Snom320/Firmware/Release Notes

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7.3 branch



7.1 branch

6.5 branch

6.5.18 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
Phone User Interface
  • handle slow detection of preset extension pads
  • fixed speed dial key 30
  • for speed dial dialed 09 is not the same as 9
  • remove ringing calls from list of possible transferees
  • use soft transfer key only for deflection of ringing call
  • ringing calls were not shown in hold state (320 and 370)
  • normalized placement of soft keys across the various call states
  • added 128k memory bin for efficient memory allocation
  • changed "Deny All" soft key caption to "Block"
Localization
  • phone switches to/from DST in time now
  • Added DST for Perth/Australia
  • DST changes for Australia
  • DST changes for New Zealand
  • added timezones for Venezuela and New Caledonia
Miscellaneous
  • Fix regarding skipping of the initial audio packets in some asterisk environments
  • reporting detection of expansion Modules right after lcs starts. Cannot get any faster. This should fix exp-keypad recognition and assoziated subscribtion sending once and for all. (FINALLY!)
  • timer range enhanced
  • Spanning Tree Protocol (STP) packets are now forwarded from the LAN to the PC ethernet port (Managable switches supporting STP can now detect network loops.)
  • write the DHCP hostname only if there is enough space
  • gracefully restart if DHCP server isn't reachable anymore
  • disable flash plugin by default
  • close flash plugin port dependent on setting 'with_flash'
Download Link

6.5.17 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
WEB
  • values of disable_blind_transfer and disable_deflection were not properly reflected at the advanced web interface page
LID
  • fixed bug where subscribtions to extensions registered to exp-keypad-keys weren't send out
Download Link

6.5.16 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
SIP
  • use request URI when selecting line key

6.5.15 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
GUI
  • port 5061 was being dropped from URI
  • improved transfer handling
SIP
  • call completion subscriptions should not be delayed
  • fixed potential crash when deregistering mailbox
Download Link

6.5.14 beta

WEBCLIENT
  • sometimes building of proper destination URL failed, so provisioning wasn't working
LID
  • making sure sending packets contain negotiated codec data only (to avoid a rare diff. codec packet)
SIP
  • fixed missing ack of bye after transfer

6.5.13 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
WEBCLIENT
  • fixed audio device left active after MWI
  • star code exception handling e.g. *1*2*3*4 not to be confused as an ip address
  • show version info on boot up
SIP
  • fixed incorrect processsing of Replaces header
  • fixed crash during reboot
WEBCLIENT
  • fixed support for Transfer-Encoding chunked
LID
  • enhancement for update to v7 (new bootloader header for smoother v6 to v7 transition)
  • fixed the rare continuous dtmf problem in inband dtmf
Download Link

6.5.12 beta

GUI
  • calls on hold are shown at a higher priority than incoming calls
  • programmable key TRANSFER could not be remapped
SIP
  • added call pickup via access code on programmable function keys e.g. ext@dmn.org|*68
  • multiple REGISTERs over TCP were not all authenticated
  • fixed problem with lost subscriptions
LID
  • fixed initial skipping of audio packets problem (in asterisk)

6.5.11 beta

GUI
  • show simple msg if Voice-Message header is not set in NOTIFY
  • handle MWI with same mw account and domain for different identities
LID
  • added support for new mac range 29
  • enhancement for update to v7
SIP
  • fixed bug where CANCEL was suppressed in some transfer scenarios

6.5.10 beta

DST
  • corrections for Spain and Portugal
LID
  • added ability to update to v7

6.5.9 beta

GUI
  • fixed local line context mapping (broken registrar might be needed on)
  • speaker to toggle headset on snom300
  • fixed updated MWI on message txt reading
  • added Automatic Redial feature
  • new feature to disable blind Xfer (REFER)
  • new feature to disable deflection (code 302)
SRTP
  • fixed ROC issue
LID
  • improved random number generation
TLS
  • added certificate parsing patch
MINIBROWSER
  • hook off in SnomIPPhoneDirectory can dial number
  • fixing wrong error message after dialing a number in SnomIPPhoneDirectory
DNS
  • fixed memory leak of webclient request in case of non resolvable DNS URL

