 |
Product Information: |
|
 |
|
 |
|
| Pingtel SIPxChange Call Manager |
|
| Price: Please call |
|
| |
|
| |
 |
|
|
 |
Usually 2-5 Working Days |
|
| |
|
|
|
 |
Description: |
|
|
Pingtel’s SIPxchange CallManager is a server-based, integrated central call routing engine that also provides directory services across an enterprise VoIP network. CallManager provides a very low-cost centralized call management solution that unifies control of enterprise voice application islands. This standards-based solution interoperates with all legacy systems such as branch PBXs, call center, voice mail and other messaging systems.
CallManager is often used by multi-site enterprises to deliver seamless Application Routing, Toll-Bypass, and WAN consolidation
Pingtel’s SIPxchange CallManager is based on the Session Initiation Protocol (SIP), and provides a powerful suite of SIP routing and management features. In combination with your existing network, SIPxchange CallManager delivers real enterprise wide communication benefits by allowing seamless integration of new SIP-based applications and technologies while preparing your network for future VoIP migration.
CallManger Software Components:
Pingtel Comm Server
Routes, authenticates, and authorizes calls (SIP messages) among other components
Pingtel Configuration Server
A browser based graphical user interface (GUI) that enables easy configuration
SIPxchange CallManager attribute highlights include the following:
Call Routing Engine
SIPxchange CallManager provides the most flexible method for routing calls to the correct phone within the IP domain or the most appropriate gateway for long distance cost savings. In fact, the powerful SIPxchange CallManager routing engine provides capabilities that surpass traditional CLASS 4 PSTN equipment. Calls received by SIPxchange CallManager from the PBX (via the gateway) can be routed in multiple ways.
Proxy Forwarding
SIP messages controlling call setup can be forwarded either to other SIP servers or to SIP gateways. Elements such as the SIP message route, the SIP message type, or other identifiable fields within the SIP message, can be used as triggers to route the message to the next desired hop. Proxy forwarding can also be used to direct call-setup (INVITE) messages to a Security Proxy.
Proxy Authentication
SIP security is insured by SIPxchange CallManager authenitcation and authorization of all SIP messages using SIP credentials.
URI Mapping Engine
The URI mapping engine maps calls to target destinations. Administrators can define rules to direct calls over the internal LAN, corporate WAN, or public internet to the target phone without using the PSTN, or to a least-cost gateway for long distance calls using the PSTN.
Rules Based Routing
A two-stage matching procedure enables if/then processing on route selection. All rules are specified using XML-based syntax, and routing can be performed based on numerous portions of the target URL, such as Host, User, or a required Permission match. |
|
|
Specification: |
|
|
Distributed enterprises looking to reduce their operating costs are using Voice over IP (VoIP) to carry calls between offices over the IP data network. This application of VoIP generates immediate savings by eliminating long distance toll charges and the need for costly PBX tie lines.
SIPxchange CallManager can be used for enterprise toll bypass and WAN consolidation, as the core infrastructure for enterprise convergence, as a PBX with 3rd party voicemail or as a powerful call router managing the interaction between various peripheral systems and the rest of an enterprise‘s IP network, including remote locations. Pingtel’s SIPxchange CallManager offers significant benefits to enterprise users:
Immediate savings on calls within the network
Dramatically reduces the costs associated with proprietary solutions.
Future-proofed voice system
Offers scalability options for easy application and technology upgrades in the future.
Accommodation of large call volumes in a VoIP network
SIP protocol usage eliminates long distance toll charges between offices and the need for costly PBX tie lines.
Low-risk/high-reward solution
Open architecture, full SIP implementation and adherence to SIP standards means development and interoperability issues are minimal.
Protection of prior telecommunications investment
VoIP phase-in while continuing to operate legacy voicemail system.
Most flexible method for routing calls
Powerful routing engine provides capabilities that surpass even those of traditional CLASS 4 PSTN equipment.
Toll Bypass/WAN Consolidation
Traditional voice calls between headquarters, branches, and home offices are converted into compressed data packets and carried over the private IP network or the public Internet to dramatically reduce long distance toll charges.
System Features
- Aliasing facility
- Automatic Route Selection
- Auto-restart services after power failure using watchdog facility
- Browser-based configuration system
- Call Admission Control
- Codec support
- Full Hot Standby (using fixed IP addressing schemes)
- Hunt group
- Multi-site / multi-location station and gateway
- Off-premises stations
- URI mapping engine for call routing and inter-company (/domain) SIP calls
- Web services APIs for Config server
SIP Implementation
- RFC 3261 Session Initiation Protocol using both UDP and TCP transports
- Advanced call control using RFCs
- 3515 Refer Method
- 3891 Referred-By header
- 3892 Replaces header
- Provide for consultative and blind transfer and third party call controls
- RFC 3263 Locating SIP Servers - use of DNS SRV records for call routing control and server redundancy.
- RFC 3581 Symmetric Response Routing (rport)
- RFC 3265 SIP Event Notification - for phone configuration and
RFC 3842 Voice mail message waiting indication (MWI)
- RFC 3262 Reliable Provisional Responses
- RFC 2833 Out-of-band DTMF tones
- RFC 3264 Offer/Answer model for SDP for Codec Negotiation
- Early media (SDP in 180/183)
- Delayed SDP (SDP in ACK)
- Re-INVITE: Codec change, hold, off-hold
- Route/Record-Route header fields
- Configurable RTP/RTCP ports
- Configurable SIP ports
|
|
|
|
