Snom 720 IP Phone Firmware 8.7.5.8 Beta

Fixes:

- 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
- Support for 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 - Support for 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 Support for 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