Firmware/V7/Release Notes/Change Log V7.3

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Change Log between 7.3 and 7.1 branch

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no features.

  • Added support for version 6 audio coefficients (snom3xx). Selecting the new room type "v6" leads to same volume behaviour as in version 6 releaeses. (SCPP-889)
  • Added support for options 132 vlan_id (802.1q) and 133 vlan_qos (802.1p). They can even be tunneled via option 43. (SCPP-859)
  • Added additional STUN requests (RTPv0) in the beginning of a call for server type Teles. (SCPP-671)

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  • no features
  • VQM reporting is no longer restricted to MetaSwitch server type anymore.
  • Added Hungarian language support.
  • Securing communication with random message id. New CSTA messages for set and get settings, xml push and firewall test. (SCPP-648)
  • Enbabled EHS support (snom300). (SCPP-612)
  • On challenged web client requests, user gets asked for username and password on the display. (SCPP-699)
  • Phone can now upload files into tmpfs and replay them as MOH. (SCPP-696)
  • Parameters like "$ip_addr" as part of the URL stored in for example setting_server are replaced with it's values before the URL is used for provisioning requests. This is the same syntax like for the Action URLs. (SCPP-698)
  • Enhanced language support for Portugues and Hebrew.
  • Added language support for Estonian and Hungarian.
  • Added VQM reporting over SIP, according to draft-ietf-sipping-rtcp-summary using SIP PUBLISH (new setting vq_report_collector).
  • If on Call Pick-up the first character in the string following the "|" is a "!" the rest is dialed out unaltered.
  • Added new setting prioritise_pbx_number_lookup to ignore PBX transmitted user name for incoming calls in order to enable correct lookup in internal LDAP address book. (SCPP-131)
  • Added support for "check-sync;reboot=reregister-only".
  • Added support for RTP listening on multiple multicast sockets (new settings mc_addressX).
  • Added key_dtmf to initiate key events with dtmf tones remotely via http://phoneIP/command.htm?key_dtmf=KEYEVENT. (SCPP-94)
  • PC port can be disabled via new eth_pc option off.
  • Added new action release to clear previously seized button line.
  • New setting user_additional_supported_header for "additional supported headers". (SCPP-413)
  • Handle header "Remote-Party-ID" for server type Cisco Call Manager (CCM). (SCPP-412)
  • Added Generic Intercom Button feature. If programmable Function Key is set to Intercom without a target number, user gets asked for the target before Intercom call starts.
  • Added advertisement flash at the index.htm page (new settings advertisement and advertisement_url).
  • Added support for IPv6 DNS server address query.
  • Iso codes (ISO_639-1) are used to name different language files.
  • Provisioning retry counter can be canceled by pressing cancel key
  • If entered incorrectly three times and/or ongoing wrong http password access will be restricted for one minute blocks only.
  • no new features
  • added timer driven settings refresh from server (new setting settings_refresh_timer)
  • enhanced Russian translation
  • do not show empty Calls screen for 3 seconds anymore
  • setting pickup_indication is available again for big display phones (snom360/snom370)
  • changed default redirect edit mode from alphabetic to numeric (SCPP-120)
  • a logging entry is created if an Action URL could not be accessed (SCPP-146)
  • split USA-10 timezone into 2 timzones to distinguish between hawaii and aleutian islands (new timezone is USA2-10 which has DST in contrast to USA-10 which has not)
  • added unused option to the drop down menu of the timezone setting at the advanced network WEB page (So you can set timezone to empty and read only and it's not showing a specific timezone but unused.)
