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Feature FAQ

  1. What does a OnSIP PSTN Gateway account come with?

    Your account will come standard with:

    • Inbound SIP and IAX Trunking: Allows any traditional telephone user to call your number. OnSIP transmits that call directly to your SIP or IAX device.
    • Outbound SIP and IAX Trunking: Allows you to configure your VOIP hardware or software in order to make calls to traditional phone numbers.

    Plus a whole lot more.

  2. What hardware or software does your service work with?

    You can use any SIP-enabled hardware or software or any IAX-enabled Hardware or Software such as Asterisk or an IAX ATA. We maintain a growing knowledgebase of configuration instructions and support for various hardware and software. We will do whatever we can to help you get up and running quickly.

  3. What protocols do you support?

    We currently support SIP and IAX.

  4. Is there a limit on the number of simultaneous callers I can have?

    No. You can have as many simultaneous callers as you would like.

    While many providers put a limit on the number of calls, we allow you to make or receive as many calls as you require at any given time.

  5. Do you support Asterisk?

    ABSOLUTELY! We use Asterisk extensively and are perfectly capable of working with your Asterisk implementation.

  6. Can you provide inbound DIDs?

    Yes. We can provide you with DIDs and inbound calling in most areas of the country.

    In the User Portal, you can put in a request for DIDs. We will fill your order as quickly as possible. This may take up to 10 business days and we cannot guarantee availability of the DID you request. You will only be charged if we provide you with a DID.

  7. How do I get charged for this service?

    All of the OnSIP services are pre-paid. As long as you maintain a positive pre-paid account balance, you can use any of the services. We will charge your account balance for each call or service used.

    You must maintain a positive account balance to use any of the services. You have multiple options to manually or automatically maintain your balance.

  8. Do you support 911, E911, or any emergency services?

    Yes. On our OnSIP Hosted PBX only.

    It is very important that you understand that our SIP and IAX Trunking connections do not support any emergency services dialing today such as 911 or E911.

    Customers of our SIP and IAX Trunking service will not be able to reach emergency service personnel.

    Junction Networks highly recommends that customer maintain an alternate means to reach 911, E911 or any emergency personnel.

  9. Do you provide call history?

    Your call details are available in real time in the User Portal.

  10. I live outside the United States, can I use OnSIP?

    Yes. However, we can only charge a valid US credit card. You can be plugged into an Internet connection anywhere in the world to use our service.

  11. Do you support faxing?

    We do not officially support faxing today. However, we plan to in the future.

  12. What happens if I move?

    Your OnSIP service will work anywhere there is a high-speed Internet connection. It will continue working normally if you move to a new location and plug the device into your Internet connection there. Please remember to update your billing information in the account center to reflect your new address.

  13. Do the people I call also need Junction Networks service?

    No. You can call any number that you could have otherwise dialed with a traditional phone.

  14. Can I use the phone over a dial-up connection?

    OnSIP service is designed for a high-speed Internet connection, not dial-up. You may be able to use a dial-up connection but OnSIP does not recommend anything other than a high-speed Internet connection.

  15. What is DID Routing?

    DID Routing provides optional extended service features including contingency failover, routing calls to static IP addresses, routing calls utilizing multiple protocols, routing calls back into the PSTN, and adjustable timeouts.

    DID Routing is not required to utilize DIDs you have requested via the Junction Networks User Center or DIDs which are otherwise associated with your account (such as via LNP). You can receive calls (call origination) on all your DIDs simply by registering with and utilizing the protocol associated with your DIDs. Likewise, DID Routing is not required to place calls (call termination).

    DID Failover Routing Service - Setup: $12.50 Monthly: $18.95

    Upon confirming your request, we immediately enable your account for DID Routing and your balance will be debited. While enabled, you will be billed automatically every month for this service.

    For $18.95 per month (total for all your numbers) and $12.50 one-time setup we can setup your numbers to fail over to another number e.g. a cell phone or a land line.

    If used as a fail-over/disaster recovery option, the following Failover Scenarios will trigger the fail-over routing:

    Failover Scenarios:
    1.) Your device is not registered at all. There is therefore nowhere to send the INVITE. We fail over immediately.
    2.) 5 SIP or IAX INVITE requests - 1 per second for 5 seconds - without a return response. Fail-over after 5 seconds.
    3.) Rejected INVITE. If you send back "congestion" or otherwise respond to the INVITE but respond in the negative, we fail over immediately upon receipt of the negative response.
    4.) If you have a SIP device and it does not pick up the call after x seconds (the seconds are up to you with a default of 120), we fail over after x seconds of ring-no-answer.

    Setup:

    • Group Limit - max number of simultanious calls to this number
    • Registration Protocol - the session protocol used to register contact information
      for this DID.
    • No Answer Timeout - the number of seconds an attempt is made to reach the registered contact.
      • If the timeout is set to '0' or there is no contact registered, no
        attempt will be made to reach a registered contact and the failover
        location will be attempted immediately.
      • If there is a contact registered, five attempts over five seconds
        (one per second) will be made to reach the contact. If the contact
        fails to respond to all five attempts, the failover location will then
        be attempted. That is, failover occurs after 5 seconds in the event
        that no registered contact responds.
      • If a registered contact is reached but signals congestion,
        unavailable, error, or otherwise rejects the call, the failover
        location will be attempted immediately.
      • If a registered contact is reached and signals ringing, the call
        will be allowed to ring until timeout seconds elapses or the call is
        answered. If timeout occurs before the call is answered, the call will
        be hung up and the failover location will be attempted.
      • If a registered contact is reached and signals busy, no failover attempt will be made.
      • No failover is attempted is the call is answered.
    • Failover Protocol - the session protocol used to reach the failover location
      for this DID.
    • Failover Location - the location to contact in the event of failure.
      • If failover protocol is 'PSTN', a valid 11 digit phone number.
      • If failover protocol is 'SIP', a SIP address of the form
        [user@]ip-address[:portno]
      • If failover protocol is 'IAX', an IAX2 address of the form
        [user@]ip-address[:portno][/exten]

      Note that host names cannot be used in lieu of an IP address.

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