From Snom User Wiki
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
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- 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"
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| Localization
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- 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
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| Miscellaneous
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- 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'
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| Download Link
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6.5.17 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
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| WEB
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- values of disable_blind_transfer and disable_deflection were not properly reflected at the advanced web interface page
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| LID
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- fixed bug where subscribtions to extensions registered to exp-keypad-keys weren't send out
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| Download Link
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6.5.16 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
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| SIP
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- use request URI when selecting line key
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6.5.15 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
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| GUI
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- port 5061 was being dropped from URI
- improved transfer handling
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| SIP
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- call completion subscriptions should not be delayed
- fixed potential crash when deregistering mailbox
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| Download Link
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6.5.14 beta
| WEBCLIENT
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- sometimes building of proper destination URL failed, so provisioning wasn't working
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| LID
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- making sure sending packets contain negotiated codec data only (to avoid a rare diff. codec packet)
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| SIP
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- fixed missing ack of bye after transfer
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6.5.13 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
|
| WEBCLIENT
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- 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
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| SIP
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- fixed incorrect processsing of Replaces header
- fixed crash during reboot
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| WEBCLIENT
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- fixed support for Transfer-Encoding chunked
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| LID
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- enhancement for update to v7 (new bootloader header for smoother v6 to v7 transition)
- fixed the rare continuous dtmf problem in inband dtmf
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| Download Link
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6.5.12 beta
| GUI
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- calls on hold are shown at a higher priority than incoming calls
- programmable key TRANSFER could not be remapped
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| SIP
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- 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
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| LID
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- fixed initial skipping of audio packets problem (in asterisk)
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6.5.11 beta
| GUI
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- show simple msg if Voice-Message header is not set in NOTIFY
- handle MWI with same mw account and domain for different identities
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| LID
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- added support for new mac range 29
- enhancement for update to v7
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| SIP
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- fixed bug where CANCEL was suppressed in some transfer scenarios
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6.5.10 beta
| DST
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- corrections for Spain and Portugal
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| LID
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- added ability to update to v7
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6.5.9 beta
| GUI
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- 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)
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| SRTP
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| LID
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- improved random number generation
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| TLS
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- added certificate parsing patch
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| MINIBROWSER
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- hook off in SnomIPPhoneDirectory can dial number
- fixing wrong error message after dialing a number in SnomIPPhoneDirectory
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| DNS
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- fixed memory leak of webclient request in case of non resolvable DNS URL
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6.5.8 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
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| DST
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- Changed daylight savings time for Canada beginning with 2007.
- Timezone Hawaii has no DST.
- Added timezone USA -7 MST no DST.
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| SIP
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- Fixed illegal connection erase with 2 calls on hold.
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| WEB
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- Added 19222 to the emergency number list.
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| Download Link
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6.5.6 beta
| DST
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- 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).
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| SIP
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- new setting user_full_sdp_answer sends codec interception in 200OK
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6.5.5 beta
| GUI
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- fixed voice recording in idle state
- added wireless headset active indication (snom360)
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| SIP
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- 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)
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| LID
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- wireless headset support added (snom360)
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6.5.4 beta
| GUI
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- fixed duration on caller id
- fixed auto leave edit number on offhook
- fixed speaker key on handsfree anomaly
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6.5.3 beta
| GUI
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- don't join 2 calls inadvertently due to auto dial
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| SIP
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- fixed codec switch problem when 2nd party leaves conference
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| TLS
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- sanity checks added for rsa keys
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| WEB
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- use stk_with_flash setting for setting up the XML port
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6.5.2 release
| GUI
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- 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)
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| WEB
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- do not clear the given number for fkeys even if context is not registered
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| Download Link
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6.5.1 beta
| SIP
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- fixed subscription refresh on reboot (if subscription_delay not set)
- new setting terminate_subscribers_on_reboot added to switch off unSUBSCRIBEs on reboot
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6.5 beta
| GUI
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- 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
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| WEB
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- 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
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| SIP
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- 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
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| PROVISIONING
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| DIALPLAN
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- dp usage was heavily slowing down the phone, fixed
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| DHCP
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- initially go on with DHCP DISCOVER for ever or press cancel
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| MINIBROWSER
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- Action URL Settings can drive minibrowser
- SNMP
- registration status is working now
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6.3 beta
| GUI
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- 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
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| LID
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- fixed mute in conference, the other two parties can still talk now if host is mute
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| SIP
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- blind xfer failure resumes the held call
- added ROC (Roll over Counter) for SRTP
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| SETTINGS
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- 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
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| WEB
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- 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
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| MINIBROWSER
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- fixed bad xml prompt for action urls
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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
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| WEBCLIENT
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- fixed support for 'Transfer-Encoding: chunked'
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| Download Link
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6.2.20 beta
| GUI
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- fixed Busy Lamp for fkeys 5 and 6 (snom 300)
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| SIP
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- fixed incorrect processing of 'Replaces header'
- fixed hanging on reboot
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6.2.19 release
| LID
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- fixed contrast (no cont. setting) for snom s300 starting with series 28XXXX
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| SIP
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- multiple REGISTERs over TCP were not all authenticated
- fixed problem with lost subscriptions
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| GUI
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- message LED wasn't showing waiting messages properly
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6.2.18 beta
| GUI
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- show simple msg if Voice-Message header is not set in NOTIFY
- handle MWI with same mw account and domain for different identities
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| Download Link
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6.2.17 beta
| LID
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- added support for new mac ranges 26-29
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| DNS
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- fixed memory leak of webclient request in case of non resolvable DNS URL
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6.2.16 beta
| GUI
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- fixed mute on 3-way conference issue
- fixed hangup in transfer screen
- don't join 2 calls inadvertently due to auto dial
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6.2.15 beta
| DST
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- 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).
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| SIP
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- Dialog states are notified by the LEDs even if the monitored extension is called.
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6.2.13
| GUI
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- Reduced pickup timeout to 1 second.
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| SIP
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- 4xx codes in NOTIFYs were crashing the phone on call park
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6.2.12 beta
| SIP
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- use a new Call-ID after subscription is terminated
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6.2.11 beta
| SIP
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- stop packet retransmission for the packets associated with a finished call
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6.2.10 beta
| SIP
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- missing 'Expires: header' in 200OK in response to REGISTER is not causing subscription loss anymore
- improved processing of NOTIFYs
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6.2.9 beta
| SIP
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- restore existing call on failed blind Xfer
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6.2.8 beta
| GUI
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- timezone Hawaii has no DST, fixed
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| SIP
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- fixed codec switch problem when 2nd party leaves conference
- PRACK usage can be disabled via setting 'send_prack'
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| DHCP
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- just go on with DHCP DISCOVER for ever or press cancel
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6.2.7 beta
| GUI
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- fixed issue on blind Xfer through extension (fkey) with 2 calls
- held call wasn't being resumed in hold-on-hold case
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6.2.6 beta
| SIP
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- avoid duplicate subscriptions on dial plan usage
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6.2.5 beta
| SIP
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- fixed park on occupied orbit
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| GUI
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- find free function key on call pickup
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6.2.4 beta
| DIALPLAN
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- dp usage was heavily slowing down the phone, fixed
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6.2.3 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
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| SETTINGS
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- new settings application_version and linux_version showing the image version information respectively
- removed wrong newline at the end of rootfs_version setting value
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| WEBCLIENT
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- added rootfs and linux version info to User-Agent line of GET request to ease mass deployment via php/perl etc.
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| Download Link
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6.2.2 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
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| DHCP
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- ignore second OFFER if the phone already decided to follow the first one
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| GUI
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- 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
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| WEB
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- added clear missed calls on Cancel key setting
- added clear desktop message on Cancel key setting
- added 19222 to the emergency number list
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| SIP
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- fixed subscriptions on Reboot, Line Active/Inactive and Re-register
|
| Download Link
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6.2 release
| IMPORTANT
|
- Updating is only possible from v5 or newer releases, please see our Update Instructions at the wiki
|
| GUI
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- save volume on digit press
- fixed redirection on timeout
- fixed pre selection on offhook dialing
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| Download Link
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6.1 beta
| GUI
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- 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
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| MINIBROWSER
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- QueryStringParam and DefaultValue can evaluate phone settings (SnomIPPhoneInput)
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6.0.5 beta
| GUI
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- 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
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| LID
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- with SIP INFO DTMF, stopped sending outband and inband dtmf's in rtp, while playing them locally
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6.0.4 beta
| GUI
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- 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
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| LID
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- fixed bad speaker click problem (due to initial srtp pkts), discard rtp packets with wrong lengths (boundary)
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| SIP
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- 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
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| WEB
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- 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
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6.0.3 beta
| GUI
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- support multiple remote name changes
- added unsecure transport indication
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| SIP
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- fixed wrong behavior on 422 response
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6.0.2 beta
| GUI
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- shortend MWI text
- added new Number Display Style 'Number + Name'
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| SIP
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- added setting to enable xml notify handling
- added setting to enable supported timer (RFC4028)
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6.0.1 beta
| GUI
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- fixed call costs and duration in lists
- added congestion tone
- fixed up scrolling for short lists
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| SIP
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- fixed terminate subscriptions on reboot
- added setting to control local display name transfer on incoming calls
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| MINIBROWSER
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- better error response handling
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6.0.0 beta
| GUI
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- 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
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| WEB
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- 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
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| LID
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- 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)
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| SIP
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- 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
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5.5.1 release
| IMPORTANT
|
- If your phone is running this FW version please update ONLY via the Update Wizard
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| GUI
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- fixed consultative Xfer with fkeys
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| Download Link
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5.5 release
| IMPORTANT
|
- If your phone is running this FW version please update ONLY via the Update Wizard
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| GUI
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- fixed cursor handling (scrolling, backspace) in edit number state
- put last active call on hold on top in holding/transfer
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| Download Link
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5.4 beta
| GUI
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- added shared line LED blink when holding
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| SIP
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| LID
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- fixed port access for keep_alive where it could access a port that didn't exist anymore
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5.3.6 beta
| LID
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- made sure audio channels are off in idle mode under all scenarios
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5.3.5 beta
| IMPORTANT
|
- For updating from release 4, please see our Update Instructions at the new wiki
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| GUI
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- added cwi ringer indication
- fixed unnecessary dialog state switches on shared line offhook
- status led for missed calls
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| SIP
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- RAck in PRACK was buggy
- added call pickup for shared lines
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5.3.4 beta
| SIP
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- added +sip.rendering parameter for BLA hold/resume NOTIFYs
- NOTIFYs with subscription-state: terminated remove the subscription
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5.3.3 beta
| GUI
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- fixed DND
- fixed bug in displaying old voice mail messages
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| SIP
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- display local LED status for shared lines
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| WEB
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- "+" in settings value isn't anymore replaced by its hex value on settings dump web interface page
- further enhanced french translation
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| SRTP
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- fixed bug with auto-answer
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5.3.2 beta
| GUI
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- 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
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| Settings
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- if setting_server is IP:port only, make a valid URL out of it
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| SIP
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- display local LED status for shared lines
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5.3.1 beta
| GUI
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- Shared Lines can be mapped to LEDs
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| LID
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- random number generated from random audio data
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5.3 beta
| GUI
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- 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
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| LID
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- further reduced ringer volumes
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| SIP
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- unsupported p-time values for codecs in responses disconnects the call
- treat all return codes > 100 and < 180 as 180 Ringing
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| WEB
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- enhanced french translation
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5.2 beta
| GUI
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- 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
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| WEB
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- added czech language support
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| LID
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- made speaker volume linear
- improved handsfree adaptation speed to remove voice chopping
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| SNMP
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- improved interoperability
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| SIP
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- blind transfer is release ===d on a 180 notification
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| SRTP
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- reversed SSRC byte ordering, which is not compatible to previous versions anymore
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5.1 beta
| GUI
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- 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
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| LID
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- fixed an audio problem (sometimes garbled audio, appeared in 5.0)
- added DTMF A-D and flash to be sent as outbound dtmfs
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| WEB
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- added keyevent F_AUTO_ANSWER
- added auto logout for web access
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5.0 beta
| GUI
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- 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
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| 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
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| WEB
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- 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
|
|
|
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
|
|
|
| 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
3.60s release
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
|
|
|
3.60m beta
| GUI
|
- disabled partial number lookup from address book
|
| SIP
|
|
|
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
|
|
|
| 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
|
|
|
| LID
|
- codecs and packet sizes tuned
- extended keyboard support added
- audio problems in transfers, conferencing, one way audio etc. fixed
|
3.56z release
3.56y release
3.52 release