IP- PBX Functionality Options With the powerful features integrated in the AtomOS system from AtomAmpd, installing & configuring a cost- effective and extensible VoIP solution is easily possible. 4/26/10
Table of Contents Product Brief... 3 Components... 3 Features... 3 IP- PBX Specifications... 4 IP Commutation Matrix... 4 Telephony Facilities... 4 Voicemail System... 5 Conference Services... 5 Auto- attendant... 5 Trunks, Routes y and User Contexts... 6 Extension Locking... 6 Least Cost Routing... 6 DISA... 6 GUI Interface... 7 CDRs and Automatic Call Cost Calculation... 7 Mobility... 7 Music on Hold... 7 Voice Recording Module... 7 Automatic Call Distribution module... 8 IP-PBX 2
Product Brief The Communication Services Feature Set module complement the functionalities of the AtomOS platform, and add Telephony and VoIP capabilities to the system. The main objective of this module is building modern and complete telephony systems, for both internal and external communications within companies or institutions of any size or nature. Our standard based communication system permit the integration with almost any standard based VoIP device, going from IP Phones and Soft- phones, Analog Telephone Adapters (ATAs) and VoIP gateways of any brand. It also permits the integration of Telephony cards or modules, that can be used to connect traditional phones or traditional lines (POTS) to our solution. Components Our VoIP communication solution consists on any network appliances loaded with our AtomOS software and VoIP modules, complemented by IP Phones, Softphones, ATAs, VoIP Gateways and any VoIP line or trunk. The Media Gateway Feature Set module adds the capabilities to integrated telephony cards or modules to the solution, and converts any capable PC or appliances in a VoIP gateway call to regular POTS or TDM calls. Standard telephony is available for analog lines or phones (FXO or FXS), and also for digital connections running over E1 or T1 ports. Features Telephony Features Presence and Messaging (on some phones) Authentication Voice mail Message waiting indications Black/White lists Call Transfers (attended or unattended) CDR automatic generation Automatic Call Cost calculation Call Forward on Busy Call Forward on No Answer Call Monitoring Call parking Call Routing: bu trunk, DID and/or ANI Call Snooping Call Waiting Voice Mail to email integration Extension Blocking Trunk passwords Extension profiles: contexts, categories and permissions Extension substitution Distinctive Ringing Do not disturb ENUM Auto Attendant ACD Agents: local and remote agents Music on Hold Music on Transfer Media Gateway (Protocol and VoIP CODEC mediation and conversion) Remote user and remote branch support Text- to- Speech (requires special integration) Voice Recognition (requires special integration 3 Party conferences Conference Bridges Facility directories Call- inr ID Call- inr ID Blocking DISA Call Recording (automatic or on- request) IP-PBX 3
Codecs ADPCM G.711 (A- Law y Mu- Law) G.723.1 (pass- through only) license required for call termination G.729 (pass- through only) license required for call termination G.726 GSM ILBC Speex Linear LPC- 10 VoIP Protocols SIP (Session Initiation Protocol) MGCP (Media Gateway Control Protocol) on request IAX (Inter- Asterisk Exchange) H.323 SCCP (Cisco Skinny) on request IP- PBX Specifications IP Commutation Matrix The core function of the IP- PBX module is the IP Commutation Matrix, in which all the call routing decisions are taken and the other telephony applications and/or modules are bonded. In our system, the core module is a software based IP commutation matrix, which inherits all the robustness and power of our operating system (Unix based), and scales almost without limits depending on the hardware used for the platform. This application was designed for global solutions supporting several types of applications such as PBX, Call Center, IVR platforms, VoIP Gateways, VoIP Media Gateways, Voice Mail platforms, CTI, etc. Complementing the IP commutation matrix, we found other vital components to the solution, which are the Signaling Module and the Trans- coding Module. The signaling Module integrates multiple standard based VoIP signaling protocols, such as SIP, H.323 and MGCP, and in that way, permits any device or adapter supporting those protocols to be configured with or against our systems. The trans- coding module permits the communication of devices that don't support the same CODECS, and even the same VoIP signaling protocols. This kind of functionality converts our system on a very sophisticated Media Gateway device, but requires a lot of processing on the appliances or units due to the trans- coding and trans- signaling that needs to be done with each call. Telephony Facilities The telephony features available in our solution are the same that can be found on any commercial Telephony solution, VoIP or not, from other companies. The list of functionalities is very large but the most important are: Call transfer services (attended or un- attended) Call waiting 3- way calling Call forwarding Music on hold Conference Bridges IP-PBX 4
Call groups, Hunt groups Call Capture and pickup groups Call recording (requires and additional module) Automatic Call Distribution (ACD required additional module) On- line CDR and call cost calculations Telephony directory services for by name dialing and other options Extension substitution Voicemail Voicemail to email integration Voicemail System The Voicemail system service is available for each extension in the system, and can be enabled by the administrator at any time. This is a basic functionality that is included for every VoIP system in our platform, and allows the end user to configure and to use features such as the following: Name personalization Busy messages personalization Unavailable message personalization Voicemail forwarding to other users Voicemail classification folders Email notifications Voicemail to email integrations, and others Conference Services Although our system permits the connection of 3- way conferences programmed on the VoIP phone, our solution also permits the configuration of conference bridges or rooms, with no limits in the number of users or facilities that can be configured. The conference rooms will be accessible through a dedicated facility number for local and external users who will be able to access it as configured by the administrator. The administrator can also define the owner of each room (password based selection), and that user will be allowed to block and unblock the room, dialing certain digits on it's regular phone. If a conference room is blocked, no user will be able to enter it unless the administrator un- lock it using the right procedure. This implementation provides the concept of personal conference room to certain users that will be able to invite other local or external users as the y require it. Auto- Attendant The Digital Receptionist service, or Auto- Attendant, allows the configuration of sophisticated IVRs trees with multiple choices, that will be selected by users using their phone keypad. The auto- attendant functionality will recognize the options requested by the call- in user, in order to route the call to the destination, facility and extension selected, and even to route the call to other auto- attendants for more complex applications. IP-PBX 5
Each option programmed on the IVR can be viewed as a facility block, which can be any of the following types: Informative Message: pre- recorded message with institutional information Auto attendant: To interconnect IVRs between then and form complete navigation trees and applications Extensions Voice Mail DISA services Trunks Hunt/Ring Groups ACD Queues Trunks, Routes and User Contexts Once the trunks that connect the system to external devices and networks are defined, it will be necessary to configure the routing and prefixes which will control the way the calls are routed within the system. Each route can be associated to certain profiles or categories of users, and the users will only be able to use such routes when his extension belongs to the corresponding category or profile. In this way, it is possible to define extensions from which only internal calls can be made, or any type of restrictions for local and/or long distance calls. Other possibility for this restriction is to configure a unique password on the route that will be asked by the system every time a user tries to make a call through it. Extension Locking The traditional VoIP protocols and phones do not integrate any method to lock and unlock the extension, but our implementations permit the administrator of the system to configure such enhancements. The locking can be made automatically by the system when the extension is idle for certain period of time, or can be triggered by the user dialing to some facility in the system. In the same way, a locked extension can be unlocked when a user is trying to make a call, dialing the password for that extension when the system asks for it. The system permits the configuration of which trunks will check the locking status for the extension, for having some destinations or trunks that can be used by everyone even on locked extensions. This is particularly useful for emergency or intra- company calls. Least Cost Routing The Least Call Routing functionality can be integrated as an extension of the regular Call Routing feature, and allows all the necessary configurations to route calls to preferred trunks or devices, which provides the least cost for that specific destination. Configurations on which two systems on different locations are connected through VoIP trunks, permits to have routing rules completely transparent to the end user, which can route call through the gateways or telephone lines of any on those systems, depending on the destination pattern or number dialed by the user. When congestion is detected on one trunk due to lack of capacity or any kind of error, the system will try handling the call through other routes with higher costs for that particular destination pattern. DISA The Direct Inward System Access feature (DISA), allows remote access to any of the functionalities or facilities in the system through any telephone line connected to the solution. Using this feature the call- in user can dial one of the phones in the system, get access to a DISA facility, dial a required password for authentication, and then receive an access tone IP-PBX 6
which permits them to dial an external number or local facility the same way as in a regular extension of the system. The DISA facilities can be programmed in certain categories or context, in order to permit access only to certain services or facilities. GUI Interface All the basic and advanced configurations for the VoIP system and telephony features can be done through our Web- based GUI interface: extension configurations, trunk configuration, routing configuration, etc. As described in the next section, the GUI also incorporates a graphical application to obtain utilization reports for extensions and facilities, based on the automatic CDR generation of our solution. Diagnostic and troubleshooting tools are also available, allowing the administrator to see the real- time status of extensions and trunk, call established, system logs, etc. CDRs and Automatic Call Cost Calculation All calls made or received on the system will generate a Call Detail Record (CDR) that will be stored on an internal database in our system. This call records are generated in real- time, at the very moment the user hangs up the call. These records can be accessed through a section on the GUI, in which the user may select complex filters in order to get only certain type of call records, using parameters such as the date, origin and/or destination of the call. Filtered or unfiltered records can be exported to a delimited plain- text file, or even to an HTML file, for further processing in external application such as Excel. Our system also permits automatic Call Cost calculation based on the parameters entered by the administrator. Our GUI permits to enter patterns and all the parameters that can be used to calculate the cost of each call made through any type of trunk or external telephone line. Mobility Other functionalities or applications on our system are the possibility to permit connection of remote users with IP Phones or Soft- phones. Using the characteristics of Signaling and voice bearing methods of VoIP, teleworking and roaming users can be connected to the solution as if they were physically located in the office on which our system is installed. The VoIP trunking options permit the configuration of remote branches or interconnection with other VoIP devices over the Internet, private WAN networks, or even using VPNs for more security and reliability. Music on Hold For call and hold, queues and IVRs, our platform has a customized Music on Hold facility, which permits the adding of songs and recordings in MP3 or WAV formats, through the administrations web page. Voice Recording Module The Voice Recording module is an add- on to the base IP- PBX module, which can be used to record voice calls made or received by any extension in the system. The recording action can be triggered by the call- in or called user, when they enter in their phone a special key sequence programmed by the administrator, or even automatically for all calls in Call Centers or Support environments. Those calls will be recorded and stored on a digital format (MP3), and can be accessed any time using our Web GUI administration tool. IP-PBX 7
Automatic Call Distribution module The Automatic Call Distribution module (ACD) is an add- on to the base IP- PBX module, and allows the configuration of call queues for end user attention for call- centers and support applications. The module allows the configuration of both dynamic or static agents, connected through local or remote extensions. Each attention queue permits the configuration of the announcements parameters, maximum number of waiting calls in the queue, the announcement of the position to the call- in user, and the ringing method and for the available agents. Our ACD module does not required additional license per agent or extension, but only for the number of configured queues. IP-PBX 8