  • new SIP server type Asterisk
  • compliance to RFC 3863 - Presence Information Data Format (PIDF)
  • set IP address of phone to for Nortel and Sylantro server type on hold_state
  • added client certificate presentation (webserver_cert also used as client certificate)
  • no new features
  • no new features
  • show number of provisioning attempts on the display
  • added Hebrew national language support for (snom370) and (snom320) as well
  • character order of Hebrew XML language files has changed (this means older hebrew texts for the Phone User Interface are not compatible to this version and vice versa, you will experience wrongly ordered Hebrew characters)
  • added additional functionality for fkey type key event; F_TRANSFER:700 would transfer the connected call to destination i.e. 700 (not the incoming call)
  • suppress individual dialog subscriptions when eventlist is on (Broadsoft)
  • change reading direction according to language name (Hebrew reads right to left)
  • supports now Hebrew and special chars like the lock symbol in one image (snom300 with PoE); this means Hebrew support is now part of the usual release and doesn't need an extra image to get displayed properly on (snom300) backlight displays
  • improved audio feedback with auto dial on busy signal
  • duration display for a connected call on (snom370) is hh:mm:ss
  • show IVR dtmf on display during connected state (snom370)
  • consistent behavior for incoming calls in menu states
  • new setting show ivr digits to control digit display while connected (snom360/ 370)
  • unicode characters are displayed properly in title line and soft key texts on (snom360)
  • added settings subscription delay, terminate subscribers on reboot and functionality (from v6.5)
  • show uboot version, uptime and memfree on system info page
  • support for $display local, $display remote, $expansion module, $active key in Action URLs (also within path)
  • print out uptime and memfree to the (sys)log once a day
  • show uboot version on OID as well
  • new OIDs and for memfree and uptime respectively
  • language files can be loaded via TFTP as well
  • modified RTCP so that it also sends SDES CNAME with the RTCP SR
  • show HTTP/Admin password warning on display dependent on ignore security warning
  • LDAP/Directory Search list dials selected contact on handset offhook
  • soft keys for Mute/Unmute are not hidden when redundant fkeys setting is off
  • F LOGON USER can be used on all phones for hot desking (log on/log off user accounts)
  • F LOGOFF ALL prompts log on wizard if logon wizard setting is on
  • Disabled 5 sec. lock on shared line key for Broadsoft/MetaSwitchas this is handled by line seize throttling mechanism
  • block programmable keys while Broadsoft-style shared line is seized
  • 200 OK without SDP is ok when SDP was received in earlier message
  • help link of setting user q corrected
  • fixed timezone selection for China
  • rework of SnomIPPhoneText, better line breaking and <br/> tag support
  • added new {index} parameter for SoftKeyItem (The index parameter is the linenumber in a minibrowser list. e.g.{index})
  • new setting offhook dial prompt controls if dial prompt is offered on Offhook and it's on by default (You may not want to make the phone busy by just taking the handset offhook. If this setting is turned off, the phone will continue to show the standard idle display even when the handset is offhook.)
  • post Action URL disconnected on Call completed elsewhere. Technically this call has been received by someone else, however to complete the transition from ringing to call end, Action URL disconnected is called to let the server know.
  • new programmable key type IVR. (The argument is a number that is dialed on key press i.e. sending out an INVITE. Once the call has been established, pressing the same IVR key would send out dtmf digits comprising that number. This can be used to control IVR applications by one key only.)
  • hebrew character support
  • user customised Alert-Info texts/files using built-in melodies (If the value of setting internal ringer text matches the incoming INVITE alert-info header, use the internal melody set up in internal_ringer_file to play the appropriate internal melody for a customized alertinfo.)
  • basic CSTA call control features (phone can be remotely controlled via CSTA protocol)
  • support for more Action URL events
  • added setting show diversion to control display of Diversion header info included in the incoming INVITE
  • new makro prov_host which can be included into provisioning URLs (like mac) in order to get replaced by the setting server host name which has been set differently before. (This makes the usage of provisioning URLs much more dynamic.)
