• new: server to server VoIP tunneling/encryption/obfuscation over multiple short living UDP sockets
• new: diversion indication (RFC 5806)
• new: configurable admin alerter module with email, sms and push notifications (sendalert settings)
• new: integration with any external outbound API or database
• new: FCM HTTP v1 API protocol implementation
• new: sendpush API (ability to send various push notifications)
• new: reloadfcm and pushkitrst commands
• new: addiec and addcc global setting (simple/fast option if you don't wish to use the Rules)
• new: handle also the old pn-type / app-id / pn-tok contact push parameters
• new: best match based local IP address usage
• new: bauastrict configuration to specify how strict the block/allow lists are checked
• new: blockhttprequests block suspicious HTTP requests and auto ban
• new: extern_api_ settings
• new: defhttpcontenttype setting
• new: push notification on cancel
• new: multi-threaded puskit apple apns push notification requests
• new: multiple separate fcm projects
• new: push sound settings
• new: replacecalleroncallspriority setting
• new: forceexternalip setting
• new: rerouteconnected setting
• new: sip_whitelist setting
• new: configurable rtp fast path
• new: automatically assign dns record if not provided by user
• changed: prioritize the SIP server CallerID if both the caller-called has it set
• improved: session handling with lost ACK for 200 OK call connect
• improved: UDP IOCP socket module
• improved: rewrite rules
• improved: quick rtp relay optimizations
• improved: IOCP socket performance
• improved: HTTP requests performance and adaptive threading
• improved: socket resend on failure
• improved: reuse endpoint for same user/ip:port
• improved: session-timer handling
• improved: WebRTC fork
• improved: block ua
• improved: blocknotbilled calls implemented also for postpaid accounts
• improved: billing will find the price also for target numbers with IEC
• improved: supervisor prevent subsequent resets
• improved: push performance
• improved: pushkit topic setting
• improved: push for IM (send push notification on no answer for SIP MESSAGE or according to your settings)
• improved: API parameter validation
• improved: mime-type guess
• improved: IVR SIP DTMF actions
• improved: routing destination list filter and ordering in compact editions
• improved: offline sip messaging message cache
• improved: built-in firewall request filtering
• improved: SDAC parameters optimizations
• improved: supervisor emails threshold
• improved: routing pattern cloning
• improved: various minor changes on the admin user interface
• improved: enduser webportal session management
• improved: call recording performance
• fix: nonce count interpreted incorrectly in some circumstances
• fix: TCP/TLS target socket mismatch on send
• fix: MManage admin client crash on close or on Logout
• fix: LAN/WAN routing bug in some circumstances
• fix: SIP server register timeouts
• fix: upper server register timeouts
• fix: call disconnect after around ~30 seconds in some circumstances
• fix: API doesn't allow space in caller-id and displayname
• fix: failure to resolve IP for new TCP/TLS connections to a domain
• fix: routing rule to enduser
• fix: create new billing group
• fix: restart on testmutex wait failure
• fix: request text description in the first line
• fix: mulithread processing not activated before database connection bug
• fix: ipv6 address length limitation bug
• fix: CDR backup initialization
• fix: db post record exception
• fix: IsRegistered access violation
• fix: webportal bindip change
• fix: bug 5985 initselfcheck
• fix: other various minor bug fixes

This is a quality upgrade with bug fixes and improvements. No new features have been merged into this release.

• new: call recording with all codecs (including also speex and ilbc)
• new: attended call transfer with replaces support
• new: support for Replaces header (in INVITE and also the transfer flag)
• new: music on hold (MOH)
• new: voicemail notifications
• new: central addressbook/contacts book API
• new: file transfer over http/ftp
• new: loglevel change from MManage will adjust also max log queue
• new: serialized maintenance tasks
• new: cdr/billing service for external sources
• new: SMS sender from separate process
• new: message length based SMS cost calculation
• new: log packet types and thread names as string (not only ID)
• new: voice recording for quality control purposes (recordpercent)
• new: disconnect code 497 Transfer Failure
• new: clone settings option
• new: checkmediatimeout configuration
• new: pushnotification_websocket configuration
• new: honormediamuteorhold configuration
• improved: RTP threading
• improved: SSE performance
• improved: maintenance tasks pipelining
• improved: call fork handling
• improved: ipv4 vs ipv6 address type
• improved: database operations speedup
• improved: message parsing performance optimizations
• improved: unregistrations
• improved: digest auth-int
• improved: auto provisioning
• improved: domain name parsing
• improved: contact expires handling
• improved: tls URI parsing
• improved: server start action
• improved: monthly users cleanup
• improved: loop detect
• improved: SIP from/to tag management
• improved: media timeout detection
• improved: offline chat
• improved: TLS/SIPS module
• improved: better handle TLS certificate no renewal required
• improved: call transfer call-leg management
• improved: SQL input extra validations
• improved: better config wizard user input validations
• improved: username validations
• improved: file rename (core api)
• improved: string parsing speedup
• improved: API requests parser
• improved: SMPP protocol
• improved: handle failed sockets on service shutdown
• improved: random for more sources
• improved: speex and opus wideband call recording
• improved: fast auth user banning
• improved: log level change
• improved: a list of other minor improvements
• fix: crop too long SDP
• fix: file handle leak
• fix: opus sample rate defaults to 48000
• fix: don't send X-Peer with 4xx responses
• fix: FCM save from config wizard
• fix: expires tag in contact headers
• fix: move old data to backup database
• fix: MManage users save exception
• fix: other minor bug fixes

This is a quality upgrade with bug fixes and improvements. No new features have been merged into this release.

