New (nicer) skin

We added a new default skin for sipXconfig and moved away from the traditional yellow background. All of sipXconfig is now easily skinnable including the creation of a custom login page.

Even easier installation / device discovery

In addition to plug & play management of phones and gateways, this release adds an auto-discovery function for devices. Phones and gateways are found automatically and presented in a table from where they can be added to the database in one click only. Also in this release a new network services test capability has been added. When sipXconfig starts all the necessary network services, such as DHCP, DNS, NTP, TFTP, FTP, HTTP, are tested for correct configuration and operation. Detailed error messages are printed with troubleshooting information. The test suite can also be downloaded to a laptop and run under Windows. That way the tests can be run on the same subnet the phones are connected to,

Extended User Portal / time based find-me / follow-me

The sipXecs user portal is available to every user of the system and allows individualized management of key user features. Tn addition to the management of unified communications and voicemail, the user portal now also supports time-based find-me / follow-me, personal call history, personal phone management, and personal management of phone book, speed dial, and presence subscriptions.

Personal auto-attendant / Individual zero-out capability

Every user gets a personal auto-attendant that can be configured on the user portal or by the admin. When a caller is redirected to the user's voicemail, the caller will hear an individually recorded greeting that provides instructions on how to reach the user or to leave a voicemail. The user can define individual keys, such as press 1 to get forwarded to my cell phone, press 2 to get transferred to my assistant, press 3 to reach my girl friend and press 4 to leave a voicemail. Also, it is possible to define an individual transfer extension for the 0 key, which is usually the operator or a personal assistant.

Import from and export to Excel

During the planning phase before an installation, many users create cut-sheets that identify users, extensions, phone models, passwords, and other necessary parameters. Once this information is captured in Excel it can be uploaded into sipXconfig, greatly simplifying the installation process. At the same time this information can now be exported to Excel as well.

Localization of the Media Server

The last release brought about localization of the Config Server as well as the voicemail user portal. In this release we are adding localization of voice prompts for the auto-attendant and voicemail systems for a first set of languages. German, Italian and Polish are currently in process with others to follow. We will define a simple format for language packs, so that localization can be easily done in the community.

Busy Lamp Field (BLF) and Presence

In release 3.8 we got BLF almost right and we added a new SIMPLE based presence server. However, because of a bug in the Polycom 2.x firmware, BLF still does not work reliably under all use cases. Release 3.10 will see improvements in the BLF implementation that will make the feature less dependent on phones and extend the capability to phones that comply with the SIP stndard (e.g. LG-Nortel phones).

Integration with Microsoft

Release 3.10 provides a unified communications solution integrating with Microsoft Exchange 2007 as well as Active Directory. Microsoft Exchange 2007 can be selected as an alternative voicemail system directly in the dialplan. This provides a speech enabled voicemail system integrated with the Exchange email and calendar system. Synchronization of users and their credentials can be done automatically using the integration with Active Directory.

Time-Based Routing

We are introducing a time-based routing capability into the dial plan. This is based on a new redirector plugin and allows all kinds of time dependent features and feature interactions.Every dialing rule has now an optional schedule attached.

Paging Server

Based on the specification we published some time ago we added a group paging server to the sipXecs system. The paging server is added as a distributed component where several paging servers can be added to the system, either on the same host as the rest of the sipXecs system or on separate HW. The paging server allows group paging of SIP phones. Different announcement audio can be selected to announce a page. Regular SIP phones that provide auto-answer capability can be used or dedicated SIP-based speakers (e.g. in-ceiling)

Overhaul of the ACD server

The ACD server has been overhauled and made a lot more stable. Additional features include agent wrap-up time as well as an agent auto-sign-out capability in case the agent does not answer a call. Also, the overflow mechanism has been enhanced with a better algorithm and more destinations. E.g. it is now possible to use a queue, a hunt group or an individual extension as an overflow destination. If no agent is signed in the call can overflow to voicemail.

Improvements to the Auto-Attendant

Several important improvement to the auto-attendant subsystem have been queued up for quite some time. In particular we added transfer rules and targets to handle invalid response. Also, the auto-attendant can now transfer to external numbers with proper permissions.