  • added a list for tftp requests to make sure only one request runs at a time
  • firmware status file can be loaded via TFTP as well
  • firmware can be updated via TFTP as well
  • XML language files can be retrieved via TFTP as well
  • firmware update mechanism optimized to use less main memory during update process
  • added bootloader lock mechanism (together with the appropriate bootloader the phone can lock the tftp update option in the bootloader to establish for instance an ITSP lock)
  • new user_server_type nortel
  • Added new setting transfer_on_hangup_with_starcode to allow transfering calls on hangup if one call was picked up with PBX starcode. (SCPP-906)
  • Clean paint area for title (snom370). (SCPP-1191)
  • Codec type and TX-RX stats could not be displayed anymore during call at the display. (SCPP-1206)
  • Don't handle key event F_LOGOFF_ALL if a line is connected or on hold. (SCPP-1203)
  • Don't resend ring command if phone is already ringing even if there is more than one incomming call. (SCPP-972)
  • Handle action of FKeys only if state isn't minibrowser state. (SCPP-1217)
  • Setting Use Speaker Key to Receive Calls is valid for headsetkey too now. (SCPP-1154)
  • Show LDAP result instead of display name if setting prioritise_pbx_number_lookup is off. (SCPP-1215)
  • Send VQ-Report for hanging up on a holding line as well. (SCPP-1213)
  • Make reregister after check-sync if necessary. (SCPP-985)
  • UnREGISTER sets unregistered identity to inactive. (SCPP-599)
  • Termination of Registration/Subscription initiated by check-sync notify works now. (SCPP-1126)
  • Some setting changes were effecting other settings even though they were not changed by the user. (SCPP-1123)
  • Switch to speaker after onhook in the states edit number and connected. (SCPP-988)
  • Accept transfer key in state transfer. (SCPP-798)
  • Handle the programmable keys in state settings as well. (SCPP-293)
  • Long numbers fit now into display in editing states (snom360). (SCPP-843)
  • Switch to state edit number if we got a 484 INCOMPLETE during overlap dialing. (SCPP-950)
  • If the provisioning location was specified without scheme, skip the generated HTTP urls if one of the phone generated TFTP urls were successful. (SCPP-1008)
  • Write settings again to flash filesystem when change was made via web interface.
  • Call was sometimes not released properly. (SCPP-1207)
  • Fixed headset from ringing only once per incoming call to now every 3 seconds. (SCPP-416)
  • Add uri in calling lists even if registrar contains parameter. (SCPP-1039)
  • Close "call stats" dialog if a call is incoming. (SCPP-1105)
  • Configured function keys work now in every state, except type is set to Key Event. (SCPP-931)
  • Custom ring melodies above 300kb will be ignored to avoid crashes. (SCPP-891)
  • Fixed cursor shape (snom360). (SCPP-844)
  • Going offhook with wireless headset can dial numbers out of internal or LDAP directory. (SCPP-1072)
  • Keep alphabetical order when provisioned directory entry overwrites existing entry. (SCPP-1174)
  • MWI changes to normal dialtone independent of GUI state. (SCPP-510)
  • Playing "Internal Ringer" set by preferences webpage if alert-info is set. (SCPP-1058)
  • Playing ringer of current line via webinterface. (SCPP-1019)
  • Removed dead softkeys for incoming call during conference. (SCPP-847)
  • Show cursor in LDAP only during editing search string. (SCPP-670)
  • Not accepting content anymore that is so huge that it could crash the phone. (SCPP-891)
  • Limit writing settings to flash filesystem to a minimum. Just write after provisioning and right before reboot in order to get the phone freezing cured (new setting apply_settings_timeout can enable the old behavior again by setting it to 20; it is set to zero by default).
  • Fixed missing Route Header field on resubscribes. (SCPP-638)
  • SDP media connection IP overwrites SDP session media IP. (SCPP-1007)
  • TFTP client supports now subdirectories. (SCPP-477)
  • Don't dial last LDAP result if number has changed and no LDAP result was found (snom300, snom320). (SCPP-1120)
  • Answering a call was failing and log said: "Failed to allocate any local SIP port!". (SCPP-1042)
  • Added missing runtime variable $phone_ip to be used with the webclient like triggering action URLs etc.
  • Fixed no ringing and no dial tone problem after opening a new call while holding the old one with the function key (in non G.711 codec call).
