sipXecs includes a simple yet complete voicemail system. Users can access voicemail through their phone, via a web browser, or receive their voicemail as email. Voicemail to email is a simple unified communications type with a twist. Included as part of the email are hyperlinks that allow the user to erase his or her voicemail from the voicemail server.

For the number of minutes of voicemail, administrators are only limited by the capacity of the storage in their servers. Additionally, there is no hard set limit for how many voice paths (ports) can be active to the voicemail server at one time. System speed is the only limiting factor.

sipXecs can optionally integrate with a Microsoft Exchange 2007 Unified Messaging Server for a fully unified messaging experience. The system administrator can also mix and match with some users on the internal voicemail system and some on Exchange.

The multilevel Auto Attendant service provides system-wide answering of incoming calls, dial by name abilities, automated transfer to local extensions, access to remote voicemail retrieval, and transfer to other auto attendants. The following screenshot shows the sipXconfig interface for modifying system Auto Attendants:

The number of auto attendants is limited only by the administrator's creativity and the callers' patience.

There are multiple methods of supporting Music on Hold (MoH) on SIP-based phone systems. For SIP phones that can use it, sipXecs supports a standard as defined in an IETF draft written by Dale R. Worley of Nortel (http://svn.resiprocate.org/rep/ietf-drafts/worley/draft-worley-service-example-01.html). This standard is dependent on the phone to transfer the call to a service that is playing the MoH, and then recall the caller when the caller is taken off hold. Presently, this method is known to be supported by Nortel, Polycom, and Snom phones.

For calls from an ITSP, the sipXbridge service can provide MoH, which allows any phone to have MoH capabilities without having to support the IETF draft.

The sipXpark service allows users to park an active call to a park extension, and then later pick up that call from any phone by dialing a retrieve code and the park extension. While the call is parked, the caller will hear call park audio, which can be uploaded by the administrator. This following screenshot shows a typical Call Park Extension and its basic configuration elements:

Park orbits can be configured to allow single or multiple callers to be parked. If multiple callers are parked, they are retrieved in a first-in first-out (FIFO) order. An unlimited number of park orbits can be created.

The sipXecs paging service (sipxpage) allows the system administrator to define multiple paging groups of phones to contact for paging. When a user dials the paging code followed by the paging group number, all the phones in the paging group go off-hook on speaker phone, a tone (which can be uploaded) is played, and then the user may broadcast their message. The following Paging Groups configuration screen allows the administrator to configure the paging dial prefix and define a group of phones that will go off-hook to play the pages:

At present, Polycom and LG Nortel phones will be automatically configured to support paging when added to a paging group. Other phones may be configured manually.

The intercom feature of sipXecs allows the administrator to configure phones to automatically answer calls. A user dials a feature code and extension, the receiving phone goes off-hook on speaker phone, and the two users can have a conversation. Polycom, LG Nortel, and Snom phones can be automatically configured to support this feature.

The conferencing service allows Meet-Me voice conferencing capabilities. Administrators can create as many conferences as they would like with the ability to have separate conferencing servers if the conference demand is high. Conference controls are also integrated into the user portal so that every user can have a personal conference bridge that can be easily administered. The following sipXconfig screenshot shows the system administrator all of the conferences defined in the system, who owns them, and how many participants are in each:

The sipXecs call center solution (sipxacd) integrates into the configuration server where call center lines, queues, agent behavior, and features are configured. The configuration server also provides real-time statistics about call volume and agent activities.

Like other services, the sipxacd service can be configured to run on the same host as the rest of the sipXecs, or it can be installed on a separate host still managed by the configuration server. It is possible to define and configure several ACD servers for the same system and manage them all through the configuration server from a central location.

The ACD Queue configuration screen is shown as follows. As with most sipXconfig pages, the ACD Queue configuration screen is well documented, explaining each of the settings.

Over 75 different types of phones and gateways can be managed directly in the sipXecs configuration server. The sipXecs configuration server provides default profiles for every managed device. Configuring a phone to register with sipXecs is very easy and will be explained further in Chapter 5.

The following screenshot is the sipXconfig phone configuration screen. As can be seen by the possible configuration options on the leftside of the screen, almost every configurable option for a phone can be modified for each phone:

There is an additional service available that will automatically discover unassigned phones on the network and allow the administrator to add them into the system.

Working with SIP provides a great flexibility for different addressing schemes based both on usernames and telephone extension numbers. As a standard SIP-based solution, sipXecs allows an organization to derive its naming scheme from its domain name. This allows the same addressing already used for email to be extended to real-time multimedia communications.

The following sipXconfig user configuration screen allows the administrator to quickly change names or email addresses for the user:

Users can be created one at a time in the sipXecs configuration server, imported from a Comma Separated Values ( CSV) file, synchronized with LDAP, or Microsoft Active Directory, or added programmatically via the SOAP interface provided by the configuration server.

The User self-service portal gives each user of the system a web portal to change many configuration items that the system administrator may need to have done for them before.

The following screenshot shows what users are greeted with after they log in to the PBX with their web browser:

Users can manage their voicemail messages, change their active voicemail greeting, set up to two email addresses to forward their voicemail to, change their Personal Identification Number (PIN), set up call forwarding with schedules, create a personal auto attendant, set up to 9 Voice Mail distribution lists, manage their conferences, add or remove speed dials from their phones, view call history, sign in and out of ACD queues, maintain a phonebook, and see what phones they may be registered on.

Users have the ability to set up call forwarding options based on any schedule they would like. For instance, a user may choose to have calls forwarded to his or her cell phone and desk phone to ring at the same time during normal working hours.

The following screenshot shows an administrator's view of a user's call forwarding configuration:

sipXecs was designed with the ability to localize the entire system for different regions of the world. Localization (language) packages provide the ability to change voice prompts, user interface prompts, regionally specific dial plans, and localization files for third-party components.

There are currently nine localization packages available for sipXecs; US English, German, French, UK English, Spanish, Mexican Spanish, Canadian French, Dutch (Netherlands), and Brazilian Portuguese.

Localization packs can also be developed by system administrators if the settings in the available packages don't really meet your regional needs. These packages need to be updated for every future release of sipXecs because of user interface screen changes and new features being added.

Increasingly, telecommunications services are being provided across the Internet by companies referred to as Internet Telephony Service Providers (ITSP). Rather than relying on physical phone lines, ITSPs utilize SIP to provide phone service to almost any location. One of the hurdles that are typically faced by SIP phone systems is dealing with company firewalls and Network Address Translation (NAT). The sipxbridge service handles the ITSP interface requirements and NAT traversal for the PBX.

The Internet calling configuration screenshot shown above allows the system administrator to configure calling routes to Internet Telephony Service Providers.

sipXecs supports near real-time reporting of Call Detail Records (CDR) in the configuration server. The sipxcallresolver service is polled by a SOAP web interface to get access to information about the ongoing calls. Historic information is also maintained in the system regarding all calls. Administrators can filter CDR information based on time, date, caller, and called party. Additionally, CDR information can be downloaded in CSV format, or accessed directly from the SQL database that houses it.

A cluster is a collection of servers working together to act like a single system to provide high availability and load balancing. sipXecs provides the ability to create a cluster of systems to form an iPBX that allows administrators to build a redundant communications system. This configuration also allows sipXecs to be deployed as a multi-branch office solution that is centrally managed. It acts as a single large system with a cohesive dial plan and number portability between branch offices. In a clustered configuration, the sipXecs scalability can extend into several thousands of users distributed over different locations or offices.