Snom m3/Firmware/Release Notes

• Phonebook loader is made for pushing phonebook (in *.CSV file format) to FP (common) or handset (private - HS1 to HS8)
• Reason header cause=200 ;text="Call completed elsewhere” handling added regarding missed call indication.
• If phonebook is local (Not common) the handsets which is ringing will not show missed call if an incoming call is answered by one of the ringing handsets.
• Call log handling for Common phone book is made Call Group dependent to avoid showing Missed Call on handsets which should not have this information.
• Phonebook directory load feature for Common phone book.
• // hide LOCAL_HTTP_SERVER_AUTH_PASS password and comment //
• DHCP HostName and ClientId added
• Phonebook loader (.CSV) is made.
• DTS server updated to support -2.30 and -4.30
• DHCP changed to clear the BootP broadcast bit, which have shown being unsupported by many DHCP servers.
• FWU updated to check handset firmware version too, to ensure all handsets are updated to the same version.
• Handset name now allow symbolic characters.
• Registration is now re-trying after receiving reason 404 (Not Found). Registration attempt will still stop if reason 403 (Forbidden) is received as response.
• Web sites Reboot/Debug/Settings/SipTrace/ is not protected by authorization.
• IP address showed at home page if static IP is chosen, is corrected.
• Negotiated IP/SubMask etc is now shown in the Advanced page
• Web site password masked with ****
• DNS SRV handling of additional records (A-Record) for the server has been added.

• Typo “Registrating” replaced with “Registering” at front page status
• Processing Update.html page twice corrected to process only once.
• Some sending of wrong User-Agent corrected
• DTMF inband bug fixed
• DTMF (rfc2833) and (SipInfo) is changed to use 160 ms as minimum duration (Inspired from Snom 320)
• Consulted transfer corrected to wait for acknowledge of active call set on Hold before the call transfer is initiated.
• Bug regarding not audio if call is transfer from peer site before codec is connected in gateway. Fx. if a called user has diverted the call to voice mail, this diversion could cause this bug if the diversion signaling was very fast.
• Wrong showing of “Voip Settings PIN” corrected
• LOCAL_HTTP_SERVER_ACCESS bitmask check corrected. Bug regarding being able to see

Debug/SipTrace/Settings is bit 15 was cleared, has been corrected. If Bitmask is set to 0x0000 the access to the web site is fully restricted.

• Settings page showing FWD_xxx corrected
• Handling URL without <> is not corrected. Seperator ‘;’ added
• Bug regarding system not answering after some days has been fixed. Could though be more bugs not solved here fx. regarding not sending keep alive packets.

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• Correction feature for GW ID added and trigged by defaulting the GW (Long key press reset) if the GW has a wrong ID
• Settings site on the Web interface added

• Transfer of two calls to same GW using STUN corrected. GW1->GW2,HS1; GW1->GW2,HS2; GW1 transfers the two call to GW2 together.
• ‘@’ is now escaped in the “Replaces” header
• Typo in web site regarding “Firmware” corrected
• Handset reset feature corrected

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• Dynamic adaption of TFTP/HTTP protocol to use for getting configuration file; tftp:// and http:// is now analyzed and removed before DNS A-record request; Web interface do also allow new type specification.
• Mail trace feature for BF messages are now able to be route to Ethernet a TCP port or UART. Setting added in Web page “Advance settings” to configure TRACE_MODE. Default is Ethernet.
• FWU improved debug prints and error handling
• NOTIFY check-sync implemented to check for new configuration file. The Reboot option in the NOTIFY check-sync is not implemented yet.
• New Display driver as preparing for change in production.
• Default HS menu added in the HS settings menu
• Common phonebook handling of long response time changed to leave the menu and not just stay. This will prevent the user to get stuck in the Contact scrolling menu.
• Call time is showing actual connected time and not just time from initiating the call.

• Bug with STUN active regarding call rejection due to disabled CW and/or active DND feature has been fixed
• Bug regarding using the To-tag from the 100 trying respond to an INVITE has been corrected. This gave error for problem with ITSP server “sip.mobilizer.com.au”
• Defense against overflowing with NOTIFY or OPTIONS signals added. This is done to fix the asterisk reboot bug.
• Bug in RegManager trigged by using MatchIPEI feature is now fixed.

