Snom 370 IP Phone Firmware 8.7.4.8 Beta

Fixes:

- MWI Indication beeps strange in idle
- G729ab not working on 3xx, 820, 710
- Directory Ring tones not working as expected
- 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 get 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 correct
- Phone stops receiving multicast after receiving an incoming call via handset
- Multicast paging receive delays audio around 3 seconds when changing audio device
- Dial tone being played and transfer message blocking screen for some seconds after a successful blind transfer
- Wireless headset doesn't work properly, audio is played via handset instead of headset and call is disconnected by pressing the headset button twice
- Incoming Call while automatic redial on busy results in ringing tone restart every second
- Pickup_indication not working any longer
- Broadsoft: (Call Center Hold Reminder) Speaker is not ringer device for second incoming call
- Resolved Arabic font issues
- 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
- UaCSTA doesn't report redirection when PBX only uses ASSERTED-header to inform the phone
- Presence stops working after the SIP proxy is down
- Extension monitoring on expansion module causing delays and the phone to become unresponsive or slow
- Hold fails because of different DTMF paylod types 101 <-> 96
- 302 Moved Temporarily during call
- Park Service is shown as connected number for forked calls with the same number
- Wrong destination port is used for voice mail if the registrar is not using the default port 5060
- PUBLISH contains a body in refresh requests
- g729 codec choice incorrectly includes annexb=no in SDP
- If registrar is set to use a different port (e.g user_host = 192.168.1.1:6050), the port is used for registration, but not for INVITEs
- Phone loses registration on Astimax
- Phone does not reboot when a uaCSTA session is the only active call
- Phone does not register due to ignored expires header on OpenSips
- Display freezes after provisioning failed (snom 320 only)
- Snom 870 XML idle screen flickers every second after a conversation
- Goto_monitor_state_on_line_activity not working as expected in ring group scenario
- Unable to initiate attended transfer if multiple identities are involved
- P-Asserted-Identity number part isn't recorded in call history
- Dialling number out of call list entry detail view not possible any longer
- Selected ring tone not highlighted in phone menu
- User mode not applied after change from admin mode to user mode and cold restart of the phone
- Status Info screen not being displayed if key is overwritten with dkey_fkey4
- Editing internal directory entries via phone menu not possible
- Unknown instead of callerID being displayed on incoming call after hold
- Phone reboots on Logoff User via phone menu
- Edit number before call feature is broken
- Really slow responsiveness of the phone after having done several calls
- Cannot open details of missed call in Status Information by pressing OK key
- Can't activate call forwarding on 2nd identity via Menu -> Call Features -> Call Forwarding
- Cannot scroll in phone menu and after a while the phone restarts (snom Meeting Point only)
- Phone restarts after dialling a number out of missed call list accessed from Status Information
- Group Call Pickup via Call Monitor not working
- Cannot access Time Zone, Tone Scheme and Language when logon wizard is active
- LDAP number attributes are not applicable for ldap_display_name
- 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 in WUI
- IP for Extension Monitoring Call Pickup List URI is not added automatically in the SUBSCRIBE packet
- P-Asserted-Identify information are not being removed from contact.htm
- Using attribute complete="true" in tbook provisioning leads to changing some settings like languages, date format
- TR-069 GetParameterValues RPC handling is broken for generic parameters
- dkey_fkey doesn't work for URL action
- SnomIPPhoneDirectory: the last entry item is highlighted instead of the first one
- Replacement of {index} in action URLs not working any longer

New Features:

- Implementation of dtmf_volume setting for all phone types
- New setting dtmf_handset_phone to disable DTMF tones during call initiation in handset mode
- Support for Line Flash DTMF event
- End all calls tied to an identity on deactivation
- UaCSTA doesn't report when sip-TO-header differs from the local identity for an incoming call
- Phone supports partial lookups using the last n digits of a number for incoming and outgoing calls
- 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
- 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: telephone-event-no-fmtp: no fmtp attribute is offered in SDP telephone-event-0-16: fmtp 0-16 is offered in SDP, which enables the use of DTMF line flash
- Phone models snom 3xx, 820 and 710 are now supporting codec strings g729-annexb=yes and g729-no-fmtp in codec_priority_list
- Moved setting publish_presence from Advanced/SIP/RTP to Identity/SIP