Page Comparison

=sipXecs - Unified Communications System for the Enterprise (IP PBX) - Roadmap=
This page is intended to provide some information around the project team's
thinking for new things coming up. We would like to solicit input and get some
feedback on our current direction, so please post comments to the sipX users
list.
== Proposed Release 4.2 ==
===Other projects integrated===
For release 4.2 we will be looking to integrate other exciting open source projects. Some examples:
*Openfire Integration
*Dimdim Integration
*Deeper FreeSWITCH Integration with new voice mail
===Release Timing===
See [[When will release X be available?]]
===Feature List===
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<table width="90%" cellspacing="0" cellpadding="5" style="border:1px solid #c2c2c2;">
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 <th><big>Release 4.1 / 4.2</big></th>
 <th>Comments</th>
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 <td colspan="2">For detailed status, see the [http://track.sipfoundry.org/browse/XX/fixforversion/10552 4.2 tracker road map]</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Openfire Integration</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release adds [[Instant Messaging and Presence for sipXecs]]. The Openfire XMPP server runs as a component of a sipXecs cluster with centrally managed configuration. Users created on the SIP side automatically synchronize with the XMPP side. Presence federation between phones (BLF presence) and IM presence allows seeing when a contact is on the phone. Presence based routing allows call handling based on presence state. Phones, such as the Counterpath Bria softphone, are auto-configured to include both SIP and XMPP capabilities. Federation with Google Talk allows for Fixed Mobile Convergence (FMC) applications. Federation with other public IM services and social networking site's IM capability.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Personal group chat rooms</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Every user on the system who has a personal conference bridge assigned also gets a personal group chat room auto configured. Escalating a group chat session to a conference call is easy using the @conf directive.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Personal Assistant</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The [[Personal Assistant]] is an IM bot that is automatically added to a user's buddy list. The personal assistant allows interacting with the system using IM. It can be used as an FMC application where it allows to initiate calls using the corporate dial plan. It allows corporate phone book lookups. And it allows dynamic control of a user's personal conference bridge, where the user can see participant entry and exit messages and the owner of the conference is able to kick, mute, isolate and invite participants. The personal assistant also provides notification of incoming calls and it notifies the user when a caller is in the process of leaving a voicemail message. The user can then choose to either listen in or brage in. Other functionality includes call history and a list of missed calls.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Federation with Google</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Server-to-server XMPP federation between sipXecs and Google Talk allows using the GTalk client as an FMC client for sipXecs on any smartphone for which GTalk is available (includes almost all of them). The GTalk client allows monitoring the presence of extensions on the sipXecs system as buddies in GTalk. If both ends are capable of IM then chat is possible. The Personal Assistant can be added as a buddy into GTalk where it provides all the functionality described above acting as an FMC client. Such server-to-server federation is possible with any XMPP server, including Google Talk.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">IM Federation</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Federation with most public IM systems is possible including Yahoo, AIM, MSN, GTalk, IBM Sametime, ICQ, IRC, Facebook IM, MySpace IM, GaduGadu. Federation is done server side so that buddies from different IM networks can be put together into a chat room. Also, message archiving still works for all chat connections independent of network if that is desired. The users are able to manage their credentials for public IM systems using the User Portal. This means that a standard XMPP client can be used (Pidgin, Counterpath Bria, Trillian, Spark, etc) and all the buddies from all the networks will show up in this client.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dynamic call routing based on presence</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Presence based dynamic call routing allows changing the user's find-me / follow-me rules dynamically by setting a custom IM presence state. For example: If the users includes a phone number into the custom presence state, then calls are automatically also forwarded (parallel forked) to this number. This is very handy when in a temporary office or in a hotel room. If the user's IM presence state is set to DND, then calls automatically divert to voicemail directly. </td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dimdim Integration</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Dimdim integration brings Web conferencing to sipXecs (screen sharing, white-boarding, presentations, and shared Web surfing. The Dimdim server runs as a component of a sipXecs cluster, centrally managed. Web conferences can be combined with audio and video conferencing provided by sipXecs.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Voicemail System</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The main objective with the new voicemail system is scalability. It is expected to offer between 5x and 10x better scale as compared to the old voicemail system. In addition, it supports HD audio and mixing between narrow band and wide band. The new voicemail system includes an IMAP interface that will turn the voicemail server into an IMAP client. This allows using a standard email server, including Microsoft Exchange, Lotus Notes, Novell GroupWise, Yahoo Zimbra and others as the central message store for voicemail, offering unified messaging.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Centralized Voicemail</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs system can act as a Centralized Voicemail for a Legacy PBX. This allows Message Waiting Indication to be sent to phones on the Legacy PBX. See the [[Centralized Voicemail]] page for more info.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Bridged (Shared) Line Appearances (BLA)</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">There is still quite a bit of uncertainty around the final IETF standard for BLA. Our current plan is to implement BLA according to what Polycom supports in their current firmware for SoundPoint and SoundStation phones. This is identical or close to BLA as implemented by Broadsoft. See the [[Bridged Line Appearance]] page for more info.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Auto Attendant Improvements</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">There is a list of pending improvements to the current auto-attendant and voicemail system. Among them an improved dial-by-name functionality.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Call Detail Record (CDR) Improvements</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The plan is to improve the data set contained in a CDR record with items such as call type, account codes, originating phone info. This allows more comprehensive reporting to be generated.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved User Portal</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The current user portal offers quite a lot of functionality, but from a usability and look & feel perspective leaves quite a few gaps. We are introducing a REST (Web Services) based API for all user portal functionality that allows creating independent widgets around the functionality offered. In addition, we are working on a new implementation of the user portal using this new API. [[Next_Generation_UserPortal|Initial design ideas]].</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Enhanced Directory Services</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release extends the information stored about a user and made available for directory lookups in a significant way. User profiles will become a lot richer, including profile pictures, IM handles, alternative phone numbers, and a location. The possibility to synchronize or share address book data between sipXecs and other applications is going to improve.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Allow Multiple ACD Servers in a Cluster</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This is a requirement we missed in the 4.0 release. In order for the ACD server to scale we need to allow several instances of the ACD server to run on dedicated hardware, centrally managed as part of the sipXecs cluster.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">E911 Notification by SMS</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This is also a feature we missed in the last release. The alarm server introduced in release 4.0 can send email when an emergency number is dialed. This capability extends this to SMS.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">ACD Wallboard</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Pawel provided an ACD wallboard application implementing a PHP API on top of the SOAP interface offered by the ACD server. The idea is to integrate the ACD wallboard directly into the user portal.</td>
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== Stable Release 4.0 ==
'''SIP trunking gateway and near-end / far-end NAT traversal for remote workers'''
For release 4.0 we are adding a fully functional native SIP trunking gateway that adds SIP trunking capabilities to sipXecs without requiring an external Session Border Controller (SBC). As a further objective we want to implement the SIP Forum SIPconnect standard for interconnection with ITSPs. [[SipXbridge Functional Requirements]] outlines the requirements we are trying to address in this project. See issue [http://track.sipfoundry.org/browse/XECS-1014 XECS-1014] and the [[SipXbridge_Overview_and_Configuration|SIP trunking Wiki page]].
'''Cluster Management'''
In release 4.0 sipXconfig will learn how to fully manage a distributed cluster. Such a cluster consists of several call control servers in high-availability load-sharing configuration combined with application servers for media services, conferencing, call center ACD, etc. All these applications can either run on a single server or be distributed to run on separate HW. This will allow sipXecs to be deployed as a multi-branch office solution that is fully centrally managed and acts as one big system with a cohesive dial plan and number portability between branch offices. Scalability should then extend into several 10,000 of users distributed over different locations / offices.
'''Release Timing'''
We originally thought that release 4.0 would become available towards the end of 2008. It however took a little longer than we wanted. Release was on April 28, 2009.
<center>
<table width="90%" cellspacing="0" cellpadding="5" style="border:1px solid #c2c2c2;">
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 <th><big>Release 3.11 / 4.0</big></th>
 <th>Comments</th>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">SIP Trunking Gateway</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXbridge project adds a new component to sipXecs to enable native SIP trunking and NAT traversal. sipXbridge is based on a B2BUA design able to anchor media and tweak SIP signaling so that it can traverse NAT. sipXbridge is integrated into sipXconfig as a managed SBC. [http://track.sipfoundry.org/browse/XECS-1192 XECS-1192], [http://track.sipfoundry.org/browse/XECS-1014 XECS-1014], [http://track.sipfoundry.org/browse/XCF-2237 XCF-2237]. As all the other sipXecs components, sipXbridge can run independently either on the same server hardware with other components or on its dedicated server. sipXbridge anchors media and the media anchoring can be configured in a redundant setup where each of the redundant proxy servers provides its own media relay.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Near-end / far-end NAT traversal support in the proxy</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs proxy server will natively support near-end and far-end NAT traversal in order to support remote workers and remote branch offices connected without a VPN. This includes support for PATH header RFC 3327. [http://track.sipfoundry.org/browse/XECS-484 XECS-484], [http://track.sipfoundry.org/browse/XECS-265 XECS-265]. The NAT traversal capability is directly integrated into the sipXecs proxy server so that it can auto-detect dynamically whether an end point requires NAT traversal assistance or not. A media relay is added to each proxy for anchoring the media as necessary. The NAT traversal capability also works in a redundant system offering an HA configuration.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Conferencing Server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs and FreeSWITCH projects cooperate to integrate FreeSWITCH as a conferencing server into sipXecs. Full plug & play management is provided for users creating and administering their conferences. See [[Conferencing_Service_for_sipXecs|here for more details]]. We are aiming for over 500 conferencing ports on regular hardware, support for different codecs, dynamic conference controls using DTMF codes or the sipXconfig user portal. The conferencing bridge is ready to be speech enabled with TTS, allows wideband conferences, and will eventually support video.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New IVR and Auto-Attendant Server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs and FreeSWITCH cooperation also led to a new IVR server based on FreeSWITCH. The underlying media server engine is used from FreeSWITCH and the sipXecs project created a new Java based IVR frontend for easy application writing. The first application using this new capability is a complete rewrite of the original sipXecs Auto-Attendant. The immediate result is significantly improved performance consuming fewer compute resources.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Click-to-dial support from the Directory in the User Portal</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The directory on the user portal becomes interactive offering click-to-dial using Third Party Call Control (3PCC). The user can enter a phone number or SIP URI and initiate a call from any phone the user has currently registered with the system. The same click-to-dial capability is used to add conference participants to an already ongoing conference using outbound dialing.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Import / export contacts using vcards</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The directory on the user portal now allows importing or exporting contacts in vcard format.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management for Counterpath softphones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Counterpath softphones will be plug & play configured using a provisioning server as part of sipXecs. [http://track.sipfoundry.org/browse/XCF-2022 XCF-2022].</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">64-bit support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The 64-bit branch is going to be merged with main rendering a unified code base to support 32-bit and 64 bi architectures using Intel or PPC CPUs. [http://track.sipfoundry.org/browse/XECS-480 XECS-480].</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Source call routing</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">There are two areas where we are planning to enhance the flexibility of the dialplan: a) Gateway selection based on who is calling for outbound calls [http://track.sipfoundry.org/browse/XECS-415 XECS-415], and b) Source routing attendant able to route calls based on incoming Caller ID [http://track.sipfoundry.org/browse/XECS-1083 XECS-1083]. These capabilities aim at improving flexibility in multi-branch deployments of sipXecs.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Cluster management</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig will become able to centrally manage a distributed cluster of sipXecs components, including high-availability configurations [http://track.sipfoundry.org/browse/XCF-2133 XCF-2133]. Ease of use for system installation and administration is the primary objective. A distributed system of sipXecs servers will allow very easy setups of multi-branch configurations. Also, sipXecs easily scales adding additional load-sharing call servers, or configuring separate servers for certain media services such as voicemail, ACD or conferencing.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Integrated advanced reporting</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Jasper Reports, like Crystal Reports, is a powerful reporting application that we plan to integrate into sipXconfig [http://track.sipfoundry.org/browse/XCF-2286 XCF-2286]. Reports can be customized and all reports are generated in several formats.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Polycom phones w/ MoH support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Support for the Polycom 3.1.3 firmware and new phones SoundPoint IP 560 & 670, and SoundStation IP 6000 & 7000. The Polycom firmware 3.1 was developed in close cooperation with sipXecs and now fully supports Music on Hold (MoH). In addition, Polycom added specific fixes to the BLF functionality that resolved outstanding issues for certain call flows.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Snom phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Support for the Snom firmware 7.x was added. This required a change of the config file format to XML.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management support for Aastra phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We are adding a plugin to support the new Aastra 5-Series phones [http://track.sipfoundry.org/browse/XCF-2193 XCF-2193].</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Grandstream</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The Grandstream plugin has been updated to support new phones and new firmware revisions. New phones include the full line of GXP phones.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Linksys phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The Linksys plugin has been updated to include support for new phones.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Cisco phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The Cisco plugin has been updated to include support for new phones.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Alarm Server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release adds a new alarm server that can be configured via sipXconfig. It collects system alarms of various severity levels and distributes these alarms to whoever needs to know.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Web Certificate management</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig is now able to manage Web certificates needed for secure (https) access to its admin and user portals. A Certificate Signing Request (CSR) can be easily generated and an official certificate can be uploaded using the Web interface. This gets rid of the security alert messages now seen in most browsers.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Time and DST management</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">To prevent glitches during daylight savings time changes, sipXconfig now provides the ability to manage time and DST changes as well as the way these parameters are updated in the phones. The result is always correct time.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Scheduled device (phone) reboot</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Phones need to be rebooted for profile changes to become active. However, during the day they might be in use and a reboot is undesired. sipXconfig now allows these reboots to be scheduled for after-hours.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved backup & restore with FTP option</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The backup and restore mechanism is enhanced. An FTP option is offered directly from the sipXconfig UI in addition to backups on the local machine or backups sent by email.</td>
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</center>
== Stable Release 3.10 ==
Release 3.10 was released GA end of March. In April a first maintenance release, 3.10.1, was release with a second one, 3.10.2, in June. As always, provide feedback on the project's users list.
'''Release Notes:''' [[Detailed Release Notes for sipXecs Release 3.10]]
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 <th><big>Release 3.9 / 3.10</big></th>
 <th>Comments</th>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New (nicer) skin</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Even easier installation / device discovery</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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,</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Extended User Portal / time based find-me / follow-me</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Personal auto-attendant / Individual zero-out capability</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Import from and export to Excel</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Localization of the Media Server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Busy Lamp Field (BLF) and Presence</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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).</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Integration with Microsoft</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Time-Based Routing</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Paging Server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Based on the [[Functional_Specification_for_a_Paging_Server|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)</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Overhaul of the ACD server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improvements to the Auto-Attendant</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improvements to Hunt Groups</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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 [http://track.sipfoundry.org/browse/XCF-831 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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Overhaul of the security and authentication system</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved E911 call routing</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Simplified dial plan configuration</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Registered phones displayed per user</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New device category: SBC</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Automated restore from backup</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The current restore from backup functionality will be integrated into Config Server.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Server and application statistics, reports, and alarms</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Support for new Polycom 320 / 330 phones / Polycom 2.2.2 firmware</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Support for Linksys Phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We are adding support for Linksys SPA941 and SPA942 phones fully integrated into the sipXconfig management system thanks to a community contribution.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Support for IpDialog SipTone V Phone</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We are adding support for the IpDialog SipTone V phone fully integrated into the sipXconfig management system thanks to a community contribution.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Support for LG-Nortel 1535 Video Phone</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Report: Login history</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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).</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Symmetric signaling / merged proxy</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">SIP loop detection</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Port to PowerPC (PPC)</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXecs was ported to the PowerPC (PPC) platform with all the big endian handling for audio processing and other issues.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Port to FreeBSD</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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 [http://track.sipfoundry.org/browse/XECS-108 XECS-108] for the port files and [[FreeBSD port for sipXecs 3.10]] for documentation..</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New XML RPC process management API</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>
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</center>
== Stable Release 3.8 ==
Release 3.8 is focused on improving SIP Trunking capabilities as well as support for directory, speed dial and BLF on the phone.
'''Release Notes:''' [[Detailed Release Notes for sipXecs Release 3.8]]
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 <th><big>Release 3.7 / 3.8</big></th>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New ACD Call Center Server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The ACD Call Center server has been a closed source software up to release 3.6 and was made available into open source in the course of the 3.8 development cycle. This call center ACD server serves up to 50 agents with several queues. It is typically used as an informal call center for IT helpdesks and other applications that require management of calls in a queue.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management support for Audiocodes Gateways</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This is a major milestone for sipX as we finally add full plug & play management support for all Audiocodes gateways. This means that gateways are managed in a very similar way as compared to phones. All configuratoin is generated by sipXconfig, where sipXconfig chooses default parameters where possible to render a working config out of the box. The gateway then picks up these generated profiles from the sipX server. We plan to support all Audiocodes gateways with initial focus on the following models: MP-114 FXS and FXO, MP-118 FXO and FXS, MP-124 FXS, Mediant 1000. TP-260 and Mediant 2000 are priority two.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management support for LG-Nortel phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.8 adds plug & play management support for LG-Nortel phones 6804, 6812, and 6830. These phones support standards based Music on Hold (MoH).</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Voicemail Portal</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The voicemail portal used by users to retrieve and manage voicemail messages from a Web browser has always been a separate application that required a separate login. We are now integrating the voicemail portal into the user poral of Config Server. Going forward only one user login will be required and the user will be able to manage all user configurable aspects of the phone system including voicemail from there. That includes configuration of forwarding rules and speed dial entries. This represents the first step towards separating the Media Server from the rest of the system. Once done, the system will support several Media Servers on separate HW and all centrally managed by Config Server.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Automated Configuration of HA Slave Systems</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This relates to a further simplification of the installation process. Certificates can now be distributed to the Slave server in an HA configuration automatically during the installation process. Config Server manages the Slave system remotely with the ability to enable and disable services on the remote Slave host.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Phone Directory Support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Depending on the phone model it is possible to load directory information into the phone. Release 3.8 will provide a capability to generate a corporate directory based on the user database in sipX augmented by a file import capability using .csv files. This information will be compiled into a directory that can be loaded by the phone. Inclusion into the directory is controlled by group membership as well as a specific permission flags that allows for inclusion.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Extended support for Localization</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig can be skinned and localized so that the presented language dependes on the users browser settings. sipXconfig is being extended to allow for full localization to be done in .properties files. In addition, the Polycom phone model is extended to support phone localization.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Speed Dial Support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In addition to directory information we plan on supporting the user specific configuration of speed dial keys (soft key assignments on the phone). The user will be able to add individual speed dial assignments using the user portal of sipXconfig.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Busy Lamp Field (BLF)</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We talked about BLF several times and we remain serious about it. Polycom has changed their BLF implementation several times now across different versions of firmware, which made it difficult to follow a straight course. We now decided to implement a sipX presence server based on dialog events. This presence server will collect status information from phones that offer it and allow subscription to such information. A centralized solution is harder to implement, but it is more economical in terms of network bandwidth requirements and it will form the basis for more extensive implementations of presence based services such as interconnection to IM systems such as Jabber and Microsoft LCS.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Domain routing with wildcards</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">As a last minute item we are adding improved SIP domain routing capabilities to release 3.8. This will allow domain based routing (including wildcards to define domain names). Calls to different domains (i.e different SIP trunking providers) can be routed along different routes.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">ISN (ITAD) Signalling</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">[http://freenum.org/ ISN signalling] is a new way of bypassing the PSTN. ISN provides an easy way for campuses, enterprises, and ASPs to acquire globally-unique subscriber numbers to support new communications services. ISNs are free and they provide a domain-based, "Internet-style" number that looks more like an email address than a traditional E.164 telephone number. An ISN is formed by joining a domain-local subscriber number to an ITAD (Internet Telephony Administrative Domain) number, using an asterisk as the delimiter. For example, subscriber 1234 in ITAD 256 would have ISN: 1234*256.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">ENUM Signalling</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">ENUM will be supported as an additional redirector plugin, configurable using sipXconfig. ENUM allows the automatic routing of calls over an IP netwrok provided that for the dialed PSTN number there is an IP address equivalent defined in an ENUM registry database. Several ENUM registries can be queried.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Redirector Plugins</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Redirector plugins provide a simple mechanism to add redirectors at start time using a simple API. A redirector implements a specific routing rule that is considered as sipX evaluates the dial plan everytime a session is initiated. ISN signalling is implemented as a redirector. ENUM is another redirector. More common dialing rules are now also implemented as redirectors so that with release 3.8 we will have about 15 redirectors in the system already. More exotic redirectors can be added easily. For example: A redirctor could use a database to map every dialed number or URI to a specific other number or URI.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">CDR Reports</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Since release 3.4 sipX supports CDR data collection for both non-redundant and HA systems. We plan on improving CDR reporting by adding a report generation mechanism that extracts the data from the database and presents it in a user friendly way. The entire CDR post-processing part is re-written to enable real-time reporting of calls. A screen inside sipXconfig will display calls as the terminate, automatically refreshing the windows in a given interval. CDR reports can then be exported to a spreadsheet.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Real-time view of ongoing calls</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In addition to near real-time reporting of CDRs for completed calls, it will be possible to see what calls are currently in process using sipXconfig. The CallResolver process is extended with a SOAP interface that allows querying currently active calls. This information will be displayed by sipXconfig.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dialplan Localization</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig will support the automatic switching and re-initialization of a localized dialplan. sipx easily supports different dialplans that can be localized both with respect to a country's or regions dialplan requirements as well as language.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Support for Grandstream GXV-3000 Video Phone</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We now support plug & play configuration management of the Grandstream GXV-3000 video phone, which means that we now have complete suppport for all the Grandstream phones and TAs. Thanks to IIPS for their help. Grandstream still does not support dialog events in their phones, so that certain features such as call park and call pickup do not work.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated SNOM Configuration Support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Support for plug & play management for SNOM phones got updated. In addition to existing capabilities the SNOM phone model now supports speed dial and directory capabilities.</td>
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</table>
</center>
== Timeframe for Release 3.7 / 3.8 ==
Development release 3.7 will become stable release 3.8. We decided to extend the development phase for release 3.8 until end of February. After the most extensive test and bug fix phase in the history of the project, we are expecting release 3.8 to become stable in June.
== Stable Release 3.6 ==
On October 3 the first beta release of 3.6 was made available on SIPfoundry. Release 3.6 is focused on improving the flexibility of the sipX dial plan based on experience gained through many production deployments. In addition, 3.6 will add additional critical features as outlined below:
<center>
<table width="90%" cellspacing="0" cellpadding="5" style="border:1px solid #c2c2c2;">
 <tr>
 <th><big>Release 3.6</big></th>
 <th>Comments</th>
 </tr>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">LDAP Support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">LDAP support has made it into the 3.6 release. We allow backend synchronization with an LDAP capable directory and upload the relevant information into Config Server. For performance reasons session authentication will still be done internal to sipX. The implementation should be compatible also with Microsoft AD but it has not yet been tested.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Intercom / Paging</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.6 adds an Intercom capability that in a first phase will support point-to-point intercom using Polycom phone's auto-answer capability.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dial Plan Templating</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In support of international deployments it will be possible to create country or region specific dial plans that can be selected within config server. As an example and in addition to the U.S. dial plan there is now a Swiss dial plan as well as a Polish dial plan. Additional dial plans are easily defined in XML and can be added to sipX as part of a language pack.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Domain Alias</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.6 will allow domain aliasing, which improves deployability of sipX in environments based on Microsoft Windows Server, as well as in cases where sipX needs to be responsible for calls from different domains or IP addresses.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Park Server Enhancements</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipX Park Server now includes several additional configurable features. There is a time-out value configurable so that after a defined period of time the parked called is transferred back to the person that parked the call. Also, there is a configurable escape key from park. When pressed the call is transferred back to the person that parked the call. Also, it is now possible to configure whether the system allows several calls to be parked on one park orbit (FILA) or not.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Gateway Configuration improvements</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The PSTN gateway configuration now allows adding a gateway specific prefix before the number is dialed. This allows the accommodation of different number conversion requirements in case several gateways are associated with a single dial plan rule.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''yum'' based Install & Updates</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We plan to eliminate the single file install script and allow installs simply using ''yum''. This is already possible for the Debian build since release 3.0.1.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Support for new Polycom phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release adds support for the new Polycom phones such as the SoundPoint IP430. We also updated the plug & play functionality to including the Polycom 2.0 firmware. The sipX Config Server now supports mixed deployment with Polycom phones on the 1.x firmware release and phones already on the new 2.0 firmware release.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''FTP Server support for phone management</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In addition to the TFTP server sipX now comes standard with a configured FTP server. The FTP server provides access to the same configuration directory used by the TFTP server for phones capable of using FTP instead of TFTP. This is especially convenient for Polycom phones that come factory configured to use FTP.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Support for Hitachi Cable WiFi Phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release will support the Hitachi Cable IP 5000 and IP 3000 WiFi phone in Config Server.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''SIP Trunk support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We added an option to add a SIP destination as a trunking gateway. This SIP trunking gateway is selectable from the dial plan in the same way a PSTN gateway is selected. A route header field allows the definition of a Session Border Controller (SBC) used to route the call across NAT / Firewall.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Custom Permissions</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipX already offered a flexible mechanism to use different permissions in dial plan rules. We now added the ability to define additional custom permissions that are administrered by the admin and used in the same way built in permissions are.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Caller ID manipulation (CLID / CLIR)</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We greatly extended the ability to manipulate and define Caller ID for outbound calls. This allows a much more flexible mapping of User ID to CLID on a per user, per user group as well as on a per gateway basis. In addition we added Caller ID Restriction (CLIR) on a per user, per user group and per gateway basis. The User definition now includes, in addition to User ID and Aliases, a line to define outgoing caller ID. This makes it possible to have e.g. an alpha-numeric or 4 digit local extension as your User ID while still send the full DID number of the users as caller ID to the PSTN.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Updated SNOM phone support</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release updated support for the Snom phones to firmware release 6.2.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Music On Hold (MOH) for Snom phones</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Snom phones are the first to support the new IETF standard for Music On Hold (MoH). Release 3.6 provides an IETF standard compliant music source to which the Snom phones can transfer a call when hold is pressed. This provides for a scalable implementation of MoH. A music file can be uploaded from Config Server.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Performance improvement of the media server</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We were able to work on some performance improvements for the sipX media server (voice mail subsystem). With that improvement we should now be able to support more virtual media server ports.</td>
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</table>
</center>
'''Note:''' We missed on the BLF feature in 3.6. We will try and make good on that in release 3.8.
== Timeframe for Release 3.6 ==
Release 3.6 BETA was made available October 3rd. We expect a stable version by mid November.
== Stable Release (3.4) ==
The objective with release 3.4 is to add CDR recording and make it available in a stable release as quickly as possible. CDR recording is a key feature for most of the larger deployments of sipX.
<center>
<table width="90%" cellspacing="0" cellpadding="5" style="border:1px solid #c2c2c2;">
 <tr>
 <th><big>Release 3.4</big></th>
 <th>Comments</th>
 </tr>
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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">CDR Reports</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Call Detail Records (CDRs) will be [http://scm.sipfoundry.org/viewsvn/checkout/sipX/branches/3.4/sipXproxy/doc/cdr/cse_database_design.txt stored in a PostgreSQL database]. This database will reside either on the sipX host or a different dedicated host. The design is optimized for performance. An additional CDR report generator (future) will allow the creation of custom reports. Third party report generatrors may be used to customize views based on raw database information.
The CDR Database will also support the new HA configuration with redundant proxy and registrar servers. Call State Events will be collected from all the proxy and registrar servers and reporting will take into account cases where a call starts on one proxy but terminates on another because of a failover condition.
</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Auto-Attendant Improvements</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Additional configuration parameters were added to the Auto-Attendant: Inter-digit timeout, overall DTMF timeout value, and maximum length of DTMFs.</td>
 </tr>
</table>
</center>
== Timeframe for Release 3.4 ==
Release 3.4 was released stable in July 2006. 
== Stable Release (3.2) ==
<center>
<table width="90%" cellspacing="0" cellpadding="5" style="border:1px solid #c2c2c2;">
 <tr>
 <th><big>Release 3.2</big></th>
 <th>Comments</th>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">High Availability</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In sipXpbx, basic calling depends on three components: the two proxies and the registrar/redirect service. The proxies can be replicated and DNS SRV records can be used to share load and provide for failover. The registrar/redirect service, however, cannot currently be deployed on multiple servers because the 'soft' state in the registry database (mappings from registered Addresses to Contacts) cannot be shared. While replicating the proxies alone does help with scaling, the registrar is a single point of failure for basic calling service. In release 3.2 we are planning to introduce high-availability for basic calling features by adding the required replication of registration information.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated SOAP Interface</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The SOAP interface first became available in release 2.8 and was then dropped in release 3.0 because of the re-design of the Config Server. Release 3.2 will re-introduce the SOAP interface. For [http://sipx-wiki.calivia.com/index.php/SipX_ConfigServer_SOAP_API more details on the API].</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Simpler Upgrades</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipX team set out to develop a solution to simplify upgrades. Data migration for configuration data is done automatically as part of the release upgrade process. With 3.2 therefore, the user will be able to install 3.2 over 3.0.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Improvements</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Already release 2.8 introduced plug & play management of phones. Release 3.0 enhanced that capability significantly providing support for many additional phones, including the first gateway, as
well as provide a simple XML-based framework to add support for new phones and gateways. Release 3.2 will add the capability to manage firmware upgrades using the sipX Configuration Server. 3.2 will also have an update to the Polycom phone templates. All testing will be conducted with 3.1 boot rom and 1.6.4 SIP firmware.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved Logging</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.2 will introduce snapshot logs that can be easily created and downloaded using the Web interface. Eventually we would like to introduce much more comprehensive server and application performance management.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Time of Day/Day of Week Auto Attendant Routing</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The 3.2 Auto Attendant has been enhanced to support a time-based routing feature. Auto Attendants can be configured to answer based on working hours, closed hours or holidays. A system overide is also available that can be envoked from a remote telephone in cases where the office may be closed unexpectedy -- snow days etc.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">CSV Import Tool ("Cut Sheet")</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Administrators will be able to bulk load users, user groups, user aliases, phones, phone MAC addresses, phone types, phone groups and group settings, </td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved Navigation</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In 3.2 the navigation structure is now configurable with a CSS. A new search feature has also been added that indexes users, phones and all associated settings.</td>
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 <td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved Performance</td>
 <td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Registration as well as subscribe / notify performance has been significantly imporved. In installations with large numbers of phones, transactions generated by re-registration and subscribe / notify can far exceed the actuall call rate. sipX release 3.2 is tested up to 5,000 users for adequate re-registration and subscribe / notify performance.</td>
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</table>
</center>
== Timeframe for Release 3.2 ==
Release 3.2 was released stable in April 2006. 
=sipXecs - Unified Communications System for the Enterprise (IP PBX) - Roadmap=

This page is intended to provide some information around the project team's

thinking for new things coming up. We would like to solicit input and get some

feedback on our current direction, so please post comments to the sipX users

list.

== Proposed Release 4.2 ==

===Other projects integrated===

For release 4.2 we will be looking to integrate other exciting open source projects. Some examples:

*Openfire Integration

*Dimdim Integration

*Deeper FreeSWITCH Integration with new voice mail

===Release Timing===

See [[When will release X be available?]]

===Feature List===

<tr>

<th><big>Release 4.1 / 4.2</big></th>

<th>Comments</th>

</tr>

<tr>

<td colspan="2">For detailed status, see the [http://track.sipfoundry.org/browse/XX/fixforversion/10552 4.2 tracker road map]</td>

</tr>

<tr>

<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Openfire Integration</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release adds [[Instant Messaging and Presence for sipXecs]]. The Openfire XMPP server runs as a component of a sipXecs cluster with centrally managed configuration. Users created on the SIP side automatically synchronize with the XMPP side. Presence federation between phones (BLF presence) and IM presence allows seeing when a contact is on the phone. Presence based routing allows call handling based on presence state. Phones, such as the Counterpath Bria softphone, are auto-configured to include both SIP and XMPP capabilities. Federation with Google Talk allows for Fixed Mobile Convergence (FMC) applications. Federation with other public IM services and social networking site's IM capability.</td>

</tr>

<tr>

</tr>

<tr>

<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Personal group chat rooms</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Every user on the system who has a personal conference bridge assigned also gets a personal group chat room auto configured. Escalating a group chat session to a conference call is easy using the @conf directive.</td>

</tr>

<tr>

</tr>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Personal Assistant</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The [[Personal Assistant]] is an IM bot that is automatically added to a user's buddy list. The personal assistant allows interacting with the system using IM. It can be used as an FMC application where it allows to initiate calls using the corporate dial plan. It allows corporate phone book lookups. And it allows dynamic control of a user's personal conference bridge, where the user can see participant entry and exit messages and the owner of the conference is able to kick, mute, isolate and invite participants. The personal assistant also provides notification of incoming calls and it notifies the user when a caller is in the process of leaving a voicemail message. The user can then choose to either listen in or brage in. Other functionality includes call history and a list of missed calls.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Federation with Google</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Server-to-server XMPP federation between sipXecs and Google Talk allows using the GTalk client as an FMC client for sipXecs on any smartphone for which GTalk is available (includes almost all of them). The GTalk client allows monitoring the presence of extensions on the sipXecs system as buddies in GTalk. If both ends are capable of IM then chat is possible. The Personal Assistant can be added as a buddy into GTalk where it provides all the functionality described above acting as an FMC client. Such server-to-server federation is possible with any XMPP server, including Google Talk.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">IM Federation</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Federation with most public IM systems is possible including Yahoo, AIM, MSN, GTalk, IBM Sametime, ICQ, IRC, Facebook IM, MySpace IM, GaduGadu. Federation is done server side so that buddies from different IM networks can be put together into a chat room. Also, message archiving still works for all chat connections independent of network if that is desired. The users are able to manage their credentials for public IM systems using the User Portal. This means that a standard XMPP client can be used (Pidgin, Counterpath Bria, Trillian, Spark, etc) and all the buddies from all the networks will show up in this client.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dynamic call routing based on presence</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Presence based dynamic call routing allows changing the user's find-me / follow-me rules dynamically by setting a custom IM presence state. For example: If the users includes a phone number into the custom presence state, then calls are automatically also forwarded (parallel forked) to this number. This is very handy when in a temporary office or in a hotel room. If the user's IM presence state is set to DND, then calls automatically divert to voicemail directly. </td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dimdim Integration</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Dimdim integration brings Web conferencing to sipXecs (screen sharing, white-boarding, presentations, and shared Web surfing. The Dimdim server runs as a component of a sipXecs cluster, centrally managed. Web conferences can be combined with audio and video conferencing provided by sipXecs.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Voicemail System</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The main objective with the new voicemail system is scalability. It is expected to offer between 5x and 10x better scale as compared to the old voicemail system. In addition, it supports HD audio and mixing between narrow band and wide band. The new voicemail system includes an IMAP interface that will turn the voicemail server into an IMAP client. This allows using a standard email server, including Microsoft Exchange, Lotus Notes, Novell GroupWise, Yahoo Zimbra and others as the central message store for voicemail, offering unified messaging.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Centralized Voicemail</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs system can act as a Centralized Voicemail for a Legacy PBX. This allows Message Waiting Indication to be sent to phones on the Legacy PBX. See the [[Centralized Voicemail]] page for more info.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Bridged (Shared) Line Appearances (BLA)</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">There is still quite a bit of uncertainty around the final IETF standard for BLA. Our current plan is to implement BLA according to what Polycom supports in their current firmware for SoundPoint and SoundStation phones. This is identical or close to BLA as implemented by Broadsoft. See the [[Bridged Line Appearance]] page for more info.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Auto Attendant Improvements</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">There is a list of pending improvements to the current auto-attendant and voicemail system. Among them an improved dial-by-name functionality.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Call Detail Record (CDR) Improvements</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The plan is to improve the data set contained in a CDR record with items such as call type, account codes, originating phone info. This allows more comprehensive reporting to be generated.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved User Portal</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The current user portal offers quite a lot of functionality, but from a usability and look & feel perspective leaves quite a few gaps. We are introducing a REST (Web Services) based API for all user portal functionality that allows creating independent widgets around the functionality offered. In addition, we are working on a new implementation of the user portal using this new API. [[Next_Generation_UserPortal|Initial design ideas]].</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Enhanced Directory Services</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release extends the information stored about a user and made available for directory lookups in a significant way. User profiles will become a lot richer, including profile pictures, IM handles, alternative phone numbers, and a location. The possibility to synchronize or share address book data between sipXecs and other applications is going to improve.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Allow Multiple ACD Servers in a Cluster</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This is a requirement we missed in the 4.0 release. In order for the ACD server to scale we need to allow several instances of the ACD server to run on dedicated hardware, centrally managed as part of the sipXecs cluster.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">E911 Notification by SMS</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This is also a feature we missed in the last release. The alarm server introduced in release 4.0 can send email when an emergency number is dialed. This capability extends this to SMS.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">ACD Wallboard</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Pawel provided an ACD wallboard application implementing a PHP API on top of the SOAP interface offered by the ACD server. The idea is to integrate the ACD wallboard directly into the user portal.</td>

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</table>

</center>

== Stable Release 4.0 ==

'''SIP trunking gateway and near-end / far-end NAT traversal for remote workers'''

For release 4.0 we are adding a fully functional native SIP trunking gateway that adds SIP trunking capabilities to sipXecs without requiring an external Session Border Controller (SBC). As a further objective we want to implement the SIP Forum SIPconnect standard for interconnection with ITSPs. [[SipXbridge Functional Requirements]] outlines the requirements we are trying to address in this project. See issue [http://track.sipfoundry.org/browse/XECS-1014 XECS-1014] and the [[SipXbridge_Overview_and_Configuration|SIP trunking Wiki page]].

'''Cluster Management'''

In release 4.0 sipXconfig will learn how to fully manage a distributed cluster. Such a cluster consists of several call control servers in high-availability load-sharing configuration combined with application servers for media services, conferencing, call center ACD, etc. All these applications can either run on a single server or be distributed to run on separate HW. This will allow sipXecs to be deployed as a multi-branch office solution that is fully centrally managed and acts as one big system with a cohesive dial plan and number portability between branch offices. Scalability should then extend into several 10,000 of users distributed over different locations / offices.

'''Release Timing'''

We originally thought that release 4.0 would become available towards the end of 2008. It however took a little longer than we wanted. Release was on April 28, 2009.

<tr>

<th><big>Release 3.11 / 4.0</big></th>

<th>Comments</th>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">SIP Trunking Gateway</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXbridge project adds a new component to sipXecs to enable native SIP trunking and NAT traversal. sipXbridge is based on a B2BUA design able to anchor media and tweak SIP signaling so that it can traverse NAT. sipXbridge is integrated into sipXconfig as a managed SBC. [http://track.sipfoundry.org/browse/XECS-1192 XECS-1192], [http://track.sipfoundry.org/browse/XECS-1014 XECS-1014], [http://track.sipfoundry.org/browse/XCF-2237 XCF-2237]. As all the other sipXecs components, sipXbridge can run independently either on the same server hardware with other components or on its dedicated server. sipXbridge anchors media and the media anchoring can be configured in a redundant setup where each of the redundant proxy servers provides its own media relay.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Near-end / far-end NAT traversal support in the proxy</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs proxy server will natively support near-end and far-end NAT traversal in order to support remote workers and remote branch offices connected without a VPN. This includes support for PATH header RFC 3327. [http://track.sipfoundry.org/browse/XECS-484 XECS-484], [http://track.sipfoundry.org/browse/XECS-265 XECS-265]. The NAT traversal capability is directly integrated into the sipXecs proxy server so that it can auto-detect dynamically whether an end point requires NAT traversal assistance or not. A media relay is added to each proxy for anchoring the media as necessary. The NAT traversal capability also works in a redundant system offering an HA configuration.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Conferencing Server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs and FreeSWITCH projects cooperate to integrate FreeSWITCH as a conferencing server into sipXecs. Full plug & play management is provided for users creating and administering their conferences. See [[Conferencing_Service_for_sipXecs|here for more details]]. We are aiming for over 500 conferencing ports on regular hardware, support for different codecs, dynamic conference controls using DTMF codes or the sipXconfig user portal. The conferencing bridge is ready to be speech enabled with TTS, allows wideband conferences, and will eventually support video.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New IVR and Auto-Attendant Server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipXecs and FreeSWITCH cooperation also led to a new IVR server based on FreeSWITCH. The underlying media server engine is used from FreeSWITCH and the sipXecs project created a new Java based IVR frontend for easy application writing. The first application using this new capability is a complete rewrite of the original sipXecs Auto-Attendant. The immediate result is significantly improved performance consuming fewer compute resources.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Click-to-dial support from the Directory in the User Portal</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The directory on the user portal becomes interactive offering click-to-dial using Third Party Call Control (3PCC). The user can enter a phone number or SIP URI and initiate a call from any phone the user has currently registered with the system. The same click-to-dial capability is used to add conference participants to an already ongoing conference using outbound dialing.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Import / export contacts using vcards</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The directory on the user portal now allows importing or exporting contacts in vcard format.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management for Counterpath softphones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Counterpath softphones will be plug & play configured using a provisioning server as part of sipXecs. [http://track.sipfoundry.org/browse/XCF-2022 XCF-2022].</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">64-bit support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The 64-bit branch is going to be merged with main rendering a unified code base to support 32-bit and 64 bi architectures using Intel or PPC CPUs. [http://track.sipfoundry.org/browse/XECS-480 XECS-480].</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Source call routing</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">There are two areas where we are planning to enhance the flexibility of the dialplan: a) Gateway selection based on who is calling for outbound calls [http://track.sipfoundry.org/browse/XECS-415 XECS-415], and b) Source routing attendant able to route calls based on incoming Caller ID [http://track.sipfoundry.org/browse/XECS-1083 XECS-1083]. These capabilities aim at improving flexibility in multi-branch deployments of sipXecs.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Cluster management</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig will become able to centrally manage a distributed cluster of sipXecs components, including high-availability configurations [http://track.sipfoundry.org/browse/XCF-2133 XCF-2133]. Ease of use for system installation and administration is the primary objective. A distributed system of sipXecs servers will allow very easy setups of multi-branch configurations. Also, sipXecs easily scales adding additional load-sharing call servers, or configuring separate servers for certain media services such as voicemail, ACD or conferencing.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Integrated advanced reporting</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Jasper Reports, like Crystal Reports, is a powerful reporting application that we plan to integrate into sipXconfig [http://track.sipfoundry.org/browse/XCF-2286 XCF-2286]. Reports can be customized and all reports are generated in several formats.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Polycom phones w/ MoH support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Support for the Polycom 3.1.3 firmware and new phones SoundPoint IP 560 & 670, and SoundStation IP 6000 & 7000. The Polycom firmware 3.1 was developed in close cooperation with sipXecs and now fully supports Music on Hold (MoH). In addition, Polycom added specific fixes to the BLF functionality that resolved outstanding issues for certain call flows.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Snom phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Support for the Snom firmware 7.x was added. This required a change of the config file format to XML.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management support for Aastra phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We are adding a plugin to support the new Aastra 5-Series phones [http://track.sipfoundry.org/browse/XCF-2193 XCF-2193].</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Grandstream</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The Grandstream plugin has been updated to support new phones and new firmware revisions. New phones include the full line of GXP phones.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Linksys phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The Linksys plugin has been updated to include support for new phones.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated plug & play support for Cisco phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The Cisco plugin has been updated to include support for new phones.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Alarm Server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release adds a new alarm server that can be configured via sipXconfig. It collects system alarms of various severity levels and distributes these alarms to whoever needs to know.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Web Certificate management</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig is now able to manage Web certificates needed for secure (https) access to its admin and user portals. A Certificate Signing Request (CSR) can be easily generated and an official certificate can be uploaded using the Web interface. This gets rid of the security alert messages now seen in most browsers.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Time and DST management</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">To prevent glitches during daylight savings time changes, sipXconfig now provides the ability to manage time and DST changes as well as the way these parameters are updated in the phones. The result is always correct time.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Scheduled device (phone) reboot</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Phones need to be rebooted for profile changes to become active. However, during the day they might be in use and a reboot is undesired. sipXconfig now allows these reboots to be scheduled for after-hours.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved backup & restore with FTP option</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The backup and restore mechanism is enhanced. An FTP option is offered directly from the sipXconfig UI in addition to backups on the local machine or backups sent by email.</td>

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</table>

</center>

== Stable Release 3.10 ==

Release 3.10 was released GA end of March. In April a first maintenance release, 3.10.1, was release with a second one, 3.10.2, in June. As always, provide feedback on the project's users list.

'''Release Notes:''' [[Detailed Release Notes for sipXecs Release 3.10]]

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<th><big>Release 3.9 / 3.10</big></th>

<th>Comments</th>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New (nicer) skin</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Even easier installation / device discovery</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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,</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Extended User Portal / time based find-me / follow-me</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Personal auto-attendant / Individual zero-out capability</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Import from and export to Excel</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Localization of the Media Server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Busy Lamp Field (BLF) and Presence</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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).</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Integration with Microsoft</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Time-Based Routing</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Paging Server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Based on the [[Functional_Specification_for_a_Paging_Server|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)</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Overhaul of the ACD server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improvements to the Auto-Attendant</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improvements to Hunt Groups</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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 [http://track.sipfoundry.org/browse/XCF-831 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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Overhaul of the security and authentication system</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved E911 call routing</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Simplified dial plan configuration</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Registered phones displayed per user</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New device category: SBC</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Automated restore from backup</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The current restore from backup functionality will be integrated into Config Server.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Server and application statistics, reports, and alarms</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Support for new Polycom 320 / 330 phones / Polycom 2.2.2 firmware</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Support for Linksys Phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We are adding support for Linksys SPA941 and SPA942 phones fully integrated into the sipXconfig management system thanks to a community contribution.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Support for IpDialog SipTone V Phone</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We are adding support for the IpDialog SipTone V phone fully integrated into the sipXconfig management system thanks to a community contribution.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Support for LG-Nortel 1535 Video Phone</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Report: Login history</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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).</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Symmetric signaling / merged proxy</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">SIP loop detection</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Port to PowerPC (PPC)</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXecs was ported to the PowerPC (PPC) platform with all the big endian handling for audio processing and other issues.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Port to FreeBSD</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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 [http://track.sipfoundry.org/browse/XECS-108 XECS-108] for the port files and [[FreeBSD port for sipXecs 3.10]] for documentation..</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New XML RPC process management API</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">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.</td>

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</center>

== Stable Release 3.8 ==

Release 3.8 is focused on improving SIP Trunking capabilities as well as support for directory, speed dial and BLF on the phone.

