SipX CommServer Functionality

From SIPfoundry sipXecs IP PBX, The Open Source SIP PBX for Linux - Calivia

Contents 1 sipX Communications Server Features 1.1 Automatic Route Selection 1.2 Auto-restart 1.3 Call Authorization 1.4 Call Forwarding 1.5 Dial Plan Definition Facility 1.6 Gateway Route selection Facility 1.7 Hunt Groups 1.8 Mapping Rules 1.9 Multi-Site/Multi-Location Phones and Gateway 1.10 Off-premise extensions 1.11 Outbound Call Blocking 1.12 System Security 1.13 Toll Bypass 1.14 Call Transfers 1.15 Call Park & Retrieve

sipX Communications Server Features

sipX open source PBX Communications Server provides the rich PBX feature set that commercial deployments require:

Automatic Route Selection

• Implemented with XML-formatted mapping rules.
• �Mapping values� re-write SIP URLs to specify the �next hop� or destination for a SIP message that has been received by the Communications Server component.
• Direct messages to different SIP/PSTN trunk gateways, either on premise or at a remote premise location, based on any portion of SIP URL or E.164 number.
• Route messages to commercial SIP/PSTN service providers, which reduces or eliminates the need for on-premise trunk gateways.

Auto-restart

• Single sipX application can start all other application processes associated with starting up sipX, including dependent processes that must be started in particular order.
• Configure from browser interface or automatically invoked as part of rc.* startup files in Linux.

Call Authorization

• Control whether a given user or phone is permitted to complete the dialed call, such as long distance, international, or 1-900 calls.
• Success can depend on destination (E.164 number or SIP URL) or on message containing valid SIP credentials.

Call Forwarding

• Options for Forward All Calls, Forward on Busy and Forward on No Answer to an alternate phone in the system.
• For each option, the same alternate number or a different number can be specified.
• Calls can be forwarded to off-system phones, such as mobile phones, with central authorizations.
• End-users employ browser interface to designate alternate phone to forward calls.

Dial Plan Definition Facility

• Dial plans, such as 3, 4, or more digits for internal calls and 8 or 9 prefix for external calls, are supported and set using the central management interface.
• Supports international dial plans with up to 26 digits.
• URI mapping rules can use wildcard-based patterns to determine where and in what form SIP session (call setup) messages are sent.

Gateway Route selection Facility

• Associates a list of SIP/PSTN gateways with specific PSTN (NPA NXX) numbers, including a default gateway.
• Outbound long distance calls can be sent directly to a commercial SIP service provider using this mechanism.

Hunt Groups

• Enables busy/no-answer forwarding to a pre-configured, sequential list of phone numbers.
• Each extension in a Hunt Group is assigned a priority (Q value), such that phone A is priority 1, phone B is priority 2, and phone C is priority 3; when an incoming call is received, the Proxy Server routes the call serially based on priority � in this case to Phone A, then B, and then C.

Mapping Rules

• Mapping rules describe routing rules, dial plans, gateways, hunt group definitions and aliases for E.164 numbers, IP addresses, and URL addresses.
• Alias facility associates alternate names or extensions for a single destination, such as 555-938-5306 ext. 128 = 555-970-0128 = ext. 128 = 10.1.20.221 = sip:fredsmith@sipfoundry.org.
• Includes innovative SMTP address to SIP URL or E.164 feature, allowing �dial by email address� that sipX forwards to SIP or traditional phone, such as fredsmith@sipfoundry.org = sip:fredsmith@sipfoundry.org or 555-970-0128.
• Mapping rules are contained in XML-formatted files that are maintained manually.
• Automatic distribution of the mapping rules to all servers.

Multi-Site/Multi-Location Phones and Gateway

• Individual phones or groups of phones can be configured with separate dial plans, permitting a single server cluster to support multiple offices and separate gateways in multiple locations.
• Normal multi-gateway operation also can direct calls to �local� or �remote� gateways as determined by dialed string, thereby using a private IP network to permit PSTN toll-charge avoidance.
• Phone-to-phone calls between separate geographical locations can stay on the IP network when a private IP network is available.

Off-premise extensions

• Use any SIP phone, or an analog-to-IP gateway, in any remote location that can be reached via the corporate intranet or public internet; appropriate bandwidth for a voice call to that location must exist.
• A centralized server can provide call setup services for any phone installed or configured in any remote geographical location connected via IP.
• Off-premise phones can use either centralized SIP/PSTN trunk gateways or gateways local to that phone.

Outbound Call Blocking

Calls from phones to PSTN numbers, or classes of numbers, can be blocked based on:

• The destination of the call; for example, when a user or device cannot initiate an international long distance call.
• The source of the call; for example, when a lobby phone can only initiate calls to internal numbers.

System Security

• HTTPS secures non-SIP communication between sipX components.
• HTTPS secures communications between sipX components and admin and user consoles.
• Secure channel for retrieving messages from voicemail repository.
• HTTP digest authentication for SIP signaling, as specified in RFC 2617, is used for authentication challenges between SIP endpoints and sipX components.
• HTTP digest implementation supports MD5.

Toll Bypass

• Use public or private IP network to permit toll avoidance.
• Phone-to-phone calls between separate geographical locations can stay on the IP network when a private IP network is available.
• In multi-server environment, server-to-server direct routing over IP for both intra-organization and inter-organization calls.
• In multi-gateway environment, direct call to �local� or �remote� gateways as determined by dialed string.
• Inter-organization SIP/IP calls can be challenged for authorized credentials.

Call Transfers

• Blind transfer (Unannounced) to a different phone without speaking to the other phone prior to transfer.
• Consultative transfer (announced) to a different phone without speaking to the other phone prior to transfer.
• Consultative transfer (announced) to a different phone after speaking to the other phone prior to completing the transfer. (Consultative transfers require a SIP phone that supports this feature)

Call Park & Retrieve

• Separate park server allows flexible implementation of call park & retrieve functionality
• User configurable call park extensions
• Supports multiple call park extensions
• Plays music on hold while on park