Example of use outgoing SIP calls service
(read on how to accept SIP calls here)
    As a Google Talk user you can use our voip gateway to route your calls to any SIP phones (like SJPhone, Gizmo Project or any other). You can do it following by these steps:

    Step 1. Subscribe to the service in one of the possible ways:

    • 1st Way. Open your GoogleTalk and invite new recipient whose user id is service@gtalk2voip.com.
    • 2nd Way. Go to the main page of this site and submit your user id by pressing "Invite" button, then accept invitation from service@gtalk2voip.com.

    Step 2. Type in a command like CALL someuser@sipprovider.com in the chat window to place a call to SIP user someuser@sipprovider.com. You may also define port number like in someuser@sipprovider.com:5060.

    Step 3. In a second or two you will get an incoming call from service@gtalk2voip.com which you have to accept (or reject if you have suddenly changed your mind).

    Step 4. When incoming call is accepted you will hear a beep signal followed by (a usual voice indicator of call progress). Text messages also displayed in chat window informing you about call setup events, call cost and service provider it is routed through.

    Step 5. When your call is answered on the remote side, "Remote answered. Now talking.." message is displayed. From this moment accounting process is started, you can talk.

    Step 6. When you drop the call or the remote party hangs up, a message like "Charged time: XX secs. Reason = EndedByLocalUser" is displayed to you. Reason is a sort of mnemo-code indicating what was the reason of call disconnection. EndedByLocalUser means that whas you who ended the call.

    Step 7. If some network problem occurs, or SIP user has been wrongly defined, lots of other call disconnection reasons displayed.

    Here is an example on how to call Gizmo Project user by user-id:

    You: CALL ekorolenko@gizmoproject.com
    service: You entered: CALL ekorolenko@gizmoproject.com 
             You have 0.6055 USD.
             Please, accept call from gtalk2voip !!!
    Incoming call from service@gtalk2voip.com at 19:18
    service: Thank you! Calling SIP phone: ekorolenko@gizmoproject.com .
             Call in progress...
             Remote answered. Now talking...
    Call ended with service@gtalk2voip.com at 19:18
    service: Thank you!
             Charged time: 16 secs. Reason = EndedByLocalUser
    	

    Here is an example on how to call Gizmo Project user by Gizmo number:

    You: CALL 17476442775@gizmoproject.com
    service: You entered: call 17476442775@gizmoproject.com 
             You have 0.6055 USD.
             Please, accept call from gtalk2voip !!! 
    Incoming call from service@gtalk2voip.com at 19:26 
             Thank you! Calling SIP phone: 17476442775@gizmoproject.com .
             Call in progress...
             Remote answered. Now talking...
    Call ended with service@gtalk2voip.com at 19:26
    service: Thank you!
    Charged time: 5 secs. Reason = EndedByLocalUser
    	

    Please note...

    • You have to keep your balance positive to be able to make calls.
    • To know the remote SIP phone IP address the following procedure is used: First, the gateway does a DNS request for SRV _sip._udp record which is used. If no SRV record available, a an ordinary domain name to IP address translation is done.
    • Default SIP port is 5060 or the one defined in _sip._udp SRV record.
    • Billing is carried on per-second basis.
    • The call is not charged if the remote does not pick up the phone (no answer).
    • This service is not expected to be used for emergency calls.
    • This service currently is FREE of charge!


    Have a nice call!

Send all your questions and ideas for further service improvements to GTalk2VoIP TEAM. We would certainly like to hear from y ou!

If you've tested and liked our services, please let your friends know about it, send them a voice mail. Thank you.

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