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Picture 6 Gatekeepers The subwindow Logs shows all the major events and alarms in Voipswitch. Importing tariffs It is possible to import the tariff rates from a csv or txt file. Pc2phone is a proprietary application that allows clients who has a valid pc2phone client account to connect to the Voipswitch and initiate and also receive calls. Connection properties section: 25 Supported codecs Allows the selection of 4 codec groups depending on what client device can support. Then we click Add new payment button. You can right click each registered user that appear in this window and see the current settings for each. To add a new rate you will need to fill the fields and click Add.
Telephone number field is the prefix of the telephone number to route. Picture 3 calls submenu The subwindow Users shows the pc2phone and web2phone users and also the Callshop users. If you will have to send calls to termination gw in direct mode then create an account in Gateways. IP number is the IP address of remote terminating gw. Greater Kailash falls in category B and therefore the circle rate applicable here is Rs.
Therefore when the system will ask for PIN it will compare the received number with the password of an IVRclient account. For example termination gateways require often receiving the dialed numbers with a tech prefix for authentication. The next step is to create a connection with the database. H323ID can be required by your termination carrier to be sent for authentication. For example if Voipswitch rejects a user you will see in the Logs window what was received from the user and therefore you can deduct why the user was rejected wrong user name or any other reason. Voice rate is the price per minute.
Under the IP numbers list there is a field where to write the new addresses to be added in the list. Like: 33 IE Display and IE Calling party number are 2 fields from H323 protocol that refers to Caller ID information. Proxy modes are set in dialing plan menu. After running VPSConfig, database settings window will open and create connection string button has to be chosen. Gateway description is a label for the terminating gateway. The team consistently delivers excellent technology solutions that enable our company to quickly adapt to evolving customer expectations and provide a great retail experience. After the proper creation of the database structure and connecting with it, menu on the left side of VPSConfig window will become active.
For example we want to route all calls from client A through destination gateway G and all calls from client B, to same country, through gateway H. In proxy all VoipSwitch does not allow codec negotiation directly between endpoints and instead will negotiate itself with each endpoint in part. VoipSwitch Config management interface. GatewaysIn this section you have to define the termination gateways where you will send the calls. Here you can decide where to send a call depending on the prefix of the dialed number. It is possible to carry out connections between H323 devices and SIP.
For example if you set Minimal duration 30 and Resolution 6 each call will be charged for a minimum of 30 seconds even if it lasted 2 seconds. Calls limit sets a limit of maximum simultaneous calls that Voipswitch is allowed to send to this terminating gateway. String Calling from the SIP equipment 42 Equipment based on the SIP protocol can connect through the switch in the two following modes: SIP proxy SIP registrar For the clients calling in the SIP proxy mode, accounts should be set up in the GW Clients menu. If your dialplan rule was for a telephone number 1214 for example and the termination gateway needs to receive a tech prefix like 444 plus 0 then the Dialed number helping window will show like this: As you see the original telephone number 1214 will be changed into 444 01214 and then will be sent to the termination gw. Proxy settings - Proxy all, connect independently: origination and termination endpoints do not see each other, the VPS connects independently with each endpoint then conference them together. Switch performs conversion from h323 to SIP and from SIP to H323. And this is even better because the client can dial either with 00 or without 00 while this replacement rule will cut only if 00 exist at the beginning of number.
The PIN number entered will be matched against the password field of an IVRclient. I strongly recommend looking at the GK platform, especially if you are an SAP shop looking for seamless integration. Also if you want to charge all calls with same value here you can set Surcharge time zero and Surcharge amount the value you want. You can set here an unlimited number of addresses but an IP can be enabled only for one GWClient at a time. The connection is carried out in proxy all mode.
We are much more efficient than in the past, especially regarding country-independent functional extensions and in the operation of our POS systems. If not required is safe to be left blank. Quality, Simplicity and Savings are the guiding principles of our business. Set them according to the terminating carrier requests or left them bank. If client has already an active call, all other attempts to connect will be rejected.
So the two endpoints can use more codecs if they both support them even if Voipswitch do not support it. After completing all the above steps the information confirming successful creating of Voipswitch database should appear. Our People Our people are at the centre of what we do. GWclients can be either h323 or SIP. Otherwise all the fields have same meanings as before. Gatekeeper h323 : H323 ID, e164, GK name, FastStart consult your carrier about these settings.
All the dialed numbers without matching prefixes in tariff table will be rejected. Rules for modifying clients data This field is complex and allows modifying different call settings last time before the call is sent to the termination gateway. From the list of drivers Microsoft OLEDB provider for ODBC drivers has to be chosen. The current connected users will appear in blue and the disconnected users in red. VoipSwitch will transmit the calls only if the user is logged.