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Nomenclature works as following, for example: 23.01.02(-rc03)
Year of initial release: Usually introducing major improvements and new features
Sub-Version: Usually contains necessary major bugfixes and improvements
Minor-Version: Usually contains very minor bugfixes
Release-Candidate-Version: Development branch with new changes before release (Beta-Testing)
Firmware Version 20.06 or 21.03 or above need to be installed before installing the 23.01 firmware. |
reworked firmware installation mechanism, to make installation more robust (Read the article above please!)
enhanced TLS feature for provisioning (upload own certificates and key. troubleshooting)
improved bfdetect to restore device (as in recovery mode)
added option to set plain password on provisioning
improved provisioning / beroCloud logs
reordered and reworked presentation of some GUI settings
added option to choose which files to export in the config
improved occasional issue of having slightly distorted audio on LTE modules
added graceful activate method
added option to force driver to load modules from configuration for FXS detection
when using port numbers in dialplan as from_id, settings from configured PSTN group will be applied
made registration fail timeout configurable
added experimental option isdn_layer_fix to bring back occasionaly lost ISDN layer 2 in some environments
added registration offset feature, so when working with multiple SIP accounts REGISTER requests are not sent all at once
added experimental option ltedtmf, to select either inband/outbound/both DTMF events to be forwarded
Firmware Version 20.05.04 or newer needs to be installed, before installing the 21.02 firmware. TLS SIP server certificate is now verified by default, make sure you upload the correct certificate or disable verification in the stack settings! |
added new installation process to install 23.01 firmware / added security checking on 21.X firmware installation
enhanced provisioning (client certificate managed by beroCloud; added button to download certificate; optimized on start-up; added debug mode; HTTP User agent customizable)
improved recovery mode (restoring device via TFTP)
improved interface with beroCloud (works now in update-/recovery-mode)
improved analog FXS detection
improved logs on installation and on start-up (keep the last five entries)
pool.ntp.org set as default ntphost on factory reset
added possibility to send and receive SMS in PDU mode
improved some issues sending SMS
added config settings sip_from_user_all_setting, sip_to_user_all_setting, sip_request_uri_all_setting
parsing SIP headers now case insensitive
setting sip_contact_all_setting for S2S outgoing contact header
ISDN int and nat prefixes are not mandatory anymore
ISDN city code not mandatory anymore
SIP User Agent customizable now
fixed issue uploading certificate (HTTPS - GUI)
fixed issues on factory reset
fixed SNMP RAM reporting when CDR logging was enabled
fixed some issue with DNS resolving
fixed sporadically sending unwanted BYE when a call comes in right after a call was ended
fixed rare case where API default user password was wrong
fixed mdio tool for new 6_12 MSP bonding firmware
fixed sending NAT probe in transparent SIP->SIP mode
fixed passing through crypto strings in transparent mode for SIP→SIP
fixed display of SNMP Server auth/encrypt values
fixed some dialplan issues
fixed some ntphost issues
fixed some memory leaks
fixed parsing of analog CID in SDMF format
fixed issues isgw ending and not restarting on some ISDN communication errors
fixed SIP-SIP transcoding call when sending to our own WAN device
fixed passing through telephony payload in transparent mode for SIP→SIP
fixed some GUI issues
improved FXS module detection
improved updating mechanism to cope with low space on the device
improved overall reliability of recovery mode
fix issue on cloud access, Market page, ConfigurationRestoreBackup API call
fixed issue on hostname configuration when an empty string is submittted