6.5.8 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
DST
  • Changed daylight savings time for Canada beginning with 2007.
  • Timezone Hawaii has no DST.
  • Added timezone USA -7 MST no DST.
SIP
  • Fixed illegal connection erase with 2 calls on hold.
WEB
  • Added 19222 to the emergency number list.
Download Link

6.5.6 beta

DST
  • Changed daylight savings time for USA beginning with 2007.
  • Fixed wrong DST calculation. Was starting at the wrong time (important especially for countries changing last sunday in march).
SIP
  • new setting user_full_sdp_answer sends codec interception in 200OK

6.5.5 beta

GUI
  • fixed voice recording in idle state
  • added wireless headset active indication (snom360)
SIP
  • remove NOTIFY related packets from lists after processing
  • stop packet retransmission for the packets associated with a finished call
  • answer with full SDP if user_full_sdp_answer is set (on by default)
LID
  • wireless headset support added (snom360)

6.5.4 beta

GUI
  • fixed duration on caller id
  • fixed auto leave edit number on offhook
  • fixed speaker key on handsfree anomaly

6.5.3 beta

GUI
  • don't join 2 calls inadvertently due to auto dial
SIP
  • fixed codec switch problem when 2nd party leaves conference
TLS
  • sanity checks added for rsa keys
WEB
  • use stk_with_flash setting for setting up the XML port

6.5.2 release

GUI
  • fixed blind Xfer through extension fkey with 2 calls
  • fixed challenge response on register (snom320 only)
  • fixed Busy Lamp for fkeys 5&6 (snom300 only)
WEB
  • do not clear the given number for fkeys even if context is not registered
Download Link

6.5.1 beta

SIP
  • fixed subscription refresh on reboot (if subscription_delay not set)
  • new setting terminate_subscribers_on_reboot added to switch off unSUBSCRIBEs on reboot

6.5 beta

GUI
  • LDAP was not working through programmable keys
  • ignore star codes dialing for redirection on reboot
  • support Xfer on hold
  • fixed Net Info stats problem caused by to much traffic
  • picked up calls are assigned to free line keys
  • call park holds the active call
  • call park/transfer failure resumes the last active call
  • Dialog info may be made visible through xml idle screen
  • option to set audible pickup indication
  • don't allow soft keys on locked keyboard
  • show only available fkeys at gui fkey selection
  • fixed speaker key on headset anamoly
  • improved auto connect indication
WEB
  • changed manual link to snom wiki
  • save complete url as argument for programmable keys
  • added local alert info (internal/external/group)
  • fixed scheme through dialplan on web interface dialing
  • setting to control active line scrolling
  • backlight can be set to always on
  • added output from /proc/net/tcp to memstat.htm
  • removed unnecassary linefeeds at memstat.htm
  • removed deprecated options „never_update_firm" and „never_update_boot" for setting „update_policy"
  • readded voice recorder functionality
SIP
  • fixed subscriptions on Line Active/Inactive and Re-register
  • restore existing call on failed blind Xfer
  • fixed RTCP stats in 200 OK of BYE
  • allow ringback status if REFER received during a blind Xfer
  • added new setting subscription_delay to delay subscriptions randomly in a given time range
  • added new setting subscription_expiry to change the expiry time of subscriptions
PROVISIONING
  • fixed HTTP request storm
DIALPLAN
  • dp usage was heavily slowing down the phone, fixed
DHCP
  • initially go on with DHCP DISCOVER for ever or press cancel
MINIBROWSER
  • Action URL Settings can drive minibrowser
  • SNMP
  • registration status is working now

6.3 beta

GUI
  • added LDAP browsing on down key
  • dont change audio device on speaker key volume
  • find latest channel for action url connection variables
  • timezone Hawaii has no DST, fixed
  • added timezone USA -7 MST no DST
LID
  • fixed mute in conference, the other two parties can still talk now if host is mute
SIP
  • blind xfer failure resumes the held call
  • added ROC (Roll over Counter) for SRTP
SETTINGS
  • restarting the phone during a not working network connection or loading settings manually will not reset the setting flag read only to read/write anymore
WEB
  • comboboxes for update_policy, answer_after_policy, call_waiting, aoc_amount_display, eth_net and eth_pc were not greyed out even though they were read only
MINIBROWSER
  • fixed bad xml prompt for action urls