'''Release Notes:''' [[Detailed Release Notes for sipXecs Release 3.8]]

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<th><big>Release 3.7 / 3.8</big></th>

<th>Comments</th>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New ACD Call Center Server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The ACD Call Center server has been a closed source software up to release 3.6 and was made available into open source in the course of the 3.8 development cycle. This call center ACD server serves up to 50 agents with several queues. It is typically used as an informal call center for IT helpdesks and other applications that require management of calls in a queue.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management support for Audiocodes Gateways</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This is a major milestone for sipX as we finally add full plug & play management support for all Audiocodes gateways. This means that gateways are managed in a very similar way as compared to phones. All configuratoin is generated by sipXconfig, where sipXconfig chooses default parameters where possible to render a working config out of the box. The gateway then picks up these generated profiles from the sipX server. We plan to support all Audiocodes gateways with initial focus on the following models: MP-114 FXS and FXO, MP-118 FXO and FXS, MP-124 FXS, Mediant 1000. TP-260 and Mediant 2000 are priority two.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & play management support for LG-Nortel phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.8 adds plug & play management support for LG-Nortel phones 6804, 6812, and 6830. These phones support standards based Music on Hold (MoH).</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">New Voicemail Portal</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The voicemail portal used by users to retrieve and manage voicemail messages from a Web browser has always been a separate application that required a separate login. We are now integrating the voicemail portal into the user poral of Config Server. Going forward only one user login will be required and the user will be able to manage all user configurable aspects of the phone system including voicemail from there. That includes configuration of forwarding rules and speed dial entries. This represents the first step towards separating the Media Server from the rest of the system. Once done, the system will support several Media Servers on separate HW and all centrally managed by Config Server.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Automated Configuration of HA Slave Systems</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This relates to a further simplification of the installation process. Certificates can now be distributed to the Slave server in an HA configuration automatically during the installation process. Config Server manages the Slave system remotely with the ability to enable and disable services on the remote Slave host.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Phone Directory Support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Depending on the phone model it is possible to load directory information into the phone. Release 3.8 will provide a capability to generate a corporate directory based on the user database in sipX augmented by a file import capability using .csv files. This information will be compiled into a directory that can be loaded by the phone. Inclusion into the directory is controlled by group membership as well as a specific permission flags that allows for inclusion.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Extended support for Localization</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig can be skinned and localized so that the presented language dependes on the users browser settings. sipXconfig is being extended to allow for full localization to be done in .properties files. In addition, the Polycom phone model is extended to support phone localization.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Speed Dial Support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In addition to directory information we plan on supporting the user specific configuration of speed dial keys (soft key assignments on the phone). The user will be able to add individual speed dial assignments using the user portal of sipXconfig.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Busy Lamp Field (BLF)</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We talked about BLF several times and we remain serious about it. Polycom has changed their BLF implementation several times now across different versions of firmware, which made it difficult to follow a straight course. We now decided to implement a sipX presence server based on dialog events. This presence server will collect status information from phones that offer it and allow subscription to such information. A centralized solution is harder to implement, but it is more economical in terms of network bandwidth requirements and it will form the basis for more extensive implementations of presence based services such as interconnection to IM systems such as Jabber and Microsoft LCS.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Domain routing with wildcards</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">As a last minute item we are adding improved SIP domain routing capabilities to release 3.8. This will allow domain based routing (including wildcards to define domain names). Calls to different domains (i.e different SIP trunking providers) can be routed along different routes.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">ISN (ITAD) Signalling</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">[http://freenum.org/ ISN signalling] is a new way of bypassing the PSTN. ISN provides an easy way for campuses, enterprises, and ASPs to acquire globally-unique subscriber numbers to support new communications services. ISNs are free and they provide a domain-based, "Internet-style" number that looks more like an email address than a traditional E.164 telephone number. An ISN is formed by joining a domain-local subscriber number to an ITAD (Internet Telephony Administrative Domain) number, using an asterisk as the delimiter. For example, subscriber 1234 in ITAD 256 would have ISN: 1234*256.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">ENUM Signalling</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">ENUM will be supported as an additional redirector plugin, configurable using sipXconfig. ENUM allows the automatic routing of calls over an IP netwrok provided that for the dialed PSTN number there is an IP address equivalent defined in an ENUM registry database. Several ENUM registries can be queried.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Redirector Plugins</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Redirector plugins provide a simple mechanism to add redirectors at start time using a simple API. A redirector implements a specific routing rule that is considered as sipX evaluates the dial plan everytime a session is initiated. ISN signalling is implemented as a redirector. ENUM is another redirector. More common dialing rules are now also implemented as redirectors so that with release 3.8 we will have about 15 redirectors in the system already. More exotic redirectors can be added easily. For example: A redirctor could use a database to map every dialed number or URI to a specific other number or URI.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">CDR Reports</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Since release 3.4 sipX supports CDR data collection for both non-redundant and HA systems. We plan on improving CDR reporting by adding a report generation mechanism that extracts the data from the database and presents it in a user friendly way. The entire CDR post-processing part is re-written to enable real-time reporting of calls. A screen inside sipXconfig will display calls as the terminate, automatically refreshing the windows in a given interval. CDR reports can then be exported to a spreadsheet.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Real-time view of ongoing calls</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In addition to near real-time reporting of CDRs for completed calls, it will be possible to see what calls are currently in process using sipXconfig. The CallResolver process is extended with a SOAP interface that allows querying currently active calls. This information will be displayed by sipXconfig.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dialplan Localization</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipXconfig will support the automatic switching and re-initialization of a localized dialplan. sipx easily supports different dialplans that can be localized both with respect to a country's or regions dialplan requirements as well as language.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Support for Grandstream GXV-3000 Video Phone</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We now support plug & play configuration management of the Grandstream GXV-3000 video phone, which means that we now have complete suppport for all the Grandstream phones and TAs. Thanks to IIPS for their help. Grandstream still does not support dialog events in their phones, so that certain features such as call park and call pickup do not work.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated SNOM Configuration Support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Support for plug & play management for SNOM phones got updated. In addition to existing capabilities the SNOM phone model now supports speed dial and directory capabilities.</td>

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</center>

== Timeframe for Release 3.7 / 3.8 ==

Development release 3.7 will become stable release 3.8. We decided to extend the development phase for release 3.8 until end of February. After the most extensive test and bug fix phase in the history of the project, we are expecting release 3.8 to become stable in June.