• new: built-in high performance TLS module using the latest openssl library
• new: TLS for multiple domains (separate TLS cert/domain for different nodes)
• new: compatibility with Azure SQL database (including TLS connectivity and trustcert/resolveip database connectivity settings)
• new: server side call transfer (not only REFER forward)
• new: a separate state machine for call transfer
• new: handle local replaces
• new: subscribe for transfer state
• new: maxregroutingthreads and regroutingthreadtreshold configuration
• new: MManage Configuration Wizard SRTP setting
• new: MManage Users and Devices port setting
• new: MManage Users and Devices transport setting
• new: MManage Users and Devices SRTP setting
• new: ping API
• new: addserver API
• new: deluser API
• new: newusersevent API
• new: userchanged API
• new: detect username/password upper/lower case (capslook) misspell
• new: SIP event-id handling
• new: named threads
• new: quick/simplified message parser for p2p/direct outbound message routing
• new: log first apple push and google FCM request success/failure
• new: keepoldregclients, forceroutingforregister, routingforregistercache and removeoldregaddr configurations
• new: sockconnecttimeout and socktimeoutmultiplier configuration
• new: tls_external configuration
• new: tls_security configuration
• new: tls_preinit configuration
• new: tls_ciphers configuration
• new: pushnotification_topic configuration
• new: pushnotification_ignoresubssame configuration
• new: reuseep and reuseepauth configuration
• new: srtp_auth configuration
• new: re-auth on user record change (apply new password or auth ip in real time)
• new: loopdetect configuration
• new: allowserverservercalls configuration
• new: strictdomainiprefresh configuration
• new: discontransfer, holdontransfer and playtransfertone configurations
• new: set config mutex protection
• new: auto convert multi-party call to simple call after no more forks
• new: block requests per user/peer/ip on too much pending sessions
• new: async message routing from separate threads for higher performance
• new: secondary bind ip and port
• new: SIP message Reason header (parsing and forwarding)
• new: auto correct if server ip (bind/local/domain) is configured incorrectly
• new: parse authentication-info
• new: implement also old/nonstandard push notification based on push-token-type / push-token-app / push-token-id
• improved: TLS certificate renewal time
• improved: SRTP
• improved: WebRTC protocol module
• improved: loop detect
• improved: pushkit certificate upgrade
• improved: regstat, epreg API
• improved: SIP call transfer
• improved: WebRTC call transfer
• improved: SIP URI username/displayname processing
• improved: IP address class
• improved: md5 auth qop and auth-int handling
• improved: called number normalizations
• improved: automatic maximum degree of parallelism for stored procedures
• improved: upper unregister for push notification sessions
• improved: call fork and redial
• improved: resend auth request on peer change
• improved: rtp stream mutex (and not using any mutex when there is no way for race conditions)
• improved: rtp candidates list
• improved: SIP MESSAGE / IM
• improved: G.722 codec
• improved: service startup logs
• improved: optimal default self-test register interval
• improved: get best device state alg and store online state
• improved: load statistics
• improved: DNS queries
• improved: prefer HTTPS over FTP for modules auto download/install
• improved: ring groups
• improved: compact database v_check_calleduser target user IEC prefix handling
• improved: server to server encrypted/obfuscated tunnel over multiple short living streams
• improved: user=phone should be removed by default for gateway
• improved: don't send admin email reports from idle servers
• improved: auth with different username/sipusername
• improved: Route / Record-Route handling
• improved: replaceanumwithdid should default to a yes value on only low load
• improved: ignore config settings category (try all also without category)
• improved: built-in webportal comment memos should not be resizable/clickable
• improved: smaller thread stack size
• improved: increase maxsipmessagesize on local lan
• improved: currency conversion
• improved: routing to webportal
• improved: codec rate calculations
• improved: memory usage meassurements
• improved: check child users on disableoldusers / deleteoldusers
• improved: do not log max line reach multiple time (remember last)
• improved: don't send user=phone by default or retry the invite first time if 404 Not found; sendsipuseruripart
• improved: check fastreg for api wsunreg, regstat, epreg, reguser and elsewhere
• improved: merge SIP MESSAGE sessions from the same peers
• improved: stricter SIP header fields usage
• improved: MManage: cache tb_settings in different server sections (important if using multiple servers)
• improved: MManage async user stats
• improved: MManage chat logs form, warn if logmessenger global config is not set to 1 or 2
• improved: MManage Config wizard -> Fix weak password (not enabled by default)
• improved: MManage rtmp protocol not remmembered if unchecked
• improved: MManage auto change user records ip address on ip change from config wizard
• improved: MManage config wizard: don't accept IP address as domain name
• improved: MManage on service stop, stop also the supervisor on local servers
• improved: MManage rules: set record to edit mode if drop down changes
• fix: SRTP key Base64Decode to binary instead of string
• fix: double CANCEL/BYE
• fix: handle missing convertdtmf field
• fix: never close the main udp listener socket
• fix: Via header is missing from SIP/2.0 405 Method Not Allowed publish
• fix: bad parameters for buildauth
• fix: push notification page doesn't always appears in the config wizard
• fix: TCP streams parsing
• fix: race condition on touching ep from both mainlogic timer and asynctransportrec
• fix: CoInitialize not called on thread before database operation
• fix: MManage rules caller id change should put the db in edit/savable mode
• fix: MManage cannot find directions.sql
• fix: MManage cannot connect if sql port is different/not specified
• fix: PBX module memory leak
• fix: Access violation on API_Info
• fix: a list of other minor bug fixes

Quality upgrade with a list of bug fixes, improvements and optimizations.

