Phone Commander. Version 1.0
Download free version (1Mb)
Phone Commander is a free portable softphone that you can use to make and receive VoIP phone calls from your PC, iPhone or Android based smartphone.
Phone Commander like a telephone to let you make calls through your computer. Call anyone via the internet who also has a softphone installed and if you sign up with a VoIP gateway service company you call regular telephone numbers as well.
The advantage of using Phone Commander is that you can leverage low cost or free VoIP calls and you can connect to the company SIP PBX and work remotely.
Easy to use and powerful Windows program was designed for people who want to improve dynamic interactions with contacts and to manage contact information. It supports multiple SIP accounts and save calls history.
See all contact information for incoming calls. For example, you can see Firstname, Lastname, Birhdate, Company name and other information before call answering. Keep detailed records and notes in your call log about each call and contact.
You can use Phone Commander as a customer database, or personal contact address/phone book, working with groups for managing contact info of individuals and organizations with relationships. Phone numbers, emails, web pages, faxes, pagers, addresses, customer notes - you can save all this data in an organized format.
You can dial phone numbers in one click via Internet. Program has a simple intuitive interface and quick and easy contact lookup. Import and export features are also available.
Attractive and easy-to-use organizer & PIM will keep track of your contacts, addresses, distribution lists, manage your schedule, remind about appointments, and keep your daily notes in order, store your contacts and call by one click. The slick user interface makes it a snap to find addresses and phone numbers, enter reminders.
With its intuitive & familiar interface, users can seamlessly transition from a traditional hard phone environment into the world of Voice over IP. Also by making the navigation simple and user friendly, Phone Commander provides users with easy access to address book management.
The program detects DTMF user input, sends DTMF (Inbound, SIP INFO, RFC2833), supports G.721 A-law/Mu-law, GSM.610, Speex. You can select audio In/Out Devices.
What Is SIP? Signaling
What Is SIP signalling signaling Introduction About SIP Overview Basics
> ABOUT SIP > WHAT IS SIP > SIP SIGNALINGSIP Signaling
SIP Signaling
SIP is based on the request-response paradigm. The following sequence is a simple example of a call set-up procedure:
1. To initiate a session, the caller (or User Agent Client) sends a request with the SIP URL of the called party.
2. If the client knows the location of the other party it can send the request directly to their IP address; if not, the client can send it to a locally configured SIP network server.
3. The server will attempt to resolve the called user's location and send the request to them. There are many ways it can do this, such as searching the DNS or accessing databases. Alternatively, the server may be a redirect server that may return the called user location to the calling client for it to try directly. During the course of locating a user, one SIP network server can proxy or redirect the call to additional servers until it arrives at one that definitely knows the IP address where the called user can be found.
4. Once found, the request is sent to the user and then several options arise. In the simplest case, the user's telephony client receives the request, that is, the user's phone rings. If the user takes the call, the client responds to the invitation with the designated capabilities* of the client software and a connection is established. If the user declines the call, the session can be redirected to a voice mail server or to another user.
* "Designated capabilities" refers to the functions that the user wants to invoke. The client software might support videoconferencing, for example, but the user may only want to use audio conferencing. Regardless, the user can always add functions - such as videoconferencing, white-boarding, or a third user - by issuing another invite request to other users on the link.
SIP has two additional significant features. The first is a stateful SIP server's ability to split or "fork" an incoming call so that several extensions can be rung at once. The first extension to answer takes the call. This feature is handy if a user is working between two locations (a lab and an office, for example), or where someone is ringing both a boss and their secretary.
The second significant feature is SIP's unique ability to return different media types within a single session. For example, a customer could call a travel agent, view video clips of possible holiday destinations, complete an on-line booking form and order currency - all within the same communication session.
SIP Methods
The commands that SIP uses are called methods. SIP defines the following methods:
| SIP Method | Description | INVITE | Invites a user to a call | ACK | Used to facilitate reliable message exchange for INVITEs | BYE | Terminates a connection between users or declines a call | CANCEL | Terminates a request, or search, for a user | OPTIONS | Solicits information about a server's capabilities | REGISTER | Registers a user's current location | INFO | Used for mid-session signalling SIP responses The following are SIP responses:
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