update provisioning to use TLS with standard HTTPS server
disable client side authentification (provisioning over tls)
non-SBC Gateways (two line interfaces) can not be upgraded to FPGA version > 9
forbid to install FPGA older than VERSION 12 on SBC (three line interfaces) device
fixed occasional showing of empty isgw memory usage value in SNMP request
SIP headers are now parsed case insensitive
made DNS SRV change registration behaviour configurable
added config settings: sip_from_user_all_setting, sip_to_user_all_setting, sip_request_uri_all_setting
added experimental option s2s_transparent_modify_rfc2833_payload (can modify rfc2833 payload type in combination with experimental_option syscall_mode=5)
added experimental option isdn_layer_fix
fixed issue when DNS server was not up on isgw start
fixed: setting correctly config option sip_contact_all_setting for S2S outgoing contact header
fixed sending NAT probe (s2s_nat_opener) in SIP->SIP in transparent/iptables mode
fixed SIP->SIP transcoding voice channel when gateway RTP target is own WAN IP
fixed: now correctly setting source port in transparent mode (SIP->SIP) when two different peers have the same remote port
fixed: passing through the telephony payload type in SDP on transparent calls
fixed: now passing through crypto strings in transparent mode for SIP→SIP
fixed: on sip-sip offering the best matching codec from the configured codec list
fixed NTP issue when DNS was not configured
fixed memory leak occurring on outgoing SIP INVITEs
fixed problems on LTE port initialization
fixed issue uploading a certificate
fixed behaviour when using an individual ISDN port as FromID in dialplan instead of the port group
fixed graceful-activate / reboot behavior
removed requirement to set national code in ISDN group (denmark, luxembourg)
displaying client certificate in the GUI
improved update-installation process
added experimental_option gsm_pdu_mode
fixed some memory leaks
fixed erroneous behaviour when call_forwarding_mode=Invite
fixed issue setting gsm_clir variable
fixed issue on firmware update via provisioning
fixed NetConf issue when DHCP mode
fixed registering issue on beroCloud settings in DHCP mode
fixed issues in SIP->SIP SRTP handling
fixed using incorrect TLS listening port in contact string
fixed setting incorrect contact header on outgoing T.38 re-invite
fixed LTE signal strength display and values
fixed issues on Provisioning Page
fixed API Calls (NetworkSetConfiguration, FirmwareUpdate, ProvisioningTriggerConfig)
added recovery mode
added fortified update and downgrade mechanism
added call statistics for SIP to SIP calls
added 1tr119 and sips_on_tls config option option
added multi actions to SIP peers page (set callgroups, sip stack, register)
added experimental option no_reason_header
added experimental option memory watchdog
added Provisioning over TLS and the corresponding 'Redirect Task' on beroCloud
quicker initialization of LTE modules, quicker GUI accessibililty during LTE initialization
improved fulltrace behaviour (speed, single file download behaviour, analog port debug)
order of header fields for the from source auto value detection can be defined now
reworked SNMP GUI, added support for SNMP-traps
updated lighttpd version to 1.4.50
ACL rules support support DROP and ACCEPT rules
reworked the SIP stack configuratioin
allowing setting empty national code for SIP and ISDN
enhanced the Info Page (SIP License and Storage information displayed)
fixed issue with DTMF passthrough on SIP->SIP calls
fixed SDP handling issue, when there were multiple 180 RINGING messages with and without SDP
fixed some issues on firmware downgrade
fixed some issues on firmware upgrade
fixed issue on incoming LTE call, when one call is already connected
fixed error in API function TelephonyGetCdr and added new parameter download_old
fixed some issues in the Fulltrace
fixed handling of TON in Calldeflect/Rerouting facility for generating 302 redirect
fixed rejecting calldeflect/rerouting with returnError Facility when calldeflect is disabled