• new: high performance server to server obfuscated/encrypted tunnel
• new: enable/disable the TTS (text to speech) module from the config wizard
• new: changed thread model. added lightweight threads with tiny initial stack size
• new: raw/clear text CLI access
• new: auto block scanning attempts
• new: RFC 8599 Push Notification for SIP
• new: HTTP/2 support for apple APNS
• new: apnspushtype setting (apns-push-type: voip/alert)
• new: SRTP support (including bypass, decoding and reencoding)
• new: periodic network connectivity tests and logs
• new: outbound trunk routing prioritization by registered status
• new: unreg api
• new: upperunreg api
• new: http proxy exceptions
• new: blockeuawithnouseragent config
• new: forcedcallernidentity/replacecalleroncalls config
• new: delayed send INVITE to secndary targets
• new: OBD mass dialer beta
• improved: user-agent based banning
• improved: faster packet type detection
• improved: auto detect aged certificate even if config is wrong
• improved: API compatibility with both the old and new auth
• improved: all internal, DB and API access encrypted by default
• improved: iOS pushkit certificate autodetect and load also from p12
• improved: auto guess best caller-id to be used by default
• improved: handling/sending user=phone flag in the SIP request URI
• improved: call hold state management
• improved: better handling b-legs and call forking using different transport protocol (UDP/TCP/TLS/WebSocket)
• improved: push reinvite on app register
• improved: auto unreg push client when/only if neccessary
• improved: apple apns push topic and certificate handling
• improved: Caller ID handling with more related settings
• improved: dns resolver; skip dnsrefresh if domain was changed
• improved: universal web service listener
• improved: temp/old files cleanup
• improved: FCM and apple apns push notifications sending
• improved: apple apns voip push notification certificate handling
• improved: upgrade to new Let's Encrypt certificate formats
• improved: ports management
• improved: default strong filestorage password
• improved: mmanage server configuration caching for faster startup
• improved: mmanage config wizard detect if supervisor is already installed
• improved: mmanage various config wizard improvements
• improved: RTP timeout handling and better default configuration
• improved: RTP payload handler
• improved: SDP manipulation routines
• improved: external port availability test
• improved: media routing/bypass decisions when set to auto
• improved: mmanage rules. display if not found due to search filter
• improved: call recording (reliablecallrecording config option)
• improved: alloweuserusercalls
• improved: many improvements for MManage
• fix: mmanage pricelist currency should default to global currency
• fix: disconnect reason playback exception
• fix: try to use a released query
• fix: duplicate dns caching
• fix: API examples
• fix: mmanage routing form always ask for save
• fix: mmanage warning at rules form if forwardcaller or forwardcalled is set
• fix: auto create ivr sets wrong campaign number
• fix: other various minor bug fixes
• fix: resolved 59 support tickets

Quality upgrade with bug fixes, minor improvements and an updated SIP/media stack.

• new: built-in HA
• new: DB server failover and failback
• new: autoban/fail2ban (see the Caller banning FAQ in the admin guide)
• new: call transfer failback to local if remote peer doesn't support REFER
• new: SMPP protocol implementation for outbound SMS messages
• new: DTMF failback to RFC2833 if remote peer doesn't support or answer for SIP INFO
• new: blockotherdomains config
• new: encrypted DB connection when necessary and possible (disabled by default on local connections)
• new: quick file backups and restore
• new: more win security rules
• new: mmanage md5 SIP auth test tool
• new: sub-enduser take the routing decision from parent user (routing after callerid)
• new: allow domain in tb_users_authip
• new: pushnotification_cancel config option
• new: securemove config option (secure call transfer strict checks if billing enabled)
• new: auto convert voice files to desired format
• new: backupdb, dbrestore, dbfailover CLI commands
• new: max pending auth request limitation
• improved: SIP and WebRTC signaling stacks general improvements
• improved: further billing speedup via v_getprice_int
• improved: media stack upgraded to latest stable release
• improved: handle separate transportprotocol and target domain per branch
• improved: session-timer handling
• improved: fast registrar module
• improved: auto route calls between nodes if target user is on other node (serverinstance)
• improved: early media handling
• improved: call forking
• improved: cseq handling
• improved: static firewall handling and DoS attack filtering
• improved: faster signaling message processing
• improved: handle duplicate packets
• improved: RTP NAT open don't try for local subnet
• improved: CANCEL send out only if required
• improved: better endpoint best local IP detection
• improved: voicerecdownload API
• improved: cdr API
• improved: unreg command
• improved: dns refresh
• improved: get best username for session
• improved: SIP message compose tracing
• improved: detect and handle no free disk space on file save
• improved: auto blacklists
• improved: auto remove push subscription on wrong statistics or subsequent wrong calls
• improved: other voip push notification related improvements
• improved: accept username for auth also like username@domain
• improved: supervisor service (MServiceHost)
• improved: minimize fast register message copy
• improved: better loglevel change handling
• improved: direct p2p calls for users behind same NAT (not if router can't route directly with public IP)
• improved: Content-Disposition header handling
• improved: media routing
• improved: username validity checks
• improved: get best IP, get if same subnet and get if routable functions
• improved: authentication security (doesn't accept upper/lowercase mismatch on strict settings)
• improved: NODE handling for WebRTC users
• improved: always load caller id for all CDR's
• improved: decrease credit on quick billing path