6.2 branch

6.2.21 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
LID
  • fixed rare continuous dtmf scenario in inband dtmf
WEBCLIENT
  • fixed support for 'Transfer-Encoding: chunked'
Download Link

6.2.20 beta

GUI
  • fixed Busy Lamp for fkeys 5 and 6 (snom 300)
SIP
  • fixed incorrect processing of 'Replaces header'
  • fixed hanging on reboot

6.2.19 release

LID
  • fixed contrast (no cont. setting) for snom s300 starting with series 28XXXX
SIP
  • multiple REGISTERs over TCP were not all authenticated
  • fixed problem with lost subscriptions
GUI
  • message LED wasn't showing waiting messages properly

6.2.18 beta

GUI
  • show simple msg if Voice-Message header is not set in NOTIFY
  • handle MWI with same mw account and domain for different identities
Download Link

6.2.17 beta

LID
  • added support for new mac ranges 26-29
DNS
  • fixed memory leak of webclient request in case of non resolvable DNS URL

6.2.16 beta

GUI
  • fixed mute on 3-way conference issue
  • fixed hangup in transfer screen
  • don't join 2 calls inadvertently due to auto dial

6.2.15 beta

DST
  • Changed daylight savings time for USA beginning with 2007.
  • Fixed wrong DST calculation. Was starting at the wrong time (important especially for countries changing last sunday in march).
SIP
  • Dialog states are notified by the LEDs even if the monitored extension is called.

6.2.13

GUI
  • Reduced pickup timeout to 1 second.
SIP
  • 4xx codes in NOTIFYs were crashing the phone on call park

6.2.12 beta

SIP
  • use a new Call-ID after subscription is terminated

6.2.11 beta

SIP
  • stop packet retransmission for the packets associated with a finished call

6.2.10 beta

SIP
  • missing 'Expires: header' in 200OK in response to REGISTER is not causing subscription loss anymore
  • improved processing of NOTIFYs

6.2.9 beta

SIP
  • restore existing call on failed blind Xfer

6.2.8 beta

GUI
  • timezone Hawaii has no DST, fixed
SIP
  • fixed codec switch problem when 2nd party leaves conference
  • PRACK usage can be disabled via setting 'send_prack'
DHCP
  • just go on with DHCP DISCOVER for ever or press cancel

6.2.7 beta

GUI
  • fixed issue on blind Xfer through extension (fkey) with 2 calls
  • held call wasn't being resumed in hold-on-hold case

6.2.6 beta

SIP
  • avoid duplicate subscriptions on dial plan usage

6.2.5 beta

SIP
  • fixed park on occupied orbit
GUI
  • find free function key on call pickup

6.2.4 beta

DIALPLAN
  • dp usage was heavily slowing down the phone, fixed

6.2.3 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
SETTINGS
  • new settings application_version and linux_version showing the image version information respectively
  • removed wrong newline at the end of rootfs_version setting value
WEBCLIENT
  • added rootfs and linux version info to User-Agent line of GET request to ease mass deployment via php/perl etc.
Download Link

6.2.2 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
DHCP
  • ignore second OFFER if the phone already decided to follow the first one
GUI
  • HELP key isn't jumping anymore to factory reset if this was canceled before
  • maintain call list in case of overlap dialing
  • show Busy on 503 Service Unavailable
  • find right key when dialing from call lists
WEB
  • added clear missed calls on Cancel key setting
  • added clear desktop message on Cancel key setting
  • added 19222 to the emergency number list
SIP
  • fixed subscriptions on Reboot, Line Active/Inactive and Re-register
Download Link

6.2 release

IMPORTANT
  • Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
GUI
  • save volume on digit press
  • fixed redirection on timeout
  • fixed pre selection on offhook dialing
Download Link

6.1 beta

GUI
  • aditional check for reboot_after_nr
  • clear xml idle screen if new url failed
  • fixed blind Xfer with Phone Book
  • added emergency number support for keyboard lock
  • status line cleanup on idle display
  • reduced Notification time in absence of late media
MINIBROWSER
  • QueryStringParam and DefaultValue can evaluate phone settings (SnomIPPhoneInput)

6.0.5 beta

GUI
  • fixed logon wizard on PnP
  • show SMS for a particular call
  • skip PnP on Cancel key in wizard
  • redirection on timeout was sending a 486 after 302
LID
  • with SIP INFO DTMF, stopped sending outband and inband dtmf's in rtp, while playing them locally