fixed LTE state table and added more information about technology
added GUI support for TLS 1.2
added SIP support for TLS 1.0, TLS 1.1 TLS 1.2, removed support for SSLv3
added TLS certificate verification now enabled by default (if there is a problem registering to your TLS SIP server, upload the correct certificate or disable verification)
added new licensing mechanism for SIP-SIP calls (transcoding and transparent channels)
added made RTP handling selectable (transparent vs transcoding calls)
added more performant handling of transparent SIP->SIP RTP streams
added access to isgw telnet konsole in the GUI
added custom tones to be configurable in the GUI for FXO
added wait4idle timeout configurable on FXO to handle incoming ringing
added verify_ringing option to FXO port group
added FXS ring oscillator intervals to be configurable
added experimental option ltedtmf, to handle non inband DTMF indications on LTE calls
added config option anonymous_from_user_on_privacy
added activationDiversionQ numbers are now available as dialplan variable
added showing the network link status in GUI
added interface can now be given for ACL rules
added reworked DoS protection settings, can now be configured for different packet types and interfaces
added more information to call statistic (formerly port statistic)
reworked the Network Settings page
improved debugging of PSTN ports in fulltrace
performance improvements in GUI
reduced memory consumption of isgw
using ephemeral portrange 50000-60000 for outgoing UDP requests
after a hardware factory reset via jumper, the SSH, HTTP and HTTPS Ports will be opened
added firmware integrity check during installation
added new installation mechanism (required to install 23.01 Firmware or higher)
added automatic reboot on update-mode (when no activity)
added recover feature
enhanced berofix modules for ANALOG module
allowing downgrade via provisioning
allowing empty national code for SIP and ISDN
blocking recent FPGA installation on SBL / gateway
improved applying configuration to new call object
fixed analog CID parsing in SDMF format
fixed ACL behavior on deletion
fixed ISGW crash on call deflect when call_forwarding_mode is invite
fixed Wizard to enable more than 9 PSTN ports
fixed hardcoded DNS refresh interval
fixed problem handling SDP, when Ringing event withous SDP was sent before
fixed bug leading to broken oad TON prefixing
fixed using correct tls listening port in contact string for outgoing invites
fixed bug when using more than 4 LTE ports
fix issue with call progress table in the dialplan popup (sip)
added experimental option ltedtmf, enabling #DTMFEV indications on lte
added config option anonymous_from_user_on_privacy
added call_forwarding_method options invite_incoming and diversion_incoming
added reason header can now be deactivated with experimental option no_reason_header
made updating more secure by checking file system space before update
bearer capability will now be passed through transparently on ISDN->ISDN calls
added config option bearer
fixed problem on ISDN->ISDN calls when CLEARMODE codec was used
fixed control of LED lights on 24 FXS gateway
changed format of analog port voltage logging
fixed problem sending Invites to different proxy after registrar failure
fixed XSS issue
fix for sdpless re-invite
fix for incoming sdpless invite
fix for authenticating incoming bye message
fixed passing through SIP INFO for SIP-ISDN
fixed Language issue on State page
fix for audio problems for new LTE module
fixed sending syslog messages after restart
fixed generating some config files after hardware factory reset
fixed problem with DNS SRV when DNS server was not available on startup
added using SRV alternative peers, when register attempt fails
reordered items on GUI logging page
fixed sporadically ocurring problem with PSTN database setup after a factory reset
added ACL rule to allow access from bfdetect tool
fixed typo in german translation, rendering the GUI inaccessible
IMPORTANT INFO: Switching to german language renders the GUI inaccessible due to a bug.
