What known SIP hardware and software phones are supported ?

    Virtually any SIP device or a SIP softphone should work fine with our SIP2VoIP service. If you have a SIP device that you believe is not compatible, please email us, so we could investigate. Here below is incomplete list of supported software and devices.

    • FreeSWITCH
    • Asterisk

What codecs are supported ?

    Currently supported codecs are: G.729, G.711u, G.711a, GSM, iLBC. It is strongly recommended that you configure your hardware ot software phone to use G.729, otherwise you may get very poor voice quality.

How do I know whether my SIP device is registered on SIP2VoIP proxy ?

    You can check your active SIP registrations on your personal account page. To get access to your page you have to sign in using your SIP username and password.

How calls are being routed ?

    In our SIP2VoIP infrastructure (same as in GTalk2VoIP service) we have many different VoIP providers which can carry calls to different destination phone numbers, each with its own communication rates, latency and voice clarity. Our system automatically calculates integrated quality parameter for each provider and uses this parameter to rank providers in such a way that best quality routes go higher and are used more frequently. It is understood that voice quality and price do not often correlate, but experience shows that cheaper routes are often give poorer quality. To provide our customers with better experience our system is programmed to choose best qaulity route by default. In other words, when you setup a call, our system retrieves a list of routes (providers) and starts pushing your call through them beginning with the best ranked provider till the call reaches its destination. Important notes and consequences:
    • In most cases, not the cheapest route/provider will be chosen.
    • Providers can change their rates without notificatins, thus leading to different charges made for calls to same destination number.
    • Ranking system always re-arranges the top list of providers, so your two consequent calls can be made through two different routes, thus leading to different call rate.
    • Providers may be get temporary disabled or go out of order without any notifications. In such a case, our system will use the other best ranked route to deliver your calls. This also leads to different call rates.
    The described above default routing algorithm can be overriden by using Dialing Plan feature.

What is Dialing Plan and how to configure it ?

    In our SIP2VoIP service (same way as in GTalk2VoIP service) we allow our customers to select routes/providers that are used to deliver your calls thus overriding default Routing scheme (see above). Basically, we allow you to setup your own routing scheme called Dialing Plan. Dialing Plan feature is available on your personal account page. It consists of one of more lines called Rules. Each rule can be associated, on one hand with a phone number or a destination (leading part of number) called Prefix, on the other hand - with a Provider which will be used to route calls to this particular destination. You can setup more than one rule with same destination (Prefix) but with different providers associated, you can prioritize providers one upon onther by defining Order value.

    Dialing Plan cna be also used to translate dialed phone numbers, which is useful when you want to dial international numbers like a local phone numbers. I.e. you can prepend dialed number with digits or strip any number of digits.

How to setup FreeSWITCH to register and make calls through SIP2VoIP ?

    You have to alter your FreeSWITCH configuration as per this sample. Please do not forget to change YourUsername and YourPassword to your username and password obtained from SIP2VoIP.

How to setup Nokia N8?

    The settings can be found in Settings -> Connectivity -> Admin. Settings -> Net Settings -> Advanced VOIP settings. You first define the SIP settings, then define a service using those settings.

How to setup X-Lite?

    Please refer to these instructions: http://www.cafesip.org/projects/jiplet/xlite-settings/x-lite-settings.html

    Then use the following settings:

    Display name: your_username
    User name: your_username
    Password: your_password
    Authorization username: your_username
    Domain: sip2voip.com

    Check "Register with domain and receive incoming calls"
    Set to Proxy, the define proxy address: sip.sip2voip.com

How to setup Nexus S or other Andoird based phone?
    Step 1. Open Settings and tap on Call Settings.
    Step 2. Under Internet Call Settings, tap on Accounts.
    Step 3. Now, add an account by punching in your SIP account details:
      pwd: your_password
      server: sip.sip2voip.com
      authentication username: your_username
      outbound proxy address: sip.sip2voip.com
      port number: 5060
      transport type: udp
    Step 4. Save the credentials and go back to the Accounts Settings.

How to setup Linphone?
    Go to Menu->Linphone->Settings and fill in below setting as shown on this pircure

    Step 1. Select tab "Manage SIP Accounts"

  • Your display name: your_username
  • Your username: your_username

    Step 2. Press "Add" button on "Proxy accounts"

  • Fill form "Configure a SIP account"
  • Your SIP identity: sip:your_username@sip.sip2voip.com
  • SIP Proxy address: sip:sip.sip2voip.com

    Press "OK"

    Step 3. Select tab "Codecs"
    Disable all codecs except Speex 8000

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