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• System log moved to Home page
• Handset name cleared when resetting the GW with long keypress on Reset button.

• Cosmetic text changed.
• SipLog page width reduced from 160 characters to 120 characters.
• Automatic update of DND/CW/HS name when upgrading to this version is made.
• Default value for Auto DST changed to EU time. Automatic change of these parameters is not made.

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• IP Address Conflict detection and handling added
• IP Address Changed detection and handling added
• Dynamic DHCP Vendor Class ID added (snom-m3-SIP/01.08//14-Mar-08 09:54)
• Separate Proxy port definition and handling added.
• CW disabling possibility added
• DND activation possibility added
• Join Active Call can now be disabled by means of joining. If disabled, no Active calls will be shown for HS request.
• Authentication for UPDATE and BYE supported
• Registration mode for GW automatic enabled 5 minutes after every startup if Match IPEI is used.
• SIP signaling TOS field added (SIP_SIP_TOS_PRIORITY)
• Broadcast address in Contact or Record Route rejected (Not Accepted Here)
• SIP debug log added
• System log added
• Swedish language added
• Slovenian language added

• Fixed problem with asterisk based systems and >> signs in the from field
• Some double logged Runtime error removed in the parsing of Configuration File Parameters.
• User name and URI names is unrestricted. All chars allowed. User need to know what he is allowed to write.
• First RTP packet now gets a “mark” flag.
• Heap Size lowered to give room for SIP Debug Trace
• PCMA and PCMU removed as mandatory codec types.
• Packet rate field removed from SDP
• CRC16 Configuration file calculation made prior for Check Sync implementation.
• Random seed planted fir SIP signaling use is now random
• SIP INFO (DTMF) and Register Recall changed
• Memory leak removed if DTMF Inband tone is selected.
• RPORT use corrected to unregister initial registration before registering using new IP information.
• Incompatible media type rejection changed to use “415 Unsupported Media Type”
• SDP Direction tag corrected in initial INVITE
• Correction of bug trigged by Asterisk Server with congestion (Especial signal sequence: ->INVITE; <-100 Trying; <-183 w/SDP; <-486 Busy Here)
• Register SeqNumber changed to Random number
• Debug log date calculation corrected
• Handset name is now not deleted when new registration is made.
• Default Handset name handling for Portuguese corrected
• Deregistration handling of all handsets simultaneously improved
• Check sync feature (FWU checked first then Configuration file) added
• Out of range Beep added for in call warning.
• Possible to dial * out as a single digit
• WMI now can show if more that 255 messages are on server
• Bug regarding display contrast setting fixed.

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• Option menu in handset in case attended transfer is removed until called party has answered the call.
• Error log and OS log cleared after FWU to clear obsolete log data

• Display Name encoding and decoding corrected to use UTF-8
• Web page handling of öäüÄÖÜß corrected so these characters can be saved.
• The software has passed an internal Tekvizion test on R14sp3
• Fixed problem with passing Australien type approval
• If the gateway was offered more that 12 codec’s in a SDP the invite would be rejected with a 406
• Pilot production units can now use the reset to factory feature
• Fixed problem with pbxnsip version 2.1.0-2098, SSRC was changed when the call was answered by remote party causing silence in RX
• Added indication of speakerphone active/inactive
• Added a Reject call soft key on incoming calls
• Play reorder tone when receiving 4xx code except "Busy Here"
• DTMF negotiations RFC2388 to INBAND if peer do not support RFC2388.
• Fix regarding handling 423 response to REGISTER (“min-Expires” header field)
• Increased robustness against torture signalling PROTOS
• Fixed problem with distinctive ringing (incl. group) when using local phonebook
• Fixed unusable back key in STUN menu
• Don't show transfer, swap or conference option before VoIP call is connected
• Improved sound performace on key clicks

• Missing domain in "From" header field when answering to an SIP INVITE