6.0.4 beta

GUI
  • stop multiple calls if cwi is off
  • fixed cwi when handset is offhook in holding state
  • user=phone has no precedence in remote name display style
  • fixed priority knocking for multiple CWI
  • fixed DTMF echo on transfer
  • support F_TRANSFER:dest through fkey setup
  • use pre selection number for dialing in terminated state
  • improved ringtones 2-10
  • fixed phone book key programming
  • send INFO dtmf for basic keys only
  • added blind Xfer with the help of Phone Book
LID
  • fixed bad speaker click problem (due to initial srtp pkts), discard rtp packets with wrong lengths (boundary)
SIP
  • offer dial prompt on 503 Service Unavailable
  • use p-asserted-identity as display name if available in INVITE/18x/200
  • fixed pnp settings retrieval mechanism, phone is waiting 10 seconds at bootup for SIP PnP
  • use the same port number for SIP tcp/udp
WEB
  • fixed Load Settings Manually via web interface
  • Message LED for Dialog States/Missed calls is setting dependent
  • TLS
  • added 512-bit certificate support
  • NTP
  • fixed rare DST problem, where the phone was showing time without daylight saving

6.0.3 beta

GUI
  • support multiple remote name changes
  • added unsecure transport indication
SIP
  • fixed wrong behavior on 422 response

6.0.2 beta

GUI
  • shortend MWI text
  • added new Number Display Style 'Number + Name'
SIP
  • added setting to enable xml notify handling
  • added setting to enable supported timer (RFC4028)

6.0.1 beta

GUI
  • fixed call costs and duration in lists
  • added congestion tone
  • fixed up scrolling for short lists
SIP
  • fixed terminate subscriptions on reboot
  • added setting to control local display name transfer on incoming calls
MINIBROWSER
  • better error response handling

6.0.0 beta

GUI
  • added XML minibrowser support
  • fixed desktop SMS
  • title texts configurable through mass deployment
  • line based ring after delay for incoming calls
  • changed MWI led behavior to reflect latest vmail status
  • fixed first dtmf echo on dialing
  • deny list now works for complete uri match using identity context
  • fixed Name+Number for numeric display names
  • double redial key press to redial last call
  • set led's for associated key events
  • DND icon has high priority on status line
  • use identity of connected call for parking/Xfer
  • enhanced spanish language support
  • reset to factory defaults restarts the phone, too
  • fixed speed dial on Xfer
  • redirection on timeout has its own target parameter now
WEB
  • reset to factory defaults restarts the phone, too
  • added new reboot info page
  • clean up e164 numbers dialed through web interface
  • added star codes based dnd/redirection
  • directory key is programmable
  • added control to filter out SIP tracing for REGISTER/SUBSCRIBE/NOTIFY
  • added ACTIVE option for fkey context selection
LID
  • adjusted knocking (call waiting) volume
  • fixed local playback of first dtmf in handsfree mode, also fixed and cleaned overall local playback of DTMF in different cases
  • fixed a conference bug (phone slowed down after a conference)
SIP
  • intercom/Push2Talk calls send an Alert-Info header
  • added server hosted conference feature
  • use display name if available in 180 Ringing
  • send NOTIFY on receiving an unsubscribe request
  • added pnp settings retrieval

5.5.1 release

IMPORTANT
  • If your phone is running this FW version please update ONLY via the Update Wizard
GUI
  • fixed consultative Xfer with fkeys
Download Link

5.5 release

IMPORTANT
  • If your phone is running this FW version please update ONLY via the Update Wizard
GUI
  • fixed cursor handling (scrolling, backspace) in edit number state
  • put last active call on hold on top in holding/transfer
Download Link

5.4 beta

GUI
  • added shared line LED blink when holding
SIP
  • fixed bug in ENUM lookup
LID
  • fixed port access for keep_alive where it could access a port that didn't exist anymore

5.3.6 beta

LID
  • made sure audio channels are off in idle mode under all scenarios

5.3.5 beta

IMPORTANT
  • For updating from release 4, please see our Update Instructions at the new wiki
GUI
  • added cwi ringer indication
  • fixed unnecessary dialog state switches on shared line offhook
  • status led for missed calls
SIP
  • RAck in PRACK was buggy
  • added call pickup for shared lines