• improved: get file time
• improved: text to speech engine
• improved: block caller number
• improved: suspend advplayback on dtmf
• improved: don't calc statistics for blocked calls by our server
• improved: better NOTIFY routing after call transfer REFER
• improved: encrypt sip server's authentication accounts
• improved: mmanage: better sort for statistics
• improved: launch external apps reliability
• improved: RewriteReferToRepalces
• improved: a long list of minor improvements omitted from here
• improved: a long list of MManage admin client related improvements
• fix: mailsend sometimes fails
• fix: recorded file playback (sometimes was disabled by default)
• fix: "via" and "supported" short format processing
• fix: not enough memory errors on servers above 24 CPU core
• fix: TCDRWriter::Execute intern Stack overflow
• fix: catch on StartProgramExA (6) Access violation at address 00000000. Write of address 00000000
• fix: direct query API char escaping
• fix: security bug related enduser ip based authentication (affecting register requests only. call sessions were not affected)
• fix: 171 closed/resolved tickets from the last major release
• fix: many minor bug fixes

• new: detailed registrar statistics
• new: SIP load balancer integration and auto reconfiguration
• new: mmanage encrypted/compressed remote server connection
• new: server IP allocation from central location
• new: user mapping module (when user URI have to be rewritten before forwarding)
• new: rerouting by protocol
• new: SQLCPUUsage if local server, then check the process cpu load
• new: integrated mservicehost
• new: native SQL client option (SQLNCLI instead of SQLOLEDB)
• new: auto OS and DB optimizations and fine-tune for VoIP purposes
• new: offline messaging caching/queuing support
• new: full support for push notifications (both FCM and Apple)
• new: responsive design for the web control panel
• new: remote access from MManage
• new: configurable days after to delete from backup db
• new: users and enduser -> webportal connect as user
• new: server console with built-in API list
• new: option to disable disconnect/exit on timeout
• new: mmanage check if server domain resolves to server IP
• new: config wizard allow fork config
• new: "version" command
• new: webportal auto login from mmanage
• new: SMS module with flexible external HTTP API handling
• new: asendsms API
• new: ability to send SMS with external app (launch exe instead of URL)
• new: SMS verify (separate request with a delay. will just modify existing CDR)
• new: SMS did number lookup
• new: mmanage settings cache (speed up mmanage load by caching the last tb_settings load)
• new: webportal softphone/webphone integration
• new: webportal new settings for endusers such as voicemail config, conference rooms, etc
• new: softphone contact list integration: savecontactlist API, publiccontacts
• new: allow user auth also with the TelNumber field (instead of the Username)
• new: api command to rebuild all clients
• new: supervisor API and protocol encryption
• new: auto-retry call to alt number
• new: SIP message quick rewrite module (at transport level)
• new: allowoutboundrouting config (disable routing to outbound if push only)
• new: apply tb_firewalls also for webrtc traffic (at socket level)
• new: separate command for loglevel reload
• new: sendcpcparameter setting set to ";cpc=ordinary" if needed
• new: sendsipuseruripartfrom setting 0=no,1=for telnumbers to outbound sip servers in sip header,2=also in to header, 3=always (add user=phone such as sip:+12125551212@gateway.com;user=phone)
• new: usesrvrecords setting 0: no, 1: on low load, 2: yes
• new: handledtmfevent setting enable dtmf based features such as transfer or call. 0: no, 1: auto, 2: always, 3: even while ringing (set to 2 if to handle dtmf in gateways)
• new: allowanonymouscaller setting routing from anonymous; -1: autoguess, 0=no (default recommended), 1=yes allow anonymous to call free access ivr, local endusers or only for freeaccessuserid if system load is low, 2=allow also if load is high, 3=allow to call anywhere (never recommended)
• new: openregistrar setting accept blind registrations. 0=no,1=yes with any username/pwd,2=yes from low level
• new: blindauthentication setting openrelay; 0=no (def, reccommended), 1=yes (never recommended) //set a def user or modify v_checkuser to don't check the password
• new: freeaccessnumber setting anonymous access to a user; this number/username can be called freely (will map to freeaccessuserid)
• new: GetOtherPartyDisplayName and GetBestDisplayname
• new: multithread optimized memory manager
• new: mmanage basic network list the secure port
• new: server side nat keepalive packets
• new: fax module setting
• new: configurable user=phone in the URI
• new: reguser from admin command
• new: nonce based API pre-auth
• new: remoteadmin setting: if 0 then don't allow remote admin socket or admin API connections (only from 127.0.0.1/localhost/localiplist)
• new: compress and encrypt remote sql traffic
• new: built-in SQL management
• new: enable/disable presence from config wizard: fwdsubscribe/cfg_routepresence/fwdsubscribeoutbound
• new: local number portability auto import ported numbers
• new: auto socket reuse
• new: option to require registration: autocreatereguser = 2
• new: srvinstance filter for logs
• new: directions cache (speedup direction lookup by 60X)
• new: ability to run commands on all servers (especially important for the reload command)
• new: autoprovisioning based on IP address (on new user, the server could remember the last ip/username/password)
• improved: major upgrade for the SIP and media stack with all the cumulative updates since 2018 June
• improved: multi-instances send admin command to all servers (apply now)
• improved: SMS send from separate thread
• improved: normalize phone numbers for sendsms
• improved: voicebackupdir handling
• improved: Identity header handling
• improved: secure/tunnel HTTPGet: separate function for SetTunnelURL(StringEx oldurl)
• improved: tls processing speedup with optimal schduling, prirotization and threadpoll
• improved: mmanage menu reorganization
