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 .
• 2nd Way. Go to the main page of this site and submit your user id by pressing "Invite" button, then accept invitation from .

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

Step 3. In a second or two you will get an incoming call from 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 
service: You entered: CALL  
         You have 0.6055 USD.
         Please, accept call from gtalk2voip !!!
Incoming call from  at 19:18
service: Thank you! Calling SIP phone:  .
         Call in progress...
         Remote answered. Now talking...
Call ended with  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 
service: You entered: call  
         You have 0.6055 USD.
         Please, accept call from gtalk2voip !!! 
Incoming call from  at 19:26 
         Thank you! Calling SIP phone:  .
         Call in progress...
         Remote answered. Now talking...
Call ended with  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!