5.3.4 beta

SIP
  • added +sip.rendering parameter for BLA hold/resume NOTIFYs
  • NOTIFYs with subscription-state: terminated remove the subscription

5.3.3 beta

GUI
  • fixed DND
  • fixed bug in displaying old voice mail messages
SIP
  • display local LED status for shared lines
WEB
  • "+" in settings value isn't anymore replaced by its hex value on settings dump web interface page
  • further enhanced french translation
SRTP
  • fixed bug with auto-answer

5.3.2 beta

GUI
  • setting_server can be set manually via GUI menu
  • ringer device should not switch to speaker if headset is enabled
  • dkeys (e.g. Redial, Retrieve) are working in edit number state, too
Settings
  • if setting_server is IP:port only, make a valid URL out of it
SIP
  • display local LED status for shared lines

5.3.1 beta

GUI
  • Shared Lines can be mapped to LEDs
LID
  • random number generated from random audio data

5.3 beta

GUI
  • blind-xfer via programmable keys doesnt require pressing the Enter key
  • incoming call context can be switched with the navigation key
  • fixed freezing during calls on hold
  • added setting cancel_on_hold which, if set to false, makes the phone ignore any cancel key press in holding state
  • fixed DND, wasn't working properly after reboot during DND on
  • enhanced french translation
  • fixed, mute key stops working after 20 seconds if no DNS server is reachable
LID
  • further reduced ringer volumes
SIP
  • unsupported p-time values for codecs in responses disconnects the call
  • treat all return codes > 100 and < 180 as 180 Ringing
WEB
  • enhanced french translation

5.2 beta

GUI
  • made F-key type SpeedDial work even with handset offhook
  • fixed unwanted conference bug in offhook/enter during ringback with an incoming call
  • blind transfer destination can be entered even with multiple calls
  • improved czech language support
WEB
  • added czech language support
LID
  • made speaker volume linear
  • improved handsfree adaptation speed to remove voice chopping
SNMP
  • improved interoperability
SIP
  • blind transfer is release ===d on a 180 notification
SRTP
  • reversed SSRC byte ordering, which is not compatible to previous versions anymore

5.1 beta

GUI
  • reinstated old Ringer 2 & 4 as Ringer 9 & 10
  • trigger action url disconnected on calling and on cancel
  • reduced ringer volumes
  • fixed DST for southern hemisphere
LID
  • fixed an audio problem (sometimes garbled audio, appeared in 5.0)
  • added DTMF A-D and flash to be sent as outbound dtmfs
WEB
  • added keyevent F_AUTO_ANSWER
  • added auto logout for web access