  • Fixed phone ringer silence on subsequent calls after using a non G.711 codec in a call (snom3xx).
  • Added more flexible way of configuring IPv6 gateway.
  • Call Pickup did not work with multiple registrations. (SCPP-991)
  • Added standard Italian dialtone. (SCPP-697)
  • General performance improvements and memory usage optimizations which fixed freezing behaviour as well. (SCPP-669, SCPP-838, SCPP-905)
  • Don't set mute after deflection to off. If deflection feature is disabled transfer key does nothing. (SCPP-825)
  • No delay after key press in minibrowser. (SCPP-794)
  • Fixed transfer issue with no destination number. (SCPP-849)
  • Text wrapping fixed (snom360, snom370). (SCPP-856)
  • Transfer on onhook now works with picked up calls as well. (SCPP-632)
  • Update display if Detect Ethernet Cable Unplug switched off. (SCPP-944)
  • Phone was misbehaving when some certain menu entries had been selected in user mode. (SCPP-953)
  • Store missed call in caller list even when DND is active. (SCPP-938)
  • "Enhanced" data returned from LDAP were interfering with dial plan. (SCPP-842)
  • Display method setting will be applied and not only stored when coming in with a check-sync. (SCPP-619)
  • Only 4 identities are valid for snom300. Handle even old setting file with 12 identities without hanging. (SCPP-772)
  • The "!" setting flag wasn't working if user was changing a setting value back to default. (SCPP-890)
  • Preference field in NAPTR record was misinterpreted as a weighting factor. (SCPP-945)
  • Stale nonce indication was not handled. Now phone evaluates the "stale" parameter in the WWW-Authenticate header. If it is set to true, phone will retry with the same credentials. (SCPP-930)
  • Contact address was not refreshed on re-SUBSCRIBE. (SCPP-923)
  • Fixed dirty host cache entries invoked by incoming ICMP packets (SCPP-895) and by UDP request timeout (SCPP-899). A phone using Proxy Failover due to dns naptr and with dirty host ttl configured adds now the failed server to the dirty host cache.
  • Fixed subscription if subscription target got new contact due to reboot. (SCPP-846)
  • Fixed a webclient issue ending up with code 500.
  • Show logout button on webUI even if setting ignore_security_warning is off.
  • Fixed G.729 distorted audio and one way audio problems (snom3x0).
  • Don't scroll to next identity if an other than the current identity was removed. (SCPP-816)
  • Enable changing audio device from speaker to headset in holding state as well (snom300). (SCPP-674)
  • Clear area of status line for XML idle screen at given position else at default position (snom360, snom370). (SCPP-815)
  • Set edit mode to numeric for editing numbers if no text to be edited is already set. (SCPP-746)
  • Don't save passwords in exported settings files. Accept empty admin password after reading setting file. (SCPP-823)
  • Could not take call off hold when Shared Line is active identity. (SCPP-445)
  • Function "Play ringer remotely" was playing the ringer of identity X, but not the ringer X. (SCPP-811)
  • Stale 401 should be ignored. (SCPP-708)
  • Scroll to next identity if current identity was removed or show logon wizard if all identities were removed. (SCPP-621)
  • Aborting transfer if phone goes onhook. Send Re-INVITE and switch audio to speaker. (SCPP-667)
  • Call deflection was not working when destination was taken from a list such as dialed number or contacts. (SCPP-797)
  • Changed fkey clear to fkey delete in details of call lists. (SCPP-807)
  • 400 class response to a REFER was not handled. (SCPP-792)
  • Fixed G.726 bad audio quality. Added the RFC3551 byte packing order for G.726 codec (new line setting user_g726_packing_order). (SCPP-526)
  • PBX transmitted user name was sometimes displayed instead of the internal addressbook name if prioritise_pbx_number_lookup was switched off (SCPP-131). Revised the whole ticket again.
