One Unified United States
One Unified specializes in consulting IP Telephone solutions for small and medium size businesses. For businesses on a tight budget but still want the all the bells and whistles, we recommend and use the Asterisk Telephone Solution. You can have the best
One Unified specializes in consulting IP Telephone solutions for small and medium size businesses. For businesses on a tight budget but still want the all the bells and whistles, we recommend and use the Asterisk Telephone Solution. You can have the best of all worlds: IP Telephones, regular telephones, auto-attendant, call center functions, voicemail, voice over the internet, integrated branch offices, softphones, plus a whole lot more for a surprisingly small price.For businesses looking for the name brand capabilities of Cisco, we support Cisco Callmanager, Unity, and CCME (what we like to call Callmanager Miniature Edition).
One Unified
137 S. Kings Drive
Charlotte
NC ,28204
United States
Phone: +1 704 576 5101
Site: http://www.oneunified.net
Contact Person
Ray Burkholder / Director, Market Development
Phone: +1 704 576 5101
- International VoIP Wholesale Provider
- Internet Telephony Service Provider
- Wireless Broadband
- Voip Termination ISP
- Network Service Provider
- Voip consulting
- Hardware
- Phone To Phone
- Phone To PC
- Web Call
- PC to Phone
- IP devices
- Web To Phone
- Voice and Video Conferencing
- Termination
- Call Relay VoIP Solutions
- Call Routing VoIP Solutions
- Partnering for Origination and Termination
- Project Mangement Services
- System Integration
- Consultancy Services
- Installation and Support Services
- H.323 Wireless/ GSM VoIP Solutions
- Cisco Systems
- Billing system of own development
Session Initiation Protocol (SIP) is a text-based standard put in place by the Internet Engineering Task Force (IETF) and is the primary protocol for Voice over Internet Protocol services. SIP is similar to a MGCP in that it's a signaling protocol that can create, maintain and terminate sessions in IP-based networks. These sessions include multimedia conferencing and two-way phone calls. SIP protocol can run on User Datagram Protocol (UDP), Stream Control Transmission Protocol (SCTP) and Transmission Control Protocol (TCP) because it was created to function independently from the underlying Transport Layer as an Application Layer.
Danec Inc.
A huge range of telecommunication solutions starting with the standard minute based rates and FREE worldwide VoIP telephony up to flat calling rates for different countries and destinations with your own inbound DID from more than 34 countries. For busine
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