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.
Newsgroups Working Groups and Mailing Lists
News Newsgroups Working Groups and Mailing Lists
> ABOUT SIP > LINKS > NEWS AND MAILING LISTSNews and Mailing Lists
News and Mailing Lists
Note - please adhere to standard mailing list protocol: make sure you check archives before stumbling into proceedings.
If you are aware of any other links we should include then please let us know
Mailing lists
The two major SIP mailing lists are:
Columbia Universities' SIP implementers' mailing list This is a mailing list for the discussion of implementations and to arrange interoperability testing. The Columbia list does deal with some basic questions. Send a mail to sip-implementors-request@cs.columbia.edu. with "subscribe-sip-implementors" in the subject.
SIP forum discussion mailing list - Membership in the SIP Forum is made up of individuals, network engineers, application developers and other Internet professionals who want to contribute to the development of global Internet communications based on IETF and W3C standards.
SIP Forum Marketing Working Group
SIP Forum Service Provider Working Group
SIP Forum Certification Working Group
SIP Forum SIP Devices Working Group
SIP and General Telecoms news
The SIP Center News page Daily updates of SIP headlines and press releases
ConferencingNews.com - The Conferencing Industry Portal: Audio - Web - Video
IPtelephony.org Voice and Fax over IP news and market research. IP xStream is an ongoing market research study conducted by Gecko Research and Publishing. It's IP Pulse is a weekly publication tracking the VoIP industry.
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Mailing lists
The two major SIP mailing lists are:
Columbia Universities' SIP implementers' mailing list This is a mailing list for the discussion of implementations and to arrange interoperability testing. The Columbia list does deal with some basic questions. Send a mail to sip-implementors-request@cs.columbia.edu. with "subscribe-sip-implementors" in the subject.
SIP forum discussion mailing list - Membership in the SIP Forum is made up of individuals, network engineers, application developers and other Internet professionals who want to contribute to the development of global Internet communications based on IETF and W3C standards.
SIP Forum Marketing Working Group
SIP Forum Service Provider Working Group
SIP Forum Certification Working Group
SIP Forum SIP Devices Working Group
SIP and General Telecoms news
The SIP Center News page Daily updates of SIP headlines and press releases
ConferencingNews.com - The Conferencing Industry Portal: Audio - Web - Video
IPtelephony.org Voice and Fax over IP news and market research. IP xStream is an ongoing market research study conducted by Gecko Research and Publishing. It's IP Pulse is a weekly publication tracking the VoIP industry.
Number Resolution
Number resolution - ENUM and TRIP
The system of telephone numbers is well defined and understood. There are billions of telephones, and billions of telephone numbers, each one no larger than 15 digits and globally unique. Alongside this system many people will be using SIP phones and SIP clients on their PCs and these will have SIP URLs associated with them. Fortunately, SIP can carry phone numbers using the new telephone URL (e.g. tel:208534845).
ENUM
A problem arises when one needs to associate a traditional phone with a resource on the Internet (this would be a requirement of any converged presence-type service). The IETF's ENUM (Telephone Number Resolution) working group is devising a scheme to enter telephone numbers onto the Internet DNS so that any application, including SIP, can discover resources available to a globally unique phone number. A SIP phone or proxy server would do the number domain translation and through classic DNS resolution discover a DNS resource that would give a SIP address at which the dialled number could be reached.
Further resources:
TRIP
Where a telephone number does not have a SIP resource associated with it, the IP network routes a call to a telephone gateway, which connects to the PSTN. In an interconnect environment with many peering relationships between service providers, resources in the IP network need to be able to discover which telephone numbers are associated with which gateways.
The IPTEL working group is developing TRIP (Telephony Routing overIP), a policy driven inter-administrative domain protocol for advertising the reachability of telephony destinations between location servers, and for advertising attributes of the routes to those destinations. TRIP is designed to allow service providers to exchange routing information in order to avoid the over-provisioning or duplication of gateways. It uses established Internet protocols such as BGP, OSPF and SCSP.
For further information see the Whitepaper - Provisioning ENUM Services (NetNumber ©copyright December 2000)
TRIP
Where a telephone number does not have a SIP resource associated with it, the IP network routes a call to a telephone gateway, which connects to the PSTN. In an interconnect environment with many peering relationships between service providers, resources in the IP network need to be able to discover which telephone numbers are associated with which gateways.
The IPTEL working group is developing TRIP (Telephony Routing overIP), a policy driven inter-administrative domain protocol for advertising the reachability of telephony destinations between location servers, and for advertising attributes of the routes to those destinations. TRIP is designed to allow service providers to exchange routing information in order to avoid the over-provisioning or duplication of gateways. It uses established Internet protocols such as BGP, OSPF and SCSP.
For further information see the OSS/BSS
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