Improvements to Hunt Groups

More flexibility is added to the management of hunt groups so that it is possible to specify destinations for no answer. Such destinations can include voicemail, auto-attendant, an extension or SIP URI, or another hunt group. See XCF-831. In addition, the difference in behavior between transferring consultative or blind to a hunt group will be eliminated. On a per hunt group basis the admin can now configure whether user call forwarding rules shall be followed or not. This allows disallowing forwarding of calls to e.g. user's cell phones as part of a hunt group.

Overhaul of the security and authentication system

The security system of sipXecs for call authorization has been overhauled. This should eliminate previous restrictions on call tromboning or other external forwarding (blind or consultative transfer of an external call to an external number) while strengthening the security of the system. Gateway templates now automatically configure Access Control Lists (ACL) to prevent unauthorized LD calling.

Improved E911 call routing

Resiliency of emergency call routing has been improved. Phones able to directly route emergency calls to a gateway without requiring the sipXecs server to be operational are now automatically configured to use this feature. Emergency calls, therefore, will now succeed even if the sipXecs server is not available as long as the phone can talk to the gateway.

Simplified dial plan configuration

Gateways can now be added to dialing rules directly from where gateways are managed. A single click adds a newly created gateway to a dialing rule. Removing a gateway automatically deletes all its references in the dialing rules. Gateways continue to offer trunk redundancy and automatic failover in case of busy or unavailable. sipXecs therefore supports more than one gateway per dialing rule.

Registered phones displayed per user

Managing a large number of users, several hundred to several thousand, can be a difficult task. sipXconfig already offers elaborate search capabilities to filter reports. In this release there is now a very simple way to just display phones registered for a specific user. This is possible both by the admin in the admin portal or the user using the user portal.

New device category: SBC

In addition to phones and gateways, sipXconfig can now also manage Session Border Controllers (SBC). A new category of a managed device has been introduced. SBCs are used for Internet call routing rules, remote workers, as well as SIP trunks.

Automated restore from backup

The current restore from backup functionality will be integrated into Config Server.

Server and application statistics, reports, and alarms

We are implementing SNMP / MRTG based statistics into Config Server that allows improved monitoring, alarming and reporting of performance and problems. In addition, the sytem will allow integration into data center management applications.

Support for new Polycom 320 / 330 phones / Polycom 2.2.2 firmware

We are adding support for plug & play management of new Polycom phones. In addition, the plug & play management system has been updated to support firmware 2.2.2. Older phones IP300 and IP500 can no longer accommodate 2.2.2 firmware because of memory constraints acording to Polycom.

Plug & Play Management Support for Linksys Phones

We are adding support for Linksys SPA941 and SPA942 phones fully integrated into the sipXconfig management system thanks to a community contribution.

Plug & Play Management Support for IpDialog SipTone V Phone

We are adding support for the IpDialog SipTone V phone fully integrated into the sipXconfig management system thanks to a community contribution.

Plug & Play Management Support for LG-Nortel 1535 Video Phone

We are adding support for the LG-Nortel 1535 Video phone fully integrated into the sipXconfig management system thanks to a community contribution. This is a new and very attractive desk video phone.

New Report: Login history

sipXconfig now provides a report on the login history. This includes successful and unsuccessful logins from all users (superadmin as well as logins of ordinary users into the user portal).

Symmetric signaling / merged proxy

We introduced symmetric signaling, which is a first step towards supporting NAT traversal natively in sipXecs. This was achieved by merging the two proxies (forking proxy and authentication proxy) into one combined proxy server that communicates on default port 5060.

SIP loop detection

sipXecs proxy server is now able to detect loops and will abort them. We implemented a new IETF draft RFC for this important feature. Previously a call, under certain conditions, could loop indefinitely in the system.

Port to PowerPC (PPC)

sipXecs was ported to the PowerPC (PPC) platform with all the big endian handling for audio processing and other issues.

Port to FreeBSD

A new port was done to FreeBSD. We are still looking for a new maintainer who would be able to maintain this port in the FreeBSD ports library. Refer to XECS-108 for the port files and FreeBSD port for sipXecs 3.10 for documentation..

New XML RPC process management API

sipXconfig now uses a new XML RPC based API to manage processes on the master and slave hosts. Additional security and efficiency is provided over the old CGI based solution. This is a pre-req for the cluster management coming in the next release.

Detailed List

New Feature