• improved: MManage user interface optimizations and simplifications
• improved: config wizard applying takes too much time (appliing the convertowebrtcgw. should be done only once)
• improved: mmanage warn if record was not saved on reload/close (user, rule, etc)
• improved: MoveFile handle out of space errors
• improved: call transfer related changes
• improved: tunneling optimizations
• improved: subnet verifications
• improved: log first fcm fail if there was no success before
• improved: sms receive (handle incoming sms); smsreceive API
• improved: default max log days depends on available space
• improved: fast internal hashmap for long numbers
• improved: better handling wrong sql connection string and bind IP at startup
• improved: auto setup optimal file permissions
• improved: do not resend INVITE on high load
• improved: last CDR id lookup failback to SELECT CAST(SCOPE_IDENTITY() AS INT) AS LAST_IDENTITY
• improved: skip using well known TCP/UDP ports for RTP and TURN purposes
• improved: allowforkforsignaling and usephisicalfromaddr default config
• improved: do not allow server/traffic sender auth ip to users transip
• improved: subsequentwrongcalls skip if not from same ip
• improved: new completely different tunnel keys (auto regenerate)
• improved: sqlupdater exec queries separately on fail if load is not too high
• improved: nonce per remote peer to avoid duplicate udp bug
• improved: log management (maxlogsperday, maxlogpermin, maxloglisten)
• improved: mmmanage search
• improved: mmanage push notifications enabled by default
• improved: socket level integrity checks
• improved: config wizard add directories: make it faster by submitting multiple sql at the same time
• improved: don't disable connectivity check on the config wizard if public ip (still can be blocked by external firewall)
• improved: addrlist: delete from addrlist then main addresses from other accounts
• improved: change upper server address should check also tb_users (remove old addr transip/addlist, etc)
• improved: called party username to the log
• improved: auto slow down status db storage on high refequency and/or too high db load
• improced: remove upper server registration if push notifications fails
• improved: def caching depends on available memory and local/remote sql
• improved: better calc pushnotification_persists (take in consideration the previous server load ...or the current server log!)
• improved: banning a valid SIP server entry should be avoided
• improved: don't start new cdr thread if terminating or allow new query on terminated
• improved: gracefully termination for endpoints (from the mainlogic thread)
• improved: set io priority for mserver to low (especially for the logger)
• improved: forwardonnoanswer is not good. we should just resend the same invite to the same destination with changed called number and changed call-id
• improved: auto separate the apple push certificate (pushkitYOURPACKAGENAME.pem)
• fix: tls high CPU load
• fix: socket objinfo related crash
• fix: old wrong logpath (and other wrong settings) are loaded from the mizuserver.ini file (and handle log can't create)
• fix: call WSAStartup and WSACleanup only once
• fix: tunnel answer for initpacket not received
• fix: catch on GetCfgValSetDef The statement has been terminated. String or binary data would be truncated
• fix: mmanage config right click popups doesn't work if we have a search filter already set
• fix: mmanage: if(NODE <= 0 && frmMain->mainnode > 0) -> WARNING, you should select a node first (View menu -> Node)
• fix: mmanage wizard don't add IP as comment
• fix: media form playback in mmanage
• fix: don't update PRManager if no PRManager used at all in the system
• fix: cannot find sip server user by id
• fix: logreset doesn't restore old loglevel if it was lovered
• fix: mmnage routing and rules unsaved record popup on close
• fix: server not disconnecting calls on close in some circumstances
• fix: auto HTTP API handling in server console
• fix: changing the MAC alone invalidates the license
• fix: invalid sms parameters
• fix: too much dns query from TFastReg::LoadAuthIPList
• fix: parser error Access violation at address 00D06CAD
• fix: catch on Message::GetParameters (14) Access violation at address 00D06CAD
• fix: multi-node setup network basic doesn't work because missing
• fix: mmanage doesn't detected that port 80 is already occupied
• fix: mmanage server console if API htto req returns or DOCTYPE HTML then retry on tcp
• fix: remove old firewall rule before adding new
• fix: ERROR: on optimize User 'xxx' does not have permission to run DBCC freeproccache
• fix: fax auth session mismatch

• new: auto normalize voice files (now it is possible to upload wav or mp3 in any format)
• new: wide-band conference calls
• new: offline chat with delayed send option
• new: mapped users option to remap username/passwords
• new: routing for SIP session timers
• new: transfer and conference for webrtc
• new: send heartbeat and load statistics to load balancer (when load balancer is deployed)
• new: conference and transcoding support for OPUS wide-band
• new: enable dynamic daily credit limits by default: Dynamic credit/spent limits
• new: option to restrict API access only for authenticated SIP user
• new: support for push notifications for by mobile and web clients
• new: volume normalization for call recording
• new: fastblacklist
• new: auto routinglist maintenance
• new: call recording download from web and voicerecdownload API
• new: flash RTMP protocol implementation (compatibility with legacy flash phones)
• new: fastpath processing of non-critical communication for local endusers (presence, chat and others)
• new: strict cutoff of multiple calls on depleted credit
• new: option to turn on db delayed durability
• new: default index fill factor optimal value
• new: optimal default values for query timeout and query governor
• new: nonce digest preauth option for the API
• new: date correlation optimization enabled
• new: options for groups to restrict calls to other groups
• new: quick wsload answer
• new: sql engine parameterization forced by default
• new: configurable timeout per socket type
