New ITS821 firmware available at http://www.telesis-pbx.com/telesis.php?ID=05-04

Version 1.13:

Major modifications.

DHCP (Dynamic Host Configuration Protocol) is implemented. Consequently, the ITS821 can be added to the network with less manual intervention.

IMPORTANT: Web server is no longer available for IP and account settings of the ITS821. Web connection is used only for upgrading the firmware.

IMPORTANT: IP and account settings of the ITS821 can be managed with using the keys on the set instead of web server connection.

To access the settings menu:

• disconnect the mains power adapter from the ITS821,
• press `ESC` and `DEL` keys on the set at the same time,
• keep `ESC` and `DEL` keys pressed and connect again the mains power adapter to the ITS821.

The ITS821 starts with the settings menu.

The settings menu has three submenus:

• `IP settings`: Enable or disable `DHCP`.
• `Xymphony account settings`: Edit or modify xsip registrar IP address (i.e., IP  address of the Telesis system where the ITS821 registers), account name, authentication  ID, and password. Numeric characters are edited with using the numeric keypad.   Alphabetic characters are edited with using the alpabetic keyboard. `Shift` key  may used for switching to capital letters. Use the navigation keys for moving the  cursor, erasing characters etc.
• `Restart`: Restarts the ITS821 with the new settings.

New routing criteria rule: `Personal greeting`

The new rule allows users to record their voices and messages as their greeting messages in Voice Mail applications. To record a personal greeting message `Personal greeting` rule and then `1` is dialed. To erease a personal greeting message `Personal greeting` rule and then `0` is dialed. When there is no personal greeting message in a user, the system plays the common, i.e. system-based, greeting message (message-9).

During listening `Voice Mail Send` message, it is possible to dial `1` to leave a voice message. As a new implementation, it is also possible to dial `0` to get the dial tone and make another call without leaving a voice message.
Several new features added to create advanced Auto Attendant (Automatic Call Distribution) scenarios.

New item in the integrated web server: `AutoAttendant` When it is clicked on `Auto Attendant list` is accessed. The list has 31 `Attendant Id`s, which are numebered from 0 to 30.
New routing criteria rule: `Auto attendant`. The new rule allows to access an `Attendant Id`. Next two digits (from 00 to 30), which follow the rule, activate the relevant `Attendant Id` for operation.
New selection box in `Common properties` of users: `Auto attendant on setup` In this selection box, numbers from 0 to 30 can be selected. This makes an `Attendant Id` to be statically assigned for the user in incoming calls. This selection box may be utilized especially in trunks, which are intended for automatic call distribution.
Each `Attendant Id` has the following fields:

• `Main msg.`    : System message number to be played if `Message set` is selected as `Main` in the analysis table in the same page.
• `Alt msg.`     : System message number to be played if `Message set` is selected as `Alt` in the analysis table in the same page.
•  `Timer expired`: If no digit is received while playing the message, call attempt is forwarded to the digits in this textbox. A new routing analysis applies for this digit sequence. The result could be a new route, port, system message etc.
• analysis table : This table has 11 rows for the prefixes 0 to 9, * and #.  Each row has `Message set`, `Law`, and `Replacement` fields. In created scenerio, dialing prefixes 0 to 9, * and # results in activation of a new `Attendant Id` or start of the routing analysis. In case of a new `Attendant Id`, it is possible to select which message (main or alternate) of this new `Attendant Id` to be played back with configuring `Message set` switch. In case of routing analysis, it is possible to start the analysis with replaced digits set in `Replacement` field.

The operation in brief: The initial state of the `Message set` is ‘Main’. When an `Attendant Id` is attached to a call, appropriate system message is played. Received digits from the user are not used for routing analysis but forwarded to the attendant. Upon reception of a digit, played system message stopped. Analysis table row selected such that the received digit matches to the ‘Prefix’ field of the row. Then the row parameters are applied to the call. 

 w.pxf.lscxsip  - Licence for a single VoIP phone user with xSIP protocol (that is ITS821 or XPhone)

Maximum Number of xSIP User Licences

PX24U: 32

PX24M: 32

PX24S: 32

PX24X: 96

X1r4: 96

PX24r4: 1

Notes about xSIP licencing:

xSIP stands for eXtended SIP protocol. It is developed by Telesis to make some value-added services in Telesis systems to be applicable for VoIP calls too. Each Telesis system supports a single xSIP user without requiring any licences. Furthermore, XPhone is a freeware softphone. Consequently, you can easily test and see the capabilities of the xSIP without any cost. However, if you need more ITS821 (on payment) or XPhone (freeware) users then you will need to purchase necessary number of xSIP user licences. xSIP users do not need VoIP-TDM gateway channel licences to operate. Only xSIP user licences are sufficient.

