VoiSmart Phones

http://www.voismart.it/en/index.php?option=com_content&task=view&id=43&Itemid=146

Brand new cost-effective SIP phone.

New Firmware 2.2.3 ( 01 March 2006 )

Supports SIP Presence (line monitoring), up to 20 lines
All 20 DSS keys can now be programmed
Supports Distinctive Ringing (alert-info header can be configured via web interface)
Supports Autoanswer (4 ways: normal, urgent, imperious and silent, SIP driven)
Supports Autoanswer on each account
Nat keepalive can be set on each account (empty udp packets to keep nat pinholes open)
Supports Instant Messaging & Flash messages
INFO DTMF relay now supported
Locally generated tones can be configured
AES encryption of firmware and configuration
Configuration can be exported, AES encrypted or not
Remote syslog support
Alarm-on-the-phone

Features:

Caller ID
Voicemail indicator and retrieval via keypad (with our without MWI subscription)
Speakerphone and mute indicator
Ethernet bridge 10/100
Hand-free full-duplex speakerphone
Menu layout for easy navigation
Track missed/outgoing/incoming calls
Autodial support
Call hold
Call transfer, consultative and blind
Call forwarding (No answer, busy, DND)
3-party conferencing
Call log
Redial
Call waiting
Graphical web configuration
Graphical telnet configuration
Autoprovisioning via tftp / http
Autoprovision can be AES encrypted

Protocol:

The IP Phone supports the following protocols at least.
SIP call signaling — RFC 2543 (over UDP), RFC3261
RTP, RTCP
IP configuration via DHCP, PPPoE, Static
SNMP (remote monitoring)
STUN (nat discovery)
SNTP (network time adjustment)

Interface:

Dual 10/100 Mbps Ethernet switch RJ-45 ports. One connects to IP network, another to IP Client.
VLAN supported
DSS Key: 2-Color-LED x 20 Keys.
LCD display: 2*16 Char. (Advanced 320*240 Graphic)
Handset and speaker: Support Speakerphone.
Volume adjustment: Increase and decrease the voice of handset, speaker, and ringer.
Message LED : 2-Color-LED Message indication.
Voice processing:
Codec: G.711 (a-law and u-law), G.723.1 and G.729AB.
Acoustic Echo Cancellation
VAD (Voice Activity Detection)
CNG (Comfort Noise Generation)

DTMF Relay:

Support in-band (RTP) , out-band DTMF, or INFO transport over IP network.

Application Info

Some examples on how to use SIP driven autoanswer & instant messaging.

Autoanswer:

Autoanswer can work in 5 modes:
  • configured on each account: this way the phone will always autoanswer on incoming calls on this account
  • Driven with a SIP proprietary header: just add the SIP header P-Auto-Answer with one of these values:
    • normal : the phone will autoanswer the call, if the user is not busy on another call.
    • urgent : the phone will autoanswer the call, if the user is already on call, the ongoing call will be put on hold and the new will be autoanswered
    • imperious: like urgent, but bypasses also DND and forward
    • silent: like imperous, but doesn't play "bip" on autoanswering
On asterisk, you can set the header with SIPAddHeader before the Dial command.

Instant Messaging:

You can send IM with the standard SIP message event (SendText app in asterisk)
or with an external UA. If the SIP MESSAGE event contains the SIP P-header
P-Flash-SMS: no (yes 'no' is right, was meant as 'on' but there's a typo in the firmware),
the message will displayed on the LCD and will disappear after 3 secs, without being stored.
Stored messages can be retrieved with the proper function key, or with the web interface.

http://www.voismart.it/en/index.php?option=com_content&task=view&id=43&Itemid=146

Brand new cost-effective SIP phone.

New Firmware 2.2.3 ( 01 March 2006 )

Supports SIP Presence (line monitoring), up to 20 lines
All 20 DSS keys can now be programmed
Supports Distinctive Ringing (alert-info header can be configured via web interface)
Supports Autoanswer (4 ways: normal, urgent, imperious and silent, SIP driven)
Supports Autoanswer on each account
Nat keepalive can be set on each account (empty udp packets to keep nat pinholes open)
Supports Instant Messaging & Flash messages
INFO DTMF relay now supported
Locally generated tones can be configured
AES encryption of firmware and configuration
Configuration can be exported, AES encrypted or not
Remote syslog support
Alarm-on-the-phone

Features:

Caller ID
Voicemail indicator and retrieval via keypad (with our without MWI subscription)
Speakerphone and mute indicator
Ethernet bridge 10/100
Hand-free full-duplex speakerphone
Menu layout for easy navigation
Track missed/outgoing/incoming calls
Autodial support
Call hold
Call transfer, consultative and blind
Call forwarding (No answer, busy, DND)
3-party conferencing
Call log
Redial
Call waiting
Graphical web configuration
Graphical telnet configuration
Autoprovisioning via tftp / http
Autoprovision can be AES encrypted

Protocol:

The IP Phone supports the following protocols at least.
SIP call signaling — RFC 2543 (over UDP), RFC3261
RTP, RTCP
IP configuration via DHCP, PPPoE, Static
SNMP (remote monitoring)
STUN (nat discovery)
SNTP (network time adjustment)

Interface:

Dual 10/100 Mbps Ethernet switch RJ-45 ports. One connects to IP network, another to IP Client.
VLAN supported
DSS Key: 2-Color-LED x 20 Keys.
LCD display: 2*16 Char. (Advanced 320*240 Graphic)
Handset and speaker: Support Speakerphone.
Volume adjustment: Increase and decrease the voice of handset, speaker, and ringer.
Message LED : 2-Color-LED Message indication.
Voice processing:
Codec: G.711 (a-law and u-law), G.723.1 and G.729AB.
Acoustic Echo Cancellation
VAD (Voice Activity Detection)
CNG (Comfort Noise Generation)

DTMF Relay:

Support in-band (RTP) , out-band DTMF, or INFO transport over IP network.

Application Info

Some examples on how to use SIP driven autoanswer & instant messaging.

Autoanswer:

Autoanswer can work in 5 modes:
  • configured on each account: this way the phone will always autoanswer on incoming calls on this account
  • Driven with a SIP proprietary header: just add the SIP header P-Auto-Answer with one of these values:
    • normal : the phone will autoanswer the call, if the user is not busy on another call.
    • urgent : the phone will autoanswer the call, if the user is already on call, the ongoing call will be put on hold and the new will be autoanswered
    • imperious: like urgent, but bypasses also DND and forward
    • silent: like imperous, but doesn't play "bip" on autoanswering
On asterisk, you can set the header with SIPAddHeader before the Dial command.

Instant Messaging:

You can send IM with the standard SIP message event (SendText app in asterisk)
or with an external UA. If the SIP MESSAGE event contains the SIP P-header
P-Flash-SMS: no (yes 'no' is right, was meant as 'on' but there's a typo in the firmware),
the message will displayed on the LCD and will disappear after 3 secs, without being stored.
Stored messages can be retrieved with the proper function key, or with the web interface.

Created by: mbranca, Last modification: Thu 31 of Jan, 2008 (15:43 UTC) by linkx
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