CALL FOR REQUIREMENTS AND PRICES
949-387-6820
*Discount Per Minutes Rates available
VOIP
Systems and Hardware Solutions
Servers
VOICE INTERNATIONAL provide you with a server and OS that has been tested and proven to work with the Asterisk Open Source PBX.. We are also pleased to offer Digium telephony boards installed and pre-configured with your Asterisk server. Whether you are looking for one server or setting up a supply chain model for hundreds of servers for your business needs.
The Asterisk Open Source PBX provides a basic audio conferencing application as part of its standard offering. VOICE INTERNATIONAL has expanded this offering to deliver a product that is competitive with most brand name conferencing appliances on the market today for a fraction of the price. Each Conferencing Server comes with use of a fully brandable web portal application to allow for provisioning and end-user managements, and moderators of audio conference have full use of the web based moderator application. The flexibility and multi-protocol nature of the Asterisk Open Source PBX also allows for interconnect options from your carrier or internal PBX in FXS/FXO, DS1, or direct VoIP interconnect via SIP, IAX2, or H.323. By partnering with a larger service provider/carrier, Conference servers also provide a unique application that allows you to setup seamless overflow of your on-demand conferencing so that when demand for your conferencing exceeds your on-premise supply, conferences can automatically overflow to your provider's conferencing resources with no change in experience to users of the service. Solutions achieve an extremely quick return on investment!
Your phone system is one of the most important tools for communications in your business. If you're going to make or have already made the decision to use the Asterisk Open Source PBX with or as your phone system, you need to protect your investment.
VOICE INTERNATIONAL provides a monthly and affordable service to automatically and securely backup all of your system specific data to two geographically diverse, off-site, carrier-class, co-location facilities. Additionally, we can use the same secure conduit to monitor your Asterisk install for application/system health on a 24x7x365 basis. We have automated monitoring tools to make sure that your Asterisk instance is running and able to answer and place calls to all of your interconnect providers. We can also do DS1 alarm monitoring for those with TDM interconnects.
Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the features one would expect of a large proprietary PBX system such as Voicemail, Conference Bridging, Call Queuing, and Call Detail Records.
Call Features
- ADSI On-Screen Menu System
- Alarm Receiver
- Append Message
- Authentication
- Automated Attendant
- Blacklists
- Blind Transfer
- Call Detail Records
- Call Forward on Busy
- Call Forward on No Answer
- Call Forward Variable
- Call Monitoring
- Call Parking
- Call Queuing
- Call Recording
- Call Retrieval
- Call Routing (DID & ANI)
- Call Snooping
- Call Transfer
- Call Waiting
- Caller ID
- Caller ID Blocking
- Caller ID on Call Waiting
- Calling Cards
- Conference Bridging
- Database Store / Retrieve
- Database Integration
- Dial by Name
- Direct Inward System Access
- Distinctive Ring
- Distributed Universal Number Discovery (DUNDi™)
- Do Not Disturb
- E911
- ENUM
- Fax Transmit and Receive (3rd Party OSS Package)
- Flexible Extension Logic
- Interactive Directory Listing
- Interactive Voice Response (IVR)
- Local and Remote Call Agents
- Macros
- Music On Hold
- Music On Transfer
- Flexible Mp3-based System
- Random or Linear Play
- Volume Control
- Predictive Dialer
- Privacy
- Open Settlement Protocol (OSP)
- Overhead Paging
- Protocol Conversion
- Remote Call Pickup
- Remote Office Support
- Roaming Extensions
- Route by Caller ID
- SMS Messaging
- Spell / Say
- Streaming Media Access
- Supervised Transfer
- Talk Detection
- Text-to-Speech (via Festival)
- Three-way Calling
- Time and Date
- Transcoding
- Trunking
- VoIP Gateways
- Voicemail
- Visual Indicator for Message Waiting
- Stutter Dialtone for Message Waiting
- Voicemail to email
- Voicemail Groups
- Web Voicemail Interface
- Zapateller
Computer-Telephony Integration
- AGI (Asterisk Gateway Interface)
- Graphical Call Manager
- Outbound Call Spooling
- Predictive Dialer
- TCP/IP Management Interface