New ACD Call Center Server

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.

Plug & play management support for Audiocodes Gateways

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.

Plug & play management support for LG-Nortel phones

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).

New Voicemail Portal

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.

Automated Configuration of HA Slave Systems

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.

Phone Directory Support

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.

Extended support for Localization

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.

Speed Dial Support

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.

Busy Lamp Field (BLF)

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.

Domain routing with wildcards

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.

ISN (ITAD) Signalling

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.

ENUM Signalling

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.

Redirector Plugins

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.

CDR Reports

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.

Real-time view of ongoing calls

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.

Dialplan Localization

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.

Support for Grandstream GXV-3000 Video Phone

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.

Updated SNOM Configuration Support

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.

Detailed List

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Improvement

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