5.0 beta

GUI
  • added line dependend ringtone selection to Line menu
  • added keyboard lock feature
  • don't jump to the first addressbook entry if one entry has been removed
  • reply to a wrong password for factory reset by asking the password again
  • fixed wrong status messages for hold during conference
  • allow making calls through programmable keys while ringing
  • clear failed/closed channels on incoming calls to conserve lines/keys
  • allow incoming calls on offhook, hold and ringback states
  • fixed initial list positioning to the top of the list for tone scheme and timezone
  • fixed return to CC from menus
  • fixed cwi on talking to attended transfer with programmable keys
  • added volume level stats display
  • improved key mapping for Destination/dialog states
  • fixed different melodies for multiple ringing calls at the same time
  • fixed wrong status message for hold during conference
  • use of backlight for longer duration
  • added dialplan lookup on number dialing in idle state
  • support Reply-To in call lists if set
  • Ringer volume is independent from Speaker
  • partial number matching looks for complete match at the end
  • fixed duration counter in received records
  • fixed stutter dial tone mixing with dtmf echo
  • small fix of french translation
  • lookup addressbook entry for user if uri match is not found
  • fixed tbook jump index by ignoring deny list items
  • switching of outgoing lines through programmable keys in edit mode
  • lookup number match in uri of call list entry
  • consolidated access to remote number display name
  • fixed return from blind transfer
  • exact match for display name when dialing out
  • fixed missing media stream while changing contrast
  • show pressed digits sent out as dtmf during connected state
  • last active call is shown first in multi party transfer
  • missed call counter is cleared on any key press in idle
  • improved preset ring melodies on the phone
  • added extended ascii characters for font_verdana
  • expanded dialog info call statistics on idle screen display
  • added support for Shared Line Appearance
  • fixed DST beginning date of Australian timezones
  • added timezones for Brazil and Argentina
  • added missing Australien time zones
  • added missing African timezones
  • added timezone for Trinidad & Tobago
  • using line context for fkey subscriptions
  • avoid indefinite loops by not redirecting to same local number
  • added record indication icon
  • record key is sending on/off
  • fixed find fkey for multiple lines on CWI
  • speaker key on speakerphone hangs up the call
  • dont show sip url if remote name/number are set, disabled exact host matching for display method
  • added text only soft keys
  • xml screen on idle display supports status info and soft keys
  • ringer animation can be switched off to save space for incoming number
  • fixed sudden melody restart during ringing
  • logon_wizard setting "off" means no logon wizard at all
  • use address book item assigned outgoing id even if dialing from addressbook via display menu
  • fixed cancelation of password input for factory reset
  • fixed new line on ringing and held calls
  • dont show xfer dest on key_left while browsing for xfer source
  • CANCEL disconnects only if there is one call on hold
  • added overlap dialing
  • added basic LDAP support
LID
  • RTCP added (preliminary)
  • Collect stats and pass them to LCS at end of call
  • Display graphically volumes of tx and rx pkts (mic, spkr) in real time
  • Mute fixed, dtmf's also stop on mute now
  • first keep_alive packet is sent immediatey now
  • stats can be requested at any time from the top
  • different warnings added like high pkt loss, high jitter, high fraction loss etc.
  • added real time packet counter (tx, rx) to the info screen
  • fixed conferencing - audio doesn't stop between the remote parties on change of device now
  • a general fix for G711 (pkt len), also fixed conf. problem on 60 ms
  • added codec info to the info screen
WEB
  • addressbook: jump to edit section if edit item was clicked
  • adressbook numbers are stored without spaces only
  • moved symmetrical rtp to rtp tab
  • moved outbound proxy to login tab
  • hash can now properly be dialed via command.htm
  • added upload setting file manually via webpage
  • enhanced outgoing id display for address book and call lists
  • trusted certificate page not anymore visible in user mode
  • fixed timezone naming for USA-Hawaii, -Alaska, -Pacific
  • added Action URL as programmable key choice
  • number guessing is off by default
  • added number guessing minimum length
  • added action URL for missed calls
  • added translation of mic volume texts
  • re-added a STUN timer interval (independend from keepalive)
  • added support for variables (settings, $local/$remote) in Action urls
  • added support for displaying call costs in call lists
  • added auto-answer policy (off, in idle, always)
  • redesigned fkey page
  • added Key Event for programmable keys
  • added logoff all button to SIP line login pages
  • added line logoff button to SIP line login pages
  • added setting for auto logoff all lines on inactivity
  • HANGUP button now really hangs up the call
  • flash plugin redial link was missing the port
  • speaker key dialing is associated with a setting, on by default
  • more programmable keys support with existing dedicated keys
  • added action url "On Connected" and "On Disconnected"
  • support call-id variable in action urls
  • added more fkeys and fkey types
SIP
  • several Session-Timer improvements
  • added Reason header (RFC3326) support for CANCEL
  • fixed replies for MESSAGE within a dialog
  • added support for multiple comma separated ENUM suffixes
  • added support for draft-ietf-sip-outbound-00
  • send NOTIFY on re-subscribe
  • added DTMF signalling via SIP INFO messages
SNMP
  • detects now broken connections and changed external addresses
  • added missing source address and changed address attributes to bind response (STUN/ICE)
LINUX
  • added eth switch control

4.5 release

GUI
  • hook switch debounced
  • handle challenge request during state_enum properly
  • logon_wizard is not hanging anymore if using mass deployment setup
  • graceful exit when held call is closed remotely on Xfer
  • reject with 480 Do Not Disturb on DND
  • fixed ringing stutter caused by register responses
Download Link

4.4 release

SETTINGS
  • phone specific settings file wasn't read in anymore, fixed
  • reduced default mic volumes for handset and speakerphone
GUI
  • unregister while logging off an account
  • fixed find fkey for multiple lines
SIP
  • no hardwired default value for session timer
  • DHCP
  • set siaddr field to 0 instead of ffffffff
  • added support for sname and field options
WEB
  • added programmable key event
Download Link