== Stable Release 3.6 ==

On October 3 the first beta release of 3.6 was made available on SIPfoundry. Release 3.6 is focused on improving the flexibility of the sipX dial plan based on experience gained through many production deployments. In addition, 3.6 will add additional critical features as outlined below:

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<th><big>Release 3.6</big></th>

<th>Comments</th>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">LDAP Support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">LDAP support has made it into the 3.6 release. We allow backend synchronization with an LDAP capable directory and upload the relevant information into Config Server. For performance reasons session authentication will still be done internal to sipX. The implementation should be compatible also with Microsoft AD but it has not yet been tested.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Intercom / Paging</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.6 adds an Intercom capability that in a first phase will support point-to-point intercom using Polycom phone's auto-answer capability.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Dial Plan Templating</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In support of international deployments it will be possible to create country or region specific dial plans that can be selected within config server. As an example and in addition to the U.S. dial plan there is now a Swiss dial plan as well as a Polish dial plan. Additional dial plans are easily defined in XML and can be added to sipX as part of a language pack.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Domain Alias</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.6 will allow domain aliasing, which improves deployability of sipX in environments based on Microsoft Windows Server, as well as in cases where sipX needs to be responsible for calls from different domains or IP addresses.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Park Server Enhancements</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipX Park Server now includes several additional configurable features. There is a time-out value configurable so that after a defined period of time the parked called is transferred back to the person that parked the call. Also, there is a configurable escape key from park. When pressed the call is transferred back to the person that parked the call. Also, it is now possible to configure whether the system allows several calls to be parked on one park orbit (FILA) or not.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Gateway Configuration improvements</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The PSTN gateway configuration now allows adding a gateway specific prefix before the number is dialed. This allows the accommodation of different number conversion requirements in case several gateways are associated with a single dial plan rule.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''yum'' based Install & Updates</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We plan to eliminate the single file install script and allow installs simply using ''yum''. This is already possible for the Debian build since release 3.0.1.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Support for new Polycom phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release adds support for the new Polycom phones such as the SoundPoint IP430. We also updated the plug & play functionality to including the Polycom 2.0 firmware. The sipX Config Server now supports mixed deployment with Polycom phones on the 1.x firmware release and phones already on the new 2.0 firmware release.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''FTP Server support for phone management</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In addition to the TFTP server sipX now comes standard with a configured FTP server. The FTP server provides access to the same configuration directory used by the TFTP server for phones capable of using FTP instead of TFTP. This is especially convenient for Polycom phones that come factory configured to use FTP.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Support for Hitachi Cable WiFi Phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release will support the Hitachi Cable IP 5000 and IP 3000 WiFi phone in Config Server.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''SIP Trunk support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We added an option to add a SIP destination as a trunking gateway. This SIP trunking gateway is selectable from the dial plan in the same way a PSTN gateway is selected. A route header field allows the definition of a Session Border Controller (SBC) used to route the call across NAT / Firewall.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Custom Permissions</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">sipX already offered a flexible mechanism to use different permissions in dial plan rules. We now added the ability to define additional custom permissions that are administrered by the admin and used in the same way built in permissions are.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Caller ID manipulation (CLID / CLIR)</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We greatly extended the ability to manipulate and define Caller ID for outbound calls. This allows a much more flexible mapping of User ID to CLID on a per user, per user group as well as on a per gateway basis. In addition we added Caller ID Restriction (CLIR) on a per user, per user group and per gateway basis. The User definition now includes, in addition to User ID and Aliases, a line to define outgoing caller ID. This makes it possible to have e.g. an alpha-numeric or 4 digit local extension as your User ID while still send the full DID number of the users as caller ID to the PSTN.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Updated SNOM phone support</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">This release updated support for the Snom phones to firmware release 6.2.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Music On Hold (MOH) for Snom phones</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Snom phones are the first to support the new IETF standard for Music On Hold (MoH). Release 3.6 provides an IETF standard compliant music source to which the Snom phones can transfer a call when hold is pressed. This provides for a scalable implementation of MoH. A music file can be uploaded from Config Server.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">''Performance improvement of the media server</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">We were able to work on some performance improvements for the sipX media server (voice mail subsystem). With that improvement we should now be able to support more virtual media server ports.</td>

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'''Note:''' We missed on the BLF feature in 3.6. We will try and make good on that in release 3.8.

== Timeframe for Release 3.6 ==

Release 3.6 BETA was made available October 3rd. We expect a stable version by mid November.

== Stable Release (3.4) ==

The objective with release 3.4 is to add CDR recording and make it available in a stable release as quickly as possible. CDR recording is a key feature for most of the larger deployments of sipX.

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<th><big>Release 3.4</big></th>

<th>Comments</th>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">CDR Reports</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Call Detail Records (CDRs) will be [http://scm.sipfoundry.org/viewsvn/checkout/sipX/branches/3.4/sipXproxy/doc/cdr/cse_database_design.txt stored in a PostgreSQL database]. This database will reside either on the sipX host or a different dedicated host. The design is optimized for performance. An additional CDR report generator (future) will allow the creation of custom reports. Third party report generatrors may be used to customize views based on raw database information.

The CDR Database will also support the new HA configuration with redundant proxy and registrar servers. Call State Events will be collected from all the proxy and registrar servers and reporting will take into account cases where a call starts on one proxy but terminates on another because of a failover condition.

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Auto-Attendant Improvements</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Additional configuration parameters were added to the Auto-Attendant: Inter-digit timeout, overall DTMF timeout value, and maximum length of DTMFs.</td>

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== Timeframe for Release 3.4 ==

Release 3.4 was released stable in July 2006.

== Stable Release (3.2) ==

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<th><big>Release 3.2</big></th>

<th>Comments</th>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">High Availability</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In sipXpbx, basic calling depends on three components: the two proxies and the registrar/redirect service. The proxies can be replicated and DNS SRV records can be used to share load and provide for failover. The registrar/redirect service, however, cannot currently be deployed on multiple servers because the 'soft' state in the registry database (mappings from registered Addresses to Contacts) cannot be shared. While replicating the proxies alone does help with scaling, the registrar is a single point of failure for basic calling service. In release 3.2 we are planning to introduce high-availability for basic calling features by adding the required replication of registration information.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Updated SOAP Interface</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The SOAP interface first became available in release 2.8 and was then dropped in release 3.0 because of the re-design of the Config Server. Release 3.2 will re-introduce the SOAP interface. For [http://sipx-wiki.calivia.com/index.php/SipX_ConfigServer_SOAP_API more details on the API].</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Simpler Upgrades</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The sipX team set out to develop a solution to simplify upgrades. Data migration for configuration data is done automatically as part of the release upgrade process. With 3.2 therefore, the user will be able to install 3.2 over 3.0.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Plug & Play Management Improvements</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Already release 2.8 introduced plug & play management of phones. Release 3.0 enhanced that capability significantly providing support for many additional phones, including the first gateway, as

well as provide a simple XML-based framework to add support for new phones and gateways. Release 3.2 will add the capability to manage firmware upgrades using the sipX Configuration Server. 3.2 will also have an update to the Polycom phone templates. All testing will be conducted with 3.1 boot rom and 1.6.4 SIP firmware.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved Logging</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Release 3.2 will introduce snapshot logs that can be easily created and downloaded using the Web interface. Eventually we would like to introduce much more comprehensive server and application performance management.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Time of Day/Day of Week Auto Attendant Routing</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">The 3.2 Auto Attendant has been enhanced to support a time-based routing feature. Auto Attendants can be configured to answer based on working hours, closed hours or holidays. A system overide is also available that can be envoked from a remote telephone in cases where the office may be closed unexpectedy -- snow days etc.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">CSV Import Tool ("Cut Sheet")</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Administrators will be able to bulk load users, user groups, user aliases, phones, phone MAC addresses, phone types, phone groups and group settings, </td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved Navigation</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">In 3.2 the navigation structure is now configurable with a CSS. A new search feature has also been added that indexes users, phones and all associated settings.</td>

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<td width="18%" valign="top" bgcolor="#f2f3f3" style="border:1px solid #c2c2c2;">Improved Performance</td>

<td valign="top" bgcolor="#ebf5fc" style="border:1px solid #c2c2c2;" align="left">Registration as well as subscribe / notify performance has been significantly imporved. In installations with large numbers of phones, transactions generated by re-registration and subscribe / notify can far exceed the actuall call rate. sipX release 3.2 is tested up to 5,000 users for adequate re-registration and subscribe / notify performance.</td>

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== Timeframe for Release 3.2 ==

Release 3.2 was released stable in April 2006. Release notes give you information about new features added in a release. Before upgrading to a new release you should consult the release notes. More detailed release notes can be extracted from the SIPfoundry tracker.

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