• new: support for one time passwords
• new: option to always enable/disable nagle
• new: wizard: route UA to UA calls through upper server tunnel setting also for webrtc
• new: wizard: on WebRTC select first, also select tls,stun and turn and maybe also rtmp
• new: config for allowupperserverselection, selfforwardto, replace, replacecalleroncalls
• new: web control panel optimized for mobile screens
• new: a new responsive web interface (optional, not set as default yet)
• new: mmanage: log search from-to
• new: mmanage if remote server, start sql studio with login parameters
• new: mmanage "how to connect" webrtc client examples
• new: mmanage: dashboard
• new: mmanage cdr search by caller number, called number, etc (until found)
• new: mapped users option to remap username/passwords (autocreatereguser=0 forwardauthpassword=4)
• new: registrations based on routing (routingforregister)
• new: endpointlist add req count and check long messages
• new: stat from packet loss, detect high packet delay on rtp recv
• new: tunneling tcp double encrypt with different encryption methods
• new: re-register with new nonce asked by server
• new: sbc user import
• new: scheduled task email send action
• new: config.viarecip
• new: insert our own TCP and UDP candidate (so it will work in all circumstances, regardless of NAT/firewall)
• new: new users and devices form (the old one is still present)
• new: auto encrypt the emailpassword in settings and all the other sensitive data
• new: handle service pause (don't accept new requests)
• new: auto detect if dir cache if needed (def setting is "auto" -1)
• new: option to forward to other server if domain is not ours
• new: apply upgrades checked by default
• new: cache tb_directions in local file
• new: tb_direction cache invalidations. also add dircacheclear command
• new: maxcostperip limitation for webrtc calls (previously only for SIP and H.323 was supported)
• new: full API coverage for a usual web portal
• new: nonce based authentication for the API with preauth
• new: integrated static code analyzer into the release build server
• new: configurable dtmf duration
• new: fast stun answer packets
• improved: multiple-registrations per device and call fork
• improved: number portability extra fields
• improved: WebRTC TCO failback improvements
• improved: auto disconnect old/wrong sockets on too many connections
• improved: usage TRUNCATE TABLE instead of delete from when need to del all records
• improved: strongauth should check whether we are on local lan and if user prepaid with 0 credit
• improved: rtpwrite first: not for sip servers/traffic senders with public ip
• improved: various ivr related optimizations
• improved: a long list of minor improvements for the admin user interface
• improved: fine-tune default process priority and cpu affinity
• improved: optimize iocp thread count
• improved: API check max auth fail. use salt with md5
• improved: API call from server console: route commands from console to api if needed (so we can use also the api from the mmanage server console)
• improved: ExtractBestSDPAddress
• improved: using fast hashmap for all lists with pressure
• improved: detect low disk space on backup
• improved: count also webrtc to webrtc calls
• improved: external SIP IP. set it explicitly if you have more ip assigned to the server or server is on internal network to help NAT traversal for clients
• improved: don't handle binding request for channles -> route it to other end... so need a separate udp socket for the channels
• improved: handle 00/countrycode for routing and billing (separate function with replace callednumber)
• improved: rewrite c=ip sent by internal pbx module to public ip (also for rtcp and candidate). also rewrite sdp from client to client fromip
• improved: upgrade tlsproxy ssl libs
• improved: forward disconnect reason from upper ep
• improved: speaking status also for traffic sender if load is low
• improved: parse correctly (remove) ttl from sdp: c=IN IP4 224.2.1.1/127
• improved: add server to firewall exception list via the installer
• improved: call recording quality
• improved: unattended transfer handling
• improved: sbc connectivity check to TCP 80
• improved: various improvements for the built-in ftp service
• improved: auto add lego to firewall exception list
• improved: auto set def db recovery model to simple (one time)
• improved: simultaneous calls billing cut-off
• improved: performance: lock free unordered_map and queue
• improved: performance: replace TList with TListEx if threads might be involved
• improved: performance: Take advantage of Pack(false) and SetNull(idx)
• improved: performance: use TSQueue more (AddData() already checked)
• improved: performance: use KSL more (check for TListEx), replace eplist with KSL
• improved: performance: optimize thread sleep times (make it dynamic based on current load)
• improved: performance: reput removed optimization to string ex (thread local storage, strinextoint intostringex optimizations) add back IntToStrEx optimization with local thread string
• improved: performance: stringex optimization. don't pass by value and avoid memory copy at all cost
• improved: performance: delete endpoints earlier on high memory usage
• improved: performance: create multiple tempdb with equal size to avoid locks
• improved: performance: review MAXDOP (Add config option: 0=no set, 1=auto (set to cpucount/8) for reports, 2=always set to 1
• improved: performance: auto fine-tune sql db based on expected load, settings and your hardware config
• improved: performance: turnudpserverclients and turnchannels mutex (replace with lock safe list)
• improved: performance: kqueue: 2 separate list: one for add, other for get
• improved: performance: tsqueue with multiple list trick (increase size only at consumer thread)
• improved: performance: real mutex for tlist (for data and capacity change with double check). also min capacity should be 8
• improved: performance: make StringEx more thread friendly
• improved: set sql server compatibility level COMPATIBILITY_LEVEL 140 has been added
• improved: TSQueue instant increase capacity on subs add (but no read)