4.3 release

SIP
  • fixed reply for MESSAGE within a dialog
LICENSE
  • retrieve license file anytime unless phone is licensed allready
SRTP
  • fixed
Download Link

4.2 beta

GUI
  • parallel challenge requests can now handled properly
  • small fix of french translation
WEB
  • added translation of mic volume texts
  • fixed timezone naming for USA-Hawaii, -Alaska, -Pacific
SIP
  • STUN and CRLF keepalives in parallel
  • fixed do not append domainname to authusername
LID
  • fixed extension keyboard led control problem
SETTINGS
  • config files can be loaded via TFTP also
  • compatible to TFTP config files
  • new settings flag '$', overwrites everything on each reboot, but settings stay writable

4.1 release

GUI
  • fixed CANCEL on confirm screen
  • fixed wrong status messages for hold during conference
  • partial number lookup is now setting dependent
WEB
  • added microphone volume control
  • added timezone for thailand
SIP
  • STUN results were not used in SDP
  • fixed potential endless loop in dialplan
  • added callerid support
  • fixed reboot=false in check-sync prevents reboot
LID
  • introduced mic volume controls for all mics
SETTINGS
  • fixed mass provisioning of license_url
  • TLS
  • server_hello with zero length session ID wasn't working
Download Link

4.0 release

GUI
  • fixed initial list positioning to the top of the list for tone scheme and timezone
  • fixed not needed showing of copyright screen
  • enhancements of french translation
  • show cursor on number guessing also
  • added audio device indication for all critical states
  • only 10 character pickup number hint is displayed
  • disabled partial number lookup from address book
  • holding reminder is now setting dependent (#789)
  • local keytone generation e.g. menu keys
  • fixed repeated park/pickup freeze
  • volume up/down keys in idle state play static ring melody
  • fixed input mode problem while editing address book entries
  • confirmation screen for adress book item deletion
  • added support for Xfer on incoming calls (ringing/cwi)
  • fixed call duration display shift
  • added desktop message support
  • screen showed still connected after transfer was rejected
  • fixed asking for password on a challenge request
  • fixed cwi on talking to attended transfer with programmable keys
  • clear failed, closed channels on incoming call
  • fixed remote call close on multiple held calls
  • dialing last entry from redial list works again
WEB
  • enhancements of french translation
  • added setting to control audio device indication on display
  • added icons for clearing the call lists
  • added "add entry from call list to address book" buttons
  • got rid of the remaining parameters in the browser URL line
  • added outgoing id to address book item and call lists
  • added reregister button for SIP lines
  • show local identity in call lists
  • recording missed calls is line dependant
  • added confirmation page for reboot, reset, delete address book
  • added line ringer playback from web interface
  • ethernet unplug detection is controlled by a setting
  • line active/inactive feature added
  • phone numbers with leading + can be handled properly via web page
  • added support for "," as separator for the AOC cost pulse
  • trusted certificate page not anymore visible in user mode
LID
  • added silence suppression (CNG/VAD)
  • added local keytone generation
  • added/enhanced sidetone
  • added license check
  • fixed voice lag problem
  • fix to correctly play incoming packets of various ptime on dif. codecs
  • led's synchronized
  • fixed a DTMF problem
SIP
  • fixed multiple authentication of INVITEs
  • fixed subscription deletion which leads to freezing on sw update
  • re-activated silence compression setting
  • support 12 local line registrations
  • fixed outgoing requests without a port in the Contact, especially in REGISTER
  • fixed missing IP address in SDP (o and c line)
  • keepalive is now independent from the STUN server
  • Expires header in Subscriptions and replies was not evaluated
  • added CRLF keepalive instead of sending STUN requests
  • fixed problem where connections have been removed to early
  • fixed re-newing of subscriptions duplicated the number of dialog states
  • fixed STUN wasn't working any more after re-registration
Download Link

3.60x release

Download Link

3.60s release

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3.60q beta

GUI
  • enhancements of french translation
  • added desktop message support on status line
WEB
  • enhancements of french translation
SIP
  • fixed a wrong on-hold state with MPO
  • expires header in SUBSCRIBE and its replies will be evaluated