  • Menu items were incorrectly mapped in user mode (snom360, snom370). (SCPP-768)
  • If a call out of holding state is terminated go to terminate state first (e.g. play signal). (SCPP-757)
  • Phone did not always send out a SIP line seize message when function keys are configured as buttons. (SCPP-723, SCPP-756)
  • Fixed on/off switching of setting flood_tracing. (SCPP-779)
  • Voice Quality Report included wrong start time value. (SCPP-760)
  • Accept KEY=TRANSFER also on snom300. (SCPP-764)
  • Call duration longer than an hour will be displayed with an h at it's end. (SCPP-767)
  • If the number of missed calls was greater than 9 the number was displayed backwards. (SCPP-740)
  • Call from VIP did not ring if DND was switched on. (SCPP-709)
  • Use big font size in edit mode if Hebrew character mode is switched off (we do not support big Hebrew characters yet). (SCPP-738)
  • When one pressed the volume keys and used the soft keys to change the volume and you return to the idle screen, calls were canceled. (SCPP-737)
  • Fixed a couple of Hebrew display bugs.
  • Fixed issues which occured in edit mode.
  • Changed default character modes in the address book and in edit mode.
  • VQM stats enabled again, they were temporarily removed before.
  • DTMF duration fixed for SIP-INFO. (SCPP-609)
  • Audio device icon fixed in holding state (snom370). (SCPP-508)
  • Extension key inoperative on shared line when phone is off-hook. (SCPP-628)
  • GRUU contact header was enclosed in quotes. (SCPP-451)
  • Call pickup did not work correctly if display name was sent by the server.
  • Phone was crashing if SIP (OPTIONS) messages were sent to it during boot up (snom820). (SCPP-702)
  • Sip UDP src port and sip port encoded in sip message is now equal (snom370). (SCPP-484)
  • Multicast paging was not working reliably (snom3x0). (SCPP-226)
  • Requeueing of failed provisioning URL was shortening the URL by one character on each attempt.
  • Re-INVITE from MOH server causes MOH stream to be redirected to the holding party. (SCPP-653)
  • Removed sending of spurious 482 message.
  • Fixed phone crash which occured after deleting one of the two call lists for missed or received calls (detailview). (SCPP-610)
  • Fixed wrapping of continuous text in minibrowser. (SCPP-551)
  • Added Estonian language support.
  • Refresh GUI after onhook in state holding to change audio icon. (SCPP-509)
  • Transfer on onhook will only connect one incoming and one outgoing call. (SCPP-632)
  • Always use a non-secure RTP connection for MOH. (SCPP-471)
  • URI parameters were being stripped from contact in processing of a 302.
  • Phone extracts now encapsulated options 66/67 out of option 43 as well. (RFC 2132)
  • Initialization of extension keypad fixed. (SCPP-614)
  • Don't accept IP addresses with leading zeros. (SCPP-534)
  • Re-)start autodial timer if enabled only after key events and not on entering LDAP search. (SCPP-584)
  • On LDAP search split a matching line into several lines if a member has more than one phone number. (SCPP-538)
  • LDAP lookup for incoming calls displays only exact matching number. (SCPP-559)
  • Fixed displaying utf8 search results (snom320). (SCPP-556)
  • The reregister function was causing the phone to sometimes loose its registration.
  • Phones are no longer using contact as refer-to header. (SCPP-523)
  • URI scheme was not recognized when it is in caps.
  • "check-sync;reboot=false" sometimes wasn't triggering settings reload from settings server. (SCPP-515)
  • Disabled overload option, wasn't working anyway, but confusing provisioning.
  • no bugfix
  • Fixed disabled ringer animation screen. (SCPP-437)
  • Consistent behaviour locking the keyboard. (SCPP-470)
  • Fixed showing guessed number (snom370). (SCPP-430)
  • Missed calls LED handling logic fixed. (SCPP-379)
  • Send reinvite after aborted transfer as well as the line is "on hold". (SCPP-467)
  • Fixed call pickup if intercom is enabled. (SCPP-495)
  • Randomize request source port and DNS query id. (SCPP-479)
  • Fix Speaker Volume Adjustment in Headset/Handset and Speaker Mode. (SCPP-480)
  • A picture from CALL-INFO header in a provisonal 18x was never shown (snom370).