(replace the string <gateway-IP> with the IP of your gateway) |
changed if ISDN port is blocked, calls are rejected with cause 44 now
added allowing schedule of graceful activate/reboot when no calls are affected
added support for callgroups
added option max_simultaneous call for SIP groups
added improved network troubleshooting
added support for SNMPv3 UDP/TLS
added config option t38_ignore_offer
added contact_all_setting allowing to modify registration contact header
added monitored_sip_peer option to allow shutting down ISDN ports when SIP peer goes down
added improved time settings configuration, Summer/Wintertime change now configurable
fixed issues in time settings management
fixed issues during factory reset
fixed showing used ISDN channel in CDR when call was rejected directly after SETUP
fixed backup/restore of hostname
fixed not accepting bye from caller in early dialog on SIP to SIP calls
fixed issues with GSM table not loading
fixed not setting contact in validating OPTIONS packets, when validating without registration
fixed dnssrv request for peers configured with TLS
fixed GSM/LTE unicode converter
fixed restoring NTP host from backup
fixed not using configured dtmflen value, when rfc2833_passthrough=0
fixed memory leak when using validate without registration
fixed handling of an re-invite not containing SDP
fixed sending out rfc2833 events when dtmfmode=inband was used
fixed not refreshing DHCP anymore when request failed
fixed problem adding some network routes
fixed issue correctly deleting and deregistering cloud settings on factory reset
fixed number prefixing when failover_account is used, using settings from failover account
fixed crash when restarting ISDN port that was not configured
several GUI fixes and improvements
GUI allowing multiple selection and deletion of SIP groups
GUI fixed dialplan javascript delete issue
GUI added from_display and other variables to list of selectable SIP settings
GUI added cancel button to Full-Trace
GUI added additional options for managing Full-Trace
GUI added ajax refresh for active calls
disabled PRACK support by default, if not explicitly enabled via experimental option enable_prack
disabled support for SNMP traps
fixed problem resetting the GUI password
fixed unnecessary restarting of voice channel on SIP->SIP calls in case of reinvite
fixed memory leaks occuring on gateways with T1E1 modules
fixed offering wrong RTP port, when dialplan was searched multiple times on analog overlap dialing calls
fixed addressing deprecated/wrong target IP on outgoing INVITEs when DNS SRV was used and DNS address changed occurred
fixed SMS not being able to contain € sign
added SMS Webhook, allowing URL to be contacted on SMS reception
added Network troubleshooting features to GUI
added config option header_passthrough_filter to be used together with config option header_passthrough
added GUI option to disable the cloud redirect function
added informative message about cloud status
restart_after_linkloss sends additional restart message on channel 0
closing dialplan_debug window on session timeout
different loglevel settings are now individually configurable in the GUI
supporting now new beroNet LTE module
cloud monitoring and status will be enabled when registering to the cloud
optimized Yellow Activate (ACL rules restarted only after have been updated)
homogenized physical and GUI LEDs for LTE module
fixed issue updating DNS Server
fixed GUI issues (Call-Progress Table, Wizard, Market)
fixed 3CX Wizard redirect issue
on GSM and LTE cards SMS stored on the SIM will be deleted during initialization (please backup any important messages)
fixed some stability issues
fixed some memory leaks
fixed some hang issuesimproved GUI performance
fixed GUI bug saving SIP peer secret
reworked DNS system, using internal DNS cache for better performance
fixed some display problems during dialplan debug
added support for "Technische Richtlinien Notruf" of the Telekom
fixed problem with ISDN calldeflect when using very long numbers
fixed some Labelling errors in the GUI
added new Wizard for 3cx and other VoiP PBX
added Api Functions CFActivate and CFDeactivate
added support for LE910C1 LTE
beroCloud Configuration now has own menu entry
showing beroCloud registration status on demand
added support for SMS delivery reports, added Api function SmsGetSendReport
incoming SMS only deleted automatically if there are more than 1000 unfetched
added experimental option telnet_max_connections, default value is 4
increased allowed syslog port range
fixed issue with SDP-less re-invite in connected SIP->SIP call
fixed wrongfully setting SDP on SIP->SIP call in some situations
fixed losing part of SIP configuration when Factory Reset is done and 'keep SIP settings' is selected
added preparations for beroCloud redirection service
fixed problem on SIP→ISDN BRI/NT calls not going to next free port if all channels are occupied
added keeping 'ACL settings' during Wizard Configuration
added a PHP validator during a GET or a POST request
improved the Javascript validator
improved the security against XSS and SQL injection vulnerabilities
fixed problem restarting LTE port on State page
fixed problem saving 'Remote Management' Settings in German-Mode
fixed some Javascript errors or not expected behavior
fixed GUI-error with 'Dialtone passthrough' settings from FXO module
fixed GUI-bug from Dialplan popup settings (Easy and Advanced Mode)
fixed prepending of VAZ (prefixes) when no national/international prefixes have been set
fixed updating and reload problems of causes map
fixed erroneous reregister request to the berocloud, after deregistering, improved GUI presentation
fixed missing values in SIP settings dropdowns (e.g. in From user/Pai User ...)