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 VOIP WITH SIP OR H.323 PROTOCOL

 w.pxf.voip002 - Licence for 2 (TWO) VoIP channels for IP to TDM and TDM to IP routing.

Maximum Number of VoIP (H.323 or SIP) to TDM Gateway Channels:

PX24U: 8

PX24M: 32

PX24S: 32

PX24X: 32

X1r4: 32

Notes about VoIP-TDM (or TDM-VoIP) licencing:

Telesis systems are with IP-TDM Gateway and media proxying capabilities. That means calls from the TDM network (circuit switched network) may be routed to the IP network and vice versa. Also, whenever it is necessary the integrated media proxy operates for calls between VoIP entities (like in H.323 and SIP integration). VoIP channel licence in Telesis systems refers to the allowed simultaneous calls from TDM to IP (SIP or H.323) and IP (SIP or H.323) to TDM network, as well as half of the simultaneous calls made by using the integrated media proxy. To illustrate, in order to have a capacity of 8 simultaneous calls from the IP network to the TDM one, a Telesis system must have been licenced with minimum 8 VoIP-TDM gateway channels (4 licences must be purchased). To remind you, whenever a licenced channel is free (not busy), it can be used by any TDM or VoIP user. Note that, VoIP calls here refer to SIP or H.323 VoIP calls only. VoIP calls with using Telesis xSIP protocol are excluded.

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 IMPORTANT NOTE:

For your large capacity xSIP, SIP, or H.323 projects, we recommed you to contact us and get confirmation about your configuration. This is beceause some complex calculations may be neccessary for resouce management.

A. There is no restriction on number of voicemail messages for X1 exchange. If you give 5 minutes voice message quote for 500 subscriber, it means 42 hours, DVR has 100 hours storage capacity, so it is not problem. You can give voicemail quote for several subscribers whatever you want.

A1. First of all record message (custom music) for system message 04, then goto common properties of DC loop trunk and mark “voice message on alerting” check box

Q2. They need to change music on hold with their own custom music, is it possible and what is maximum duration?

 A2. You need to change “hold” system message (messge number 08) with your custome one. Max duration of system messages is 10 minutes. The format for these sound files is the Windows PCM format: Wave file, with a `.wav` extension. Files compatible with the exchange should be 16 bits and have an 8 kHz sampling rate. Only one channel (mono) wave files are supported. You can use for example GoldWave Editor software to save your message in proper format.

Auto Voice recording licence is independent from DVR licences. So if you want for instance 16 port auto voice recording channel licence you don’t need additional DVR licence. You can install 16 auto voice recording channel licence for PX24X and PX24M exchanges.

If you need more than 16 port  auto voice recording channel licence please contact TELESIS.

In order the voice recorrding option to appear for E1 DSS1, you need minimum 30 recording channel licences to be installed in the system!

So for example, if we have 4xE1 control & switching unit, 160 analog subscribers and 80 DC loop trunks:

• Control & switching board: 0.450 mA
• DASF16 (Analog subscriber board 16 ch.) 10 x 0.065A = 0.65A
• DATF08 (Analog DC loop trunk board 8 ch.) 10 x 0.040A = 0.40A

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          Totally 1.50 A

 If you assume 20% traffic, 32 subscriebs are busy so 32×0.030A=0.960A

So totally 2.46A

Power Supply Capacity:

• 1.1 Amps at 54V
• 1.25 Amps at 48V

First of all you have to use Multistation call group,

Then fill the Multistation table with analog, digital, sip or H323 access code, You can create or edit “Multi station call” groups. Up to 8 groups can be defined.Use “Next” button to surf among the groups. Each group can have up to 8 user access codes. Edit access codes in textboxes to add these into the multi station call group.

Finally When multi station group call feature is operated, all available users in the relevant group ring when a call comes in. When one of the group member answers, he/she joins the conference. For example if you dial 8140, all available users of table 0 will ring or, if you dial 8141 all available users of table 1 will ring