  • Transport parameter was dropped on redirected register.
  • Initialization was broken.
  • By default answer_after_policy is set to "off". (SCPP-506)
  • Flash plugin didn't work with current Flash player releases due to new Flash security requirements. (SCPP-43)
  • No entry was added in the missed call history if call waiting was disabled. (SCPP-450)
  • If one was to start dialing a number the DTMF of the first key was heard albeit dtmf_speaker_phone is switched off. (SCPP-424)
  • Fixed timezone settings for Vladivostok and Tonga. (SCPP-420)
  • Reset edit mode to numeric in transfer state. (SCPP-419)
  • Set all monitored LEDs to on after transfer. (SCPP-391)
  • Fixed one way audio with G.726-32 and Payloadtype 2. (SCPP-446)
  • Don't process SUBSCRIBE packet if contact header is missing (could happen by an Asterisk bug). (SCPP-431)
  • Fixed rejection of anonymous calls. (SCPP-418)
  • Wrong user image has been shown if a second call comes in (snom370). (SCPP-321)
  • LDAP lookup for incoming calls displayed wrong name when phone number contains same digits. (SCPP-70)
  • Removing an identity (that requires authentication) caused corrupt password prompt. (SCPP-283)
  • Fixed a couple of backlight issues (snom360, snom370). (SCPP-267)
  • Pickup worked via fkey but not via tick key or softkey. (SCPP-306)
  • Resolved sticky volume state. (SCPP-262)
  • Holding reminder did not work. (SCPP-319)
  • $remote parameter in action URL was indicating wrong number. (SCPP-349)
  • Fixed handling of deny-all for states ringing, connected and holding (SCPP-343).
  • Removed Identity settings when in user-mode. (SCPP-338)
  • Value "dim" for backlight on snom370 phone gui added. (SCPP-337)
  • Redial will use current identity if original id is unregistered except if current id is unregistered too.
  • Play current user ringer during changing ringer volume. (SCPP-133)
  • Added type of answering for Intercom (new setting intercom_connect_type in contrast to auto_connect_type). You may select now the audio device you want incoming intercom calls to use. (SCPP-116)
  • Clear area of status line before repaint (snom360, snom370). (SCPP-388)
  • Clearing call list was crashing phone. (SCPP-370)
  • Corrected wrong soft key assignment when a call is being hold in hands free mode. (SCPP-58, SCPP-310)
  • Call waiting was disabled during transfer. (SCPP-42)
  • On holding state no action performed by speaker off if speaker mode is active because no hang up is allowed. Dialing a new number can simply be started by typing numbers. (SCPP-60)
  • The assigned action for menu key is handled in connected state too (snom320). (SCPP-318)
  • Clean area of status line before repaint. (SCPP-388)
  • Using Flexor Manager to select the outbound line to use, did not change the outgoing identity on the phone, if multiple identities were configured. (SCPP-365)
  • Call did not disconnect after on-hook when a minibrowser file is being displayed and an on_hook action url is set. (SCPP-8)
  • If entered incorrectly three times and/or ongoing wrong http password access will be restricted for one minute blocks only.
  • Added wiki help link to Fkeypage as well.
  • Fixed CSV-import for directory. (SCPP-384)
  • Call transfer was not working with Flexor Manager. (SCPP-385)
  • IP's are now parsed and corrected before saving, because numbers like were causing problems (IPv4 only). (SCPP-325)
  • Fixed a bug in XML decoding. (SCPP-285)
  • STUN derived port number was not reflected in SDP (SCPP-108)
  • Display name decoding must be done if remote display name is encoded via setting encode_display_name. (SCPP-73)
  • Fixed missing of WWW-Contact on Registration if do_http_contact says so. (SCPP-74)
  • Refresh publish presence if time expires and publish the last state on bootup. (SCPP-256)
  • If the phone received a "422 Session Expires Too Small" error in response to the first INVITE with the Session-Timer, hangup on the subsequent INVITE resulted in the phone sending a BYE instead of a CANCEL. (SCPP-339)
  • Old To and From Tags were not cleared if subscription was restarted. (SCPP-360)
  • Properly refresh URI in SUBSCRIBE message (SCPP-316).
  • Fixed corrupt SDP full answer after SDP offer with unsupported RTP payload types.
  • Contact header of some requests were not properly replaced with valid STUN identity.
  • New setting ip_frag_enable to control the DF-BIT in IP packets.
  • Initial rtp-keep-alive was missing, which resulted in skipping of first seconds of MOH; if user_server_type is set to metaswitch do not send it. (#2008082810000098)
  • Fixed encryption of DTMF payload.
  • Polish special characters are displayed properly on new backlight display and at least as normal chars on old display. (snom300)
  • Delayed trap handling to fix issue which caused the phone to hang on bootup.
  • Fixed X-key to cancel a minibrowser slideshow. (SCPP-266)
  • Parsing and converting "&#…;" codes properly now. (SCPP-395)
  • Pressing keys in minibrowser during calls will not generate dtmf echo.
  • ASN.1 encoded length fixed. (SCPP-363)
  • fixed redirection via FKEY type BLF
  • fixed bugs regarding displaying numbers with multiple digits together with hebrew texts (BZQ-4)
  • phone was rebooting with DHCP lease times bigger than 16 hours
  • network for some phones was not working with 7.3 versions, fixed
  • initial RTP-keep-alive packet was missing, is send out now except MetaSwitch is chosen as SIP server type (SCCP-XXX)
  • network for some phones was not working with 7.3 versions, fixed (SCCP-134)
  • fixed issue which causes the phone to hang on bootup (SCCP-14)
  • setting utc_offset wasn't used if it and dst was set without using timezone
  • hold on conference was not working (snom370) (SCPP-59)
  • BLF key wasn't working in off-hook state (SCPP-33)
  • incoming call was not rejected when the phone was in a conference and call waiting was turned off (snom370) (SCPP-150)
  • fixed first line in idle state regarding Hebrew character order for snom320
  • fixed idle screen regarding Hebrew character order in call forward hint (snom360/snom370)
  • fixed conference in speaker phone mode
  • setting subscription_delay is working now
  • multiple DHCP OFFERs from different DHCP servers were confusing the phone, now it sticks to the first OFFER
  • fixed conference in speaker phone mode
  • CCNR was broken (SCPP-XXX)
  • in some instances, the wrong line key was selected on incoming calls (SCPP-XXX)
  • do not invoke dial plan when generating URI for call-info subscription (SCPP-XXX)
  • WWW-Contact was no longer being generated (SCPP-XXX)
  • TFTP client was not resending ACK when a previously received data packet is received again (SCPP-XXX)
  • programmable keys 6 and up were blinking at a lower frequency on incoming calls
  • last held call was not presented first during transfer after implicit hold
  • extension key was not working with Broadsoft/MetaSwitch style shared line
  • channel was not being cleared completely after transfer while ringing
  • if uboot version string is corrupt, don't store it anymore (this may fix the issues provisoning server had with the added corrupt uboot version in the HTTP client header)
  • it retries the provisioning request several times on internal error codes 500 (ICMP error tftp / read error http), 400 (DNS timeout tftp/http) and 408 (tftp error), this may fix the well known STP issue on startup provisioning wise
  • No bugfixes
  • missing password warning was shown on display even if setting ignore security warning was enabled (snom370), fixed
  • redirection * codes were not working
  • setting http scheme doesn't need restart to get applied
  • default value of setting http scheme was wrong, it's off now
  • on error code 500 (which may mean ethernet port isn't open, most probably due to a switch which blocks it temporarily) retry the request several times
  • Park was not working with Broadsoft-style shared lines
  • don't invoke auto-redial when line seizure times out
  • Audio was not switching to handset when it was lifted after a line key had been pressed
  • No bugfixes
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