fixed displaying incorrect value in the GUI for Destination/Source Number format
fixed not capturing localhost SIP message during fulltrace
fixed early audio on PSTN->PSTN calls
fixed instability when SIP INFO package without payload was received
fixed problem with downgrade from 19.xx to 16.xx not starting, occurring on very old gateways
fixed losing of transport settings for the SIP stack when upgrading from 16.xx to 20.xx
fixed showing of SMS Tab in the GUI for LTE modules
fixed problem on LAN/WAN using same MAC addresses, using different MACs now
some minor GUI layout improvements
added new value auto for SIP dialplan source and dialplan destination and made it default
added new config option sdp_codec_selection, passing through the offered codecs
using second DNS server if the first on fails
GUI session timer is refreshed on click activity
on BRI rejecting now SETUP, when two calls are already
improved Dialplan Debugging
some GUI modifications (some pages are now organized in tabs, showing additional information)
added GSM/LTE provider selection
improved presentation of GSM/LTE status
added validation as option to SIP peers without registration
fixed issue when configuring
fixed issue with 2FXO-1FXS module combination
improved behaviour of the LEDs of the PSTN RJ45 ports
fixed application of early_audio settings on SIP->FXO calls
added option for permanent remote support
fixed encoding REDIR_NR in ISDN NT-stack
fixed short hangs triggered by non-responding DNS server
default values changed: sip_from_display_setting=none (from display empty)
default values changed: sip_ppi_user_setting=none (PPI header not added)
default values changed: codecs=pcma (only pcma is now offered by default)
default values changed: t38=0 (no t38 on PSTN/SIP calls)
default values changed: s2s_t38_passthrough=0 (no t38 on SIP2SIP calls)
default values changed: cd=1 (call deflect is enabled by defaull)
default values changed: i2s_destination_numformat (default now: +(int)(nat)num, old default unknown: (num))
default values changed: i2s_source_numformat (default now: +(int)(nat)num, old default unknown: (num))
default values changed: TCP and UDP are by default activated now on each SIP stack
NTP server pool.ntp.org configured by default
no default ACL rules anymore, only access from local network is initially configured
fixed bug losing the causes map on firmware update
fixed bug re-registering SIP peers on DNS server timeout
fixed bug on SIP->FXO calls, when FXO->SIP call was already active on analog port
added support of ie_on_sip X-BF_FACILITY_MLPP header
fixed generation of pcap traces when VLAN was activated
fixed issues with PCM bridging
SIP CLEARMODE calls automatically now offer UNRESTRICTED_DIGITAL as bearer capability on ISDN, and vice versa by default
added clearmode_auto option, allowing to configure the CLEARMODE/UNRESTRICTED_DIGITAL bearer configuration
fixed issue when re-invite SDP target url and gateway IP are identical
fixed problem of displaying T1E1 modules in the GUI
fixed small memory leak
fixed issue using wrong cidlen and crlen on PRI
added codec selection for LTE
added config option gsm_clir to configure
added configurability of ISDN cidlen and crlen for BRI
fixed issue uploading/deleting SIP TLS certificates
fixed incorrect prepending of country and city code
fixed collecting of debug data during full-trace
fixed slow down of the GUI while checking for firmware
cloud based remote session can be now ended via the GUI
better handling of privacy header in auto mode on SIP->SIP calls (outgoing leg uses values of first leg)
fixed issue with incomplete configuration after update from 16.xx firmware
fixed log rotating pcap files during full trace
fixed problem of setting nameserver when DHCP is on
fixed issue on SIP->ISDN calls when externip of stun server was used
fixed netconfigserver reported ip when bonding mode is selected
added configurability of NTP server when DHCP is activated, explicitly given NTP server has preference
fixed missing files in Full Trace (gsm debug.log on LTE, LAN pcap in bonding mode)
added missing Termination Causes to CDR
Reworked and improved the different Gateway Wizards
fixed erroneous prepending of configured prefix to automatic number format
added GUI field for configuring SIP contact header
fixed problem with FXS not working
fixed file descriptor leaks
removed s2s_302_passthrough
added call_forwarding_mode instead, allowing to either send a 302 or a reinvite (with optional diversion header)
config values from_sip_dest_setting (dialplan destination) and from_sip_src_setting (dialplan source) can be now defined with a regular expression
added config values sip_priv*, to add custom headers to an outgoing Invite
Most notable changes in comparison to current 16.15 firmware:
Support for beroNet 3LI gateways with two ethernet ports
Separate SIP stacks can be configured for each network interface
Sip peers can be individually configured to use their own bind port
Facilities for DoS Protection
fixed slowness issue in GUI
fixed initialization issue with FXO and FXS after hardware factory reset was done
better handling of privacy header in auto mode on SIP->SIP calls (outgoing leg uses values of first leg)
added config option isdn_no_sending_complete
fixed incorrect prepending of country and city code
fixed sending two SETUPs on SIP->ISDN calls, when stun or externip was enabled
fixed problem with setting iptables rules for SIP peers with long group name
fixed printing of configuration settings
fixed error in SDP when rejecting T.38, if multiple media sections were offered
fixed problem on SIP->FXO calls accidentally terminating FXO->SIP call on used port
fixed potential resource leak of RTP ports on SIP->SIP calls
for SIP->SIP calls, dtmfmode can be configured for both SIP legs individually
new options for s2s_nat_opener
better handling of potentially broken https certificates
fixed error when restoring isgw.causes
fixed XSS vulnerability
fixed prefixing dad with ton prefix
fixed problem with no audio on SIP->SIP calls when send_only was offered in SDP
fixed problem for with SDP offers offering codecs with higher sampling rate
fixed offering of tracing ISDN ports in fulltrace only when no ISDN ports are available
fixed issue with analog port configuration during firmware update
added config option config option isdn2sip_source_numformat
added support for FXO adminstrative offhook mode
added new parameters to use_privacy_header config option, options are now auto, off, none and id
added config option sip_302_contact_user
added config option sip_302_contact_url
removed config_option s2i_transfer_url_cd
removed config option d2s_oad_prefix_setting
removed SIP profile management from GUI
config option disable_prack now used for inbound sip calls too
on T1/E1 modules E1-HDLC is used as default, for reconfiguring to CAS/T1 update the hardware configuration during a factory reset now
now displaying hardware configuration table when the module has been be just plugged
now display pipe installation-progress when a new tab is opened
improved collecting of debug data during full-trace
improved handling of FSK caller-id detection on FXO ports
a couple of cosmetic improvements in the GUI
fixed disabling of echo canceller
fixed VAZ prepending in case of usage of failover proxy
fixed VAZ prepending in GSM and FXS calls
fixed timeslot allocation for T1E1 module
fixed problem with s2s licence reading for new cards starting with serial number 30
added config option ec_sparse_params
added config option s2s_rtp_passthrough
added config option r2config for setting MFCR2 parameters
added codes for rejecting RTP or T.38, status code can be configured now in the causes map
changed default reject status code from 415 to 488 for T.38
changed default for sip_info_on_facility to 0
fixed bug affecting incoming invites unintentionally being rejected sometimes
fixed rtp_packet_size config option for SIP→SIP calls
fixed SIP→SIP license parsing on new 3-line-interface gateway
fixed setting correct cause from causes.map in ISDN DISCONNECT and RELEASE_COMPLETE, if no dialplan entry was found
fixed wrong setting of contact string, when externip/stun was not used
fixed a memory leak when using DNS SRV
fixed error when registering to the cloud
fixed password problem between Zeroconf and PBX Wizard
fixed error in SIP→SIP dialplan entry when using dialplan variables
fixed bug with timeshift of timezone with half hours
contact header SIP URL port will always be formatted including port, even if it is 5060
fulltrace starting now rotates the logfiles
added config option s2s_t38_passthrough
changed maximum SIP registration refresh value from 2000 to 3600
fixed error configuring PCM slave mode
fixed behaviour when T.38 reinvite was rejected and a G711 re-invite was received
fixed setting correct port when sip domain is used
using uppercase for codecs in SDP now
isdn2sip_destination_numformat now used for fxs call as well
added config option sip_info_on_facility
added config option s2s_nat_opener
handling 302 correctly on SIP→SIP calls
waiting longer time to go into idle state on FXO calls
added experimental_option emergency_numbers
fixed stability issues
experimental option no100fix can be now 0, 1 or 2
now sending 302 transfer in SIP direction if calldeflect is received on ISDN side and cd=1
fixed no audio problem when SDP was contained in 182 message
refreshing registrations when DNS query result has changed
added toneset for Ireland
fixed hostname setting
fixed userapp-version issue
fixed not printing first line in dialplan debug
fixed call progress table backup&restore bug
changed default setting of ppi/rpi/xheaders, they are now disabled by default, sip_pai_user=${new_source} now
hostname/gateway not displayed in dhcp mode
ntpserver setting overrides DHCP provided ntp server if set
fix for parsing sip ppi header
added option no100fix, fixing problem when no 100 trying is received
fixed sip failover behaviour on gsm
fixed memory leak when using stun
fixed problem encoding ISDN connected party number
fixed security issues with telnet access to card
fixed SIP problems with peers not listening on standard SIP port
fixed GUI error message after language switch
fixed GUI error saving FXS Clip
fixed problem with dialplan restore
debug files get tagged with their md5sum now
improved DNS SRV handling
improved SMS handling
fixed GUI error setting t38 in dialplan
fixed GUI error saving GSM extension
fixed GUI error setting call progress table
fixed GUI Dialplan-Debug behaviour
fixed one way audio problem on SIP→SIP calls
fixed reinvite handling on SIP→SIP calls
fixed error storing "more settings" in dialplan
fixed limitation on density of inbound SIP calls
new responsive GUI
new logging facilities
new config option skip_setup_ack
new config option resync_on_notify
new config option isdn_dad_setting
new config option dialtone_digits
added dialplan variable new_destination_auto
fixed VLAN issues
fixed problem with dialplan copy and multiple delete
fixed SMS problems
fixed problem with SIP info messaged on FXS calls
try option 67 on DHCP if option 66 is missing
display name in register requests now in quotes
in-call ISDN Information messages will now converted to DTMF
fixed some minor issues in the GUI
fixed problem with provisioning polling
fixed config option outgoing_bchannels
new REST-API functions
netconfigserver is now configurable
improved qsig name handling
support for DNS SRV
minimum accepted session expires now configurable
Major security fixes, more info here
T38 disable doesn't save in the Dialplan - as workaround you can configure "t38=0" in the text-area under the "more" tab.
added option no100fix, fixing problem when no 100 trying is received
fixed sip failover behaviour on gsm
fixed memory leak when using stun
fixed problem encoding ISDN connected party number
Security fix, telnet interface not more accessible from LAN by default
fixed instablity problems in SIP stack occurring when a large time change shift happens
fixed problem in dialplan debug
fixed problem with incoming 181 and 182 provisional responses
fixed bug in Factory Reset
fixed return code for provisioning
reworked session handling for dialplan debug for more security
reworked GUI session handling. GUI is now more secure as its files can only be accessed via the GUI
new_source_auto will now use TON_SUBSCRIBER as default
new config option skip_setup_ack
new config option dialtone_digits for FXS inbound cals
new config option emergency
display name in register requests now in quotes
in-call ISDN Information messages will now converted to DTMF
fixed config option outgoing_bchannels
bugfix gsm provisioning
fixed problem with SIP info messaged on FXS calls
fixed problem with SDP offers containing 16000Hz Codecs