debit card, masspay).
    • Credit/debit card.
    • Bitcoin.
    • Bank wire transfer or ACH payment.
    • Paypal and credit/debit card payments are not accepted from certain countries, the only available options are bitcoin and bank wire transfer.
  • Q. What is the best way to not miss a monthly payment for my DID (and not loose the number!)?
  • A. Set the "Alert Balance" in "User settings" menu to be higher than the monthly DID price. Our system 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 enter 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 (up to "Max Route" parameter in "User settings" menu). We do the best to complete your call using the least cost route. However, you can select a route to a destination which works the best to you, see *4TTT dial prefix above.
  • Q. Can I prevent accidental calls to destinations with a high call rate?
  • A. Yes. 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.
    You can override per call "max route" and "max rate" values in your account settings with custom SIP headers in INVITE request.
    • X-CWU-maxrate: N.NNNN - set max call rate to $N.NNNN, we will try to deliver the call using routes with call termination rate to the dialed number no higher than $N.NNNN.
    • X-CWU-maxroute: N - set "max route" parameter to N, we will try up to N routes to deliver the call, the "max rate" parameter set in your account settings will be used for call routing decision too.

    Examples of usage:

    Asterisk: 

    exten => _X.,n,SIPAddHeader(X-CWU-maxroute: 2)

    FREESwitch: 

    <action application="set" data="sip_h_X-CWU-maxrate=0.028"/>
  • Q. I have multiple accounts with you. Can I transfer funds from one account to another?
  • A. No, we do not transfer funds between accounts.
  • Q. Do you send invoces?
  • A. Login into your account, select "Invoices" menu, in the page header set the desired date range, select "Export PDF" radio button and click on the "Search" button to generate the invoice.
  • Q. I made a call, the call was not answered, but my account was charged for the call, how come?
  • A. This is a "false answer". 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 the remote end equipment malfunction and do not refund charges for failed calls. Use another trunk with *4TTT prefix to call that phone number. We appreciate if you report this kind of problem and we will work with the call termination carrier or remove the ill behaving route from the call path. Please let us know if the "false answer" problem is persistent or sporadical, if call goes through OK with another trunk, 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.
Fine print: All prices are final, there are no bogus fees and unfees other than specified at Services page. Period. Only SIP devices that have already been created can be connected to sip.callwithus.com to make calls. Please ensure you only use devices approved by you (Please do not try and connect using two tin cans and a piece of string as we do not yet support this, but we may support this in the future, the work is in progress and preliminary results are positive). Callwithus.com monthly subscription charge of $0 must be paid in advance and does not include tax of $0 which also must be paid in advance. You will be billed an activation fee of $0 plus tax and this must be paid in advance. Calls made incur tax at the rate of 0% each month and must be paid in advance. On cancellation of the service you will be charged a one time disconnection charge of $0. Additional features will be billed at the additional rate of $0 per call. All **YOUR** rights reserved.

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