3.60p beta

GUI
  • fixed to long timeout for terminated screen after canceled transfer with AOC
SIP
  • fixed subscription deletion which leads to freezing on sw update
  • fixed wrong domain names in REGISTER after 3xx message
LID
  • fixed voice lag problem

3.60m beta

GUI
  • disabled partial number lookup from address book
SIP
  • IP dialing reenabled

3.60l beta

GUI
  • fixed multiple incoming calls on function keys
SIP
  • parsing error fixed in SIP URI
LID
  • keep alive mechanism changed, fixed g726 problem (second call ringing)

3.60k beta

GUI
  • fixed offhook call jump problem on multiple incoming calls
  • fixed led for incoming call on terminated line set to a function key
SIP
  • stale NOTIFY's will be rejected now
  • rejected NOTIFY's will delete the subscription
  • subscription termination via NOTIFY are now processed
  • fixed SDP version number was not increased in 200 OK
  • removed un-necessary DNS lookup for STUN if ICE support is off
  • added support for 3xx responses on REGISTER requests
  • subscription dialog state will be periodically verified
LID
  • added support for receiving 10ms and 30ms RTP packets

3.60i release

GUI
  • fixed wrong first line address book lookup match independent from the number
  • DST works now for absolute dates (e.g.Iran), too
  • first line as title of a selectionbox can not be selected anymore
  • enhanced displaying of selected selctionbox items, if the last one or the one before is selected
  • change of backlight setting is recognized immediately
WEB
SIP

line status appears now at sysinfo page

  • flash plugin usage on by default
  • 24 hours time format option makes sense for big display (call lists etc.) also
SIP
  • fixed missing new SUBSCRIBE when function key destination was changed
  • added un-SUBSCRIBE for the former function key destination
  • registrations and subscriptions are now cleaned up before reboot
  • ignore Record-Route from PRACK
LID
  • fixed a problem with infinite DHCP lease time
Download Link

3.60h beta

GUI
  • fixed CALL-INFO answer-after
SIP
  • SDP Offer/Answer changes for improved third party call control
  • by default turn ICE off
  • fixed disturbed dialtone when non-RFC3264 devices on-hold
LID
  • added sending media keep alive packets (STUN requests)

3.60g beta

GUI
  • MWI count was wrong according to RFC
  • fixes for danish texts
  • knocking for priority cwi
WEB
  • fixes for danish texts
SIP
  • reboot on check-sync can now only be avoided by adding parameter reboot=false
LID
  • fixed RTP stream wasn't following re-INVITE (MPO)
  • speaker gains increased for handset and headset
  • tone generation improved for easily playing different kind of tones
  • fixed missing audio when entering CMC

3.60f beta

GUI
  • fixed Ringer1 for SIP lines
  • fixed edit mode problem for Xfer on AddressBook jump
WEB
  • added new option to use SIP compact headers
SIP
SIP
  • stack should not unnecessarily prevent reboot any more
  • added missing Expires header in SUBSCRIBE reply
  • added missing Subscription-Status header to NOTIFY
  • fixed bad Contact in SUBSCRIBE reply
  • fixed problem when the ACK was missing
  • fixed problem with stale connections when call was rejected
LID
  • fixed increasing timestamps for out-of-band DTMF

3.60d beta

GUI
  • speed dial support for Xfer
  • Ringer device for headset is only used for ringing and not CWI
  • fixes for non peer to peer pickup
  • added auto exit on volume change
  • fixed irish timezone DST
  • look for partial number match in Address Book lookup
  • added Auto Answer per SIP line (#692)
WEB
  • dont show user passwords on settings log
  • fixes for french texts
  • added DTMF for programmable keys in SIP (#732)
LID
  • added multiple key generation set by programmable keys
  • corrected RTP playout time
SIP
  • wasn't respecting DNS NAPTR records with sips content
  • DNS
  • wasn't respecting the retry-timeout for not-found domain names, was using 60 seconds always

3.60b release

LID
  • echo cancellation added
  • audio sub system improved, headset volume increased

3.60a beta

GUI
  • added Norwegian NLS
LID
  • codecs and packet sizes tuned
  • extended keyboard support added
  • audio problems in transfers, conferencing, one way audio etc. fixed

3.56z release

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3.56y release

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3.52 release

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