• improved: add single upper server to trusted ip list
• improved: don't check port bind if server is running (Accessport)
• improved: analyze: warn if webrtc is enabled but no domain or no ssl cert
• improved: tunneling send extra fake packets: either small or big
• improved: auto delete sipcallid callid and billingentry from old cdrs
• improved: handling of Reason-Phrase: Bad Via header
• improved: unreg handling from client
• improved: can't create new user if we already have a user with the same username but disabled
• improved: same cseq received
• improved: better target user auto-detect with multiple devices and call fork
• improved: forget old ip/port on unregister
• improved: selftest algorithms
• improved: auto convert ivr/announcement wav/mp3 files
• improved: don't send keepalive multiple time to same ip:port
• improved: tunnel random double send rtp
• improved: when checking if user is offline, check also huser status
• improved: hide callcenter and simplatform settings for compact db
• improved: send x-reason disc clause (so it will be passed to the client)
• improved: handling of expires: 1 (min expires accepted by asterisk)
• improved: a long list of other server and admin client improvements
• improved: tunnel: remove read-write sockets rwmode
• improved: tunnel: state lookup after ep for rtp
• improved: set builddynamicblacklist for gateways
• improved: socket max bandwidth management
• improved: respond to keepalive also on tcp
• improved: presence routing
• improved: SIP session timer re-invite routing
• improved: reset some parameters if server path was changed (for example the firewallopened flag and others)
• improved: if no restart (except daily), no info and no logrename commands for 5 days, then set back the loglevel to 5 from 9 (if it was 9 continuously)
• improved: update mestate for fastuser even if not authenticated
• improved: update all portmaps on new user succ auth
• improved: sql performance
• improved: string ex free list garbage collect
• improved: selfcheck and maintenance should wait until config::loasasync is finished
• improved: remove non authenticated calls from cdr's by default. if duration = 0 and caller was not authenticated
• improved: disable lcr if billing is not set
• improved: reput all old optimizations stringextoint
• improved: increase max router thread count
• improved: contact line normalization
• improved: load balancer auto-detect and integration
• improved: various webportal improvements
• improved: wait for successful child process creation
• improved: lots of improvements for the config wizard
• improved: a long list of other minor improvements mostly related to the core, sip and media stack, API and database caching
• fix: critical bug fix for tunneling recv buffer overrun
• fix: codec transcoding bug fixes
• fix: storeonlinestatus not called on subsequent register
• fix: forwarding failed reg answer
• fix: ExtractBestSDPAddress ...check config.mainaport
• fix: don't send so much echo on readonly sockets
• fix: mencryption: don't lookup after ip if sender also from the same ip (check sender contact field)
• fix: new portmaps are not created for user on succ connect (create new portmap mestate)
• fix: thread local storage can't be used if string is deleted from another thread
• fix: mserver process/handle count limitation
• fix: other tunnel related fixes (check outbound audio over tcp and silencesupress)
• fix: max accept event slot reached
• fix: branch and to tag for call fork
• fix: mmanage don't run "Cash-flow analysis" if no pricing is set
• fix: mmanage if localip not visible, then null it
• fix: mmanage block not billed calls should be deselected by default
• fix: mmanage don't try to autoset autossl if server is on local lan
• fix: mmanage: no call match your search criteria "quick filter"
• fix: mmanage false alert: cannot bind to tcp:80 ...maybe because ivp6
• fix: mmanage freezing on network page next
• fix: mmanage freezing on close (hung on UnInitializeSession sslogclient->Socket->SendText)
• fix: mmanage how to connect should list private ip if nataddrtype is 2
• fix: mmanage search users for IP doesn't work
• fix: mmanage load the correct consolekey if multiple nodes
• fix: contact address in requests
• fix: refer handling
• fix: close log in selfrestart
• fix: remove MEMORYCHECKEXT from release version
• fix: PacketIsSTUN
• fix: verify STUN_ATTRIBUTE_XOR_PEER_ADDRESS (allow only myip)
• fix: mainaportudp can remain the def if mainaport is not 80
• fix: cannot restart hosted servers
• fix: don't retry failed dns requests
• fix: not enough calls (3 5) disconnecting running calls. should be much more moderate if there are any pending connected not too old call
• fix: don't create empty settings for nodes with # (or create only from main node without #)
• fix: block too many not connected calls from trial server
• fix: server TCP register and incoming calls
• fix: catch on CheckScheduledtasks
• fix: catch on GetCallDirectionPlain
• fix: catch on HostToIpExFromCache
• fix: max accept event slot reached 64 (using separate tcp client groups)
• fix: Invalid column name 'sipcallid'.. sqltxt is: insert into mserver_backup.dbo.tb_cdrresellers
• fix: 70+ other bug fix for the server and for the admin client

• SIP processing optimizations
• voicemail module complete rewrite
• text to speech for the IVR via espeak and tts (available in multiple languages)
• chat improvements (improved reliablity for all paths)
• MManage config wizard: numerous improvements for NAT auto configuration, port availability and others
• A long list of WebRTC related improvements
• voice call recording upload related bug fixes
• improvements for TURN and tcp candidates so it is capable to route the media even if UDP is completely blocked for the clients
• offline chat: offlinechat global config (0=no,1=yes best effort,2=always must ), maxofflinemessagesperuser (def 100), AddOffMessage, XOFFMESSAGE, apiex: presence_set, [PRESENCE chat
• sms auth (otp): global config uauthverify: 1=sms,2=email code,3=email link; newuseru api
• improved datbase/sql performance (optional delayed durability, parameterization: forced, default index fill factor optimal value, optimal def value for max paralelism and degree)
• presence improvements (PUBLISH/SUBSCRIBE/NOTIFY, now also between WebRTC and SIP)
• mapped users feature (forwardauthpassword=4); option to remap username/passwords
• add command to list sapi numbers
• send X-Reason header to certain devices
• better NAT handling (server behind NAT now works in all circumstances)
• improved rtp candidates: CandidateRewriteForClientPublic
• fastblacklist, blacklist cache (blacklist_cache config) 
• improved security for local calls (allow only to local user if no reason for outbound)
• call recording on/off during call
• improved TLS tunneling
• socket timeout and linger configurations
• registrar reports
• recorded voice playback related fixes
• fix ftp service availability for recorded voice download (with strong random password auto changing every day)
• warn if ftp is not enabled and want to download file
• mmanage block not billed calls should be deselected by default
• option for blind accept registrations: openregistrar, blindauthentication
• improved auto thread prioritization and cpu affinity (cpuaffinity, autocpuaffinity)
• bug fix: mmanage freezing on network page next
• strongauth should check wether we are on local lan and if user prepaid with 0 credit
• rtpwrite first: not for sip servers/traffic senders with public ip
• if nat is configured for internet only, still accept from the local pbx service
• auto enable on firewall port tcpcandidatesrvport  (TCP relay port, 10080 by default)
• mmanage usage TRUNCATE TABLE instead of delete from when need to del all records
• improved selfrestart process
• improved exception and bug reports
• refresh letsencrypt cert 20 days before expire to avoid notification emails
• call recording for webrtc to webrtc calls (force calls via sip server)
• higher timeout (readORwrite_timeout,thisreadORwrite_timeout) for other_tcp_client and main_tcp_server
• configurable timeout per socket type (so we can quickly increase for other_tcp_client and main_tcp_server if clients reports websocket disconnects when
• used with third party webrtc clients)
• optimized iocp thread count
• increase the 'min memory per query' option if your server has a lot of memory available
• pbx features in the server documentation (conf rooms and others)
• transfer and conference via webrtc (catch REFER if the other ep is on SIP)
• mmanage: dashboard improvements
• mmanage: open direct query in mssql
• uip and commong.GetLocalIp for public/private ip
• ExtractBestSDPAddress check config.mainaport
• fix callid -> sipcallid in sql script then run db cleanup and create new db backup
• traffic sender and sip server users were not created during config wizard
• fixed chat recording def config value
• selftestclient/selftestserver authentication mismatch
• performance option set cdrextra to 0
• perf: checkwithzerocredit 0
• hide currency settings for gateways on the localization tab (if no billing is enabled)
• mmanage: unified commonication option
• sbc module
• forwarding failed reg answer for gateways
• handle Reason-Phrase: Bad Via header
• TSQueue instant increase capacity on subs add (but no read)
• set sql server compatibility level COMPATIBILITY_LEVEL 140 has been added
• config wizard: option to set process priority and auto boost
• mmanage analyze improvements. better first start analyzis
• mmanage global config groups (Caller ID Settings and many others)
• mencryption: don't lookup after ip if sender also from the same ip (check sender contact field)
• tunnel: remove read-write sockets  rwmode
• keeprtpssrc
• bugfix: webrtc stun/turn/ice timeout problem
• auto detect country prefix and prefix the entered number with it if not entered by the user
• language parameter for NumberToFileList (and try with suffix)
• bugfix: double auth issue
• informix integration
• ivr load and store operations
• ivr better playback timing (and new ivrfiledelay,ivr_monoliticclock, ivr_fileplaythreadpriority config options)
• option to forward unknown sip headers (fwdextraheaders)
• sqlsetupscript insert into tb_langtest
• unreg fast user on wsunreg
• webrtc call forward 301/302
• webrtc gateway reject registration if incorrect username/password on the upper server
• lego with -new if cert already expired
• try pipe on tcp fail (sqlinstancename=\\.\pipe\MSSQL$SQLEXPRESS\sql\query)
• implemented delayed registration
• fastauth unregistration; fastauth re-reg/unreg with different username and auth username
• forwardauthpassword 4 send auth username as from username
• re-register with new nonce asked by server
• change def password encrypt
• handle service pause (don't accept new requests)
• speaking status also for traffic sender if load is low
• handle vcodec settings passed with wsuser (remove other codecs from the sdp)  
• don't retry failed dns requests: EVENT, dnsresolver failed for x.y.z
• auto detect if dir cache if needed (def setting is "auto" -1)
• option to forward to other server if domain is not ours
• auto set db recovery model
• bugfix error max accept event slot reached 64 for turn
• upgrade tlsproxy ssl libs
• forward disconnect reason from upper ep
• better feedback from the server. disconnect reason of the last call
• performance improvements for Base64 encoder/decoder
• ESocketGroup vs ESocketType
• handle wsuser codec, prefcodec, proxyaddress
• change candidate priorities (firt should be the srflx raddr)
• rewrite audio port if mainaport and c=in ip is local  m=audio 18080 UDP/TLS/RTP/SAVPF 109 9 0 8 
• bugfix: 503 maximum calls in progress/Program just now started 
• esg_relay_tcp with nagle off and rtp sock like characteristics
• catch call forward from webrtc and send it directly to the caller
• update mestate for fastuser even if not authenticated
• new portmaps are not created for user on succ connect (create new portmap mestate): encryption initialized successfully for client, state obtained from
• portmaplistu
• tunnel: should prefer lookup after ip if ip:port lookup state is bad
• cdr filter for marker
• decrease loglevel if need to skip
• fixed ivr announcement cut at end
• add no answer for disc reasons
• more rate limiter options (maxcallsperday) 
• api: check max auth fