SIP Commander. Version 1.0
Download free version (1Mb)
SIP 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.
SIP 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 SIP 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 SIP 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, SIP 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.
MGCP/Megaco - The Basics
MGCP Megaco The Basics
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MGCP/Megaco - The Basics
When a gateway detects an off hook condition, it tells the gateway controller, which might respond with a command to instruct the gateway to put dial tone on the line and listen for DTMF tones indicating the dialled number. After detecting the number, the gateway controller determines how to route the call and, using an inter-gateway signalling protocol such as SIP, H.323, or Q.BICC, contacts the terminating controller. The terminating controller could instruct the appropriate gateway to ring the dialled line. When the gateway detects the dialled line is off hook, both gateways could be instructed by their respective gateway controllers to establish two-way voice across the data network. Thus, these protocols have ways to detect conditions on endpoints and notify the gateway controller of their occurrence; place signals (such as dial tone) on the line; and create media streams between endpoints on the gateway and the data network, such as RTP streams.
There are two basic constructs in MGCP/Megaco: terminations and contexts. Terminations represent streams entering or leaving the gateway (for example, analogue telephone lines, RTP streams, or MP3 streams). Terminations have properties, such as the maximum size of a jitter buffer, which can be inspected and modified by the gateway controller. A termination is given a name, or TerminationID, by the gateway. Some terminations, which typically represent ports on the gateway, such as analogue loops or DS0s, are instantiated by the gateway when it boots and remain active all the time. Other terminations are created when they are needed, get used, and then are released. Such terminations are called "ephemerals" and are used to represent flows on the packet network, such as an RTP stream.
Terminations may be placed into contexts, which are defined as when two or more termination streams are mixed and connected together. The normal, "active" context might have a physical termination (say, one DS0 in an E3) and one ephemeral one (the RTP stream connecting the gateway to the network). Contexts are created and released by the gateway under command of the gateway controller. Once created, a context is given a name (ContextID), and can have terminations added and removed from it. A context is created by adding the first termination, and it is released by removing the last termination.
MGCP/Megaco uses a series of commands to manipulate terminations, contexts, events, and signals: