Snom 760 IP Phone Firmware 8.7.5.8 Beta
Dosya tipi: Firmware.
Işleti̇m Si̇stemi: AnyOS.
Dosya adı: snom760-8.7.5.8-SIP-r.bin.
Bit: 32/64 Bit.
Li̇sans: Ücretsiz.
Açıklama
Düzeltmeler:
– Phone duplicates not supported media type (video) on Re-Invite (Session Timeout)
– Incorrect REFER causes server based conference to fail with 3CX
– Invite responded too incorrectly before the ack of 180 ringing that included 100rel
– Incorrect response to identity-request with blank credentials
– 302 Moved Temporarily during call
– RTP – SSRC / Sequence Number not correct
– Hold fails because of different DTMF paylod types 101<->96
– No DTMF outband events are send using the snom conference server
– Broadsoft/BLF does not work anymore with default function key settings
– Unauthorized for REFER will not be resend if user_moh is being used
– RegEx dial plan affects also SUBSCRIBE URI
– DTMF SIP INFO not working in early-media state
– Wrong DST port is used for voicemail if the registrar is not using the default port 5060
– PUBLISH contains a body in refresh requests
– Park Service is shown as Connected Number for forked calls with the same number
– Phone looses presence subscription after receiving any response code 4xx without expires or expires 0
– If the registrar is set to use a different port (e.g user_host = 192.168.1.1:6050), the port is only used for registration but not for INVITEs
– Phone loses Registration with Astimax
– OpenSips: Phone does not register due to ignored Expires header, gruu
– P-Asserted-Identity number part isn’t recorded in call history
– Unable to initiate attended transfer if multiple identities are involved
– Broadsoft/Timer Support: phone adds required header even when not necessary / not advised
– Broadsoft: adhoc conferencing not working
– Broadsoft/BLF: Setting up Contact List Buddy needs Re-Register before working
– Broadsoft: phone shows HOLD state even when HOLD Request was rejected / Unhold not possible
– Broadsoft/AOC: phone shows AOC Screen after Call even when not activated
– MWI indication will be stopped to early
– Ring tone disappears in certain situations
– DTMF Tones: missing audio at the beginning of IVR announces (circa one second)
– Sometimes 821/870 phones are playing disturbing noise on speaker in idle state
– Call Deflection: Current Call is put on HOLD instead of 2-way-audio while deflecting the other call
– Call Deflection: Dialtone is played out when deflecting a call
– Disabled cw_dialtone has no effect. Dial tone will always be played when setting a call on hold. (snom 870 only)
– MWI Indication beeps strange in idle
– Multicast Paging Play out doesn’t work (snom 7xx only)
– Shared Line Identity plays short “Busy Tone” during first entered digit. Possible “line-seize” authorization issue.
– Early Media DTMF (RTP Out of Band) does not work with specific tone schemes
– Directory Ring tones don’t work as expected
– The first DTMF tone is missing during hold when using speaker (snom 710 only)
– One way audio with G729 codec after selecting one conference participant, pressing OK and then resuming the conference
– Phone sends 711u codec on blind transfer to IMS mobile
– Muted connected call gets unmuted when another incoming call stops ringing
– If DTMF keys are pressed very fast in a call, the remote phone sometimes didn’t recognize the outband DTMF events correctly.
– Phone stops receiving multicast after receiving an incoming call via handset
– Dial tone being played and transfer message blocking screen for some seconds after a successful blind transfer
– Incoming Call while automatic redial on busy results in ringing tone restart every second
– Pickup_indication doesn’t works any more
– Broadsoft: (Call Center Hold Reminder) Speaker is not ringer device for second incoming call
– G729 codec choice incorrectly includes annexb=no in SDP
– TR69: GetParameterNames with NextLevel=1 results in crippled ParameterNames
– Parammap for all phone types needs to be updated
– Provisioning sometimes does not work when the network uses STP and static IPs for the phones
– Provisioning in progress message is not shown during provisioning
– If DNS fails for some time NTP will not be renewed and registration fails
– Using the attribute complete=”true” in tbook provisioning leads to changing some settings like languages, date format etc.
– Provisioning via TR-069 (Friendly) is not initialized correctly
– Display freezes after provisioning failed via timeout (snom 320 only)
– Jabra Wireless Jabra PRO 9470 doesn’t stop ringing on unanswered Inbound Calls
– Wireless headset Wireless Jabra PRO 920, EMEA doesn’t work properly
– DHSG mode at GN9330 results in delayed audio connection to opposite
– Wireless headset doesn’t work properly, audio is played via handset instead of headset and call is disconnected just press the headset button twice
– USB Headset Input is not consistent/working with phone (snom 7xx only)
– Headset: Endless toggling between Headset and Speaker on Plantronics Wireless SAVI W745/A
– No Audio on Outgoing Calls with Wireless Jabra and Plantronics headsets on snom 7xx
– When a firmware update fails the phone does not automatically restart (snom 300 only)
– Enabling Wifi Ethernet Bridge leads to phone being stuck in sending DHCP requests
– DHCP – PA1 identifies itself as Snom300 (Vendor Class Identifier)
– DHCP Lease renewal: Phone does not reboot if NACK from server
– Phone forgets VLAN setting if downgrade to firmware version with old VLAN setting format
– VLAN – Prio not working if ToS is set up
– ToS Out-Of-Range and no Reboot indication
– When the 802.1X-Authentication fails on the phone it still boots up but without a working IP address
– Snom 710 doesn’t forward EAP packets to the PC port
– If DNS fails for some time NTP will not be renewed and registration fails
– Freeze after changing network ID port (snom 820 only)
– Presence doesn’t work after the SIP proxy was down
– Mmake outgoing call using web GUI from call list doesn’t work
– Missing Login box for entering admin_mode in WUI
– WUI Call Lists – when clearing “Dialled Numbers” from caller lists the whole list from received/missed gets cleared.
– Switching account details for a SIP identity that first didn’t have an outbound proxy to one that does, will fail to send the unregister to the correct IP
– Phone always adds <– active identity to the name of a number when a contact without a name is added to the directory
– Spelling mistake in download parameter mapping
– IP for Extension Monitoring Call Pickup List URI is not added automatically in the SUBSCRIBE packet as it was in the past.
– dkey_fkey doesn’t work for URL action
– P-Asserted-Identify information’s are not being removed from the contact.htm page
– RegEx dial plan broken
– UaCSTA doesn’t report redirection when PBX only uses ASSERTED-header to inform the phone
– Phone does not reboot when a UaCSTA session is the only active call
– LDAP search in 8.7.x appears slow in comparison with 8.4.x
– LDAP search with multiple entries shows wrong number on display
– LDAP number attributes are not applicable for ldap_display_name
– SnomIPPhoneDirectory: the last entry item is highlighted instead of the first one
– XML Idle Screen incompletely or incorrectly loaded on reboot
– XML idle screen flickers every second after a conversation (snom 870 only)
– No redial function after a call without hanging up phone
– Scroll bar does not highlight the selected ring tone in phone menu
– Image/Icon for connected call is missing when in a call (snom 760 only)
– : Edit number before call feature is broken
– Blind (Unattended) Transfer isn’t working
– Really slow responsiveness of the phone after having done several calls
– Group Call Pickup via Call Monitor does not work
– Phone restarts after dialling a number from missed call list accessed from Status Information
– Resolving Arabic font issue
– Local address book matching does not work any longer for direct IP calls
– Changes to user_ringer via phone menu isn’t saved
– Conference soft key should not auto hide when conferencing setting is not empty
– goto_monitor_state_on_line_activity not working as expected in ring group scenario
– Several actions are always applied to connected call instead of waiting call even if waiting call is selected in joined call screen
– Call Pickup: Wrong LED Behaviour (Ringing, BLF Key instead Line Key)
– Can’t activate call forwarding on 2nd identity via phone menu
– Status Info screen will not be displayed if key is overwriten with dkey_fkey4
– user mode not applied after change from admin to user and cold restart of the phone
– Pressing the speaker key (short press) changes mode from Headset to None every two times, must do it every time
– Volume state does not time out
– Parking call blinks up too fast to read on snom 7xx
– internal directory entry lookup fails when call comes in to identity which is not set as outgoing and partial lookup is enabled
– phone shows the time instead of the call duration during a call (snom 710 only)
– editing internal directory entries via phone menu not possible
– Editing destination SIP URI for push-to-talk in phone menu results in some random number
– Cannot access Time Zone, Tone Scheme and Language when logon wizard is active
– Cannot scroll up in Settings and after a while the phone restarts (snom Meetingpoint only)
– can’t dial a number of a call list entry while the detail view is shown
– Settings value for language and web_language inconsistent (Nederlands/Dutch)
– Sending a long message to the phone display results in chopped message
New Features:
– Default value for setting always_show_active_call changed to on since this the expected behaviour in most scenarios
– 870 UI Changes according Spec for all Non-Idle Screens
– Phone supports partial lookups using the last n digits of a number for incoming and outgoing calls
– Transfer successful message not shown by default any longer. The behaviour can be changed via new setting seconds_to_show_transfer_success_for
– Desteği Line Flash DTMF event. New codec names available in codec_priority_list:
– Phone models snom 3xx, 820 and 710 are now supporting codec strings g729-annexb=yes and g729-no-fmtp in codec_priority_list
– Rreintroduce old behaviour from 8.4.x: although record_missed_calls is disabled action_missed_url has to be fired
– Incorrect REFER causes server based conference to fail with 3CX
– Introduced new setting: user_replaces_when_referring_to_conference_server
– G729 codec choice incorrectly includes annexb=no in SDP New codec name g729-annexb=yes available in codec_priority list Under -> Setup -> Identity -> RTP ->Codec
– Moved setting publish_presence from Advanced/SIP/RTP to Identity/SIP
– Provisioning sometimes does not work when the network uses STP and static IPs for the phones.
– Status message now shows ETA of max provisioning duration (calculated on the duration of the first attempt) respectively.
– WLAN – Desteği 802.11N
– Certificate Based Network Authentication (802.1x)
– Implement handling of different headset types
– New setting dtmf_handset_phone to disable DTMF tones during call initiation in handset mode
– Implementation of dtmf_volume setting for all phone types
– Push2talk now also works when picking up the handset first and then pressing the assigned push2talk function key
– UaCSTA doesn’t report when sip-TO-header differs from the local identity for an incoming call.
– Using server side call logs should be configurable
– End all calls tied to an identity on deactivation
– Hot Desking – Identities have a new setting user_no_auto_logoff which defines that the identity is not deleted during Logoff all. This can be used e.g. for emergency lines.
– When using server side call logs we should fetch the logs once upon (re-)boot to ensure the icons on 870 are properly shown.
– Broadsoft: Implementation of the Call Log and Address Book components of the user interface
– Broadsoft: Implementation of the XSI Protocol stack
– Broadsoft: XMPP Desteği Buddylists and presence
– Broadsoft/BLF/Call Park Indicator/Call Park Pickup in Buddy List XML Definition
– XML Buddy key now also signals a parked buddy and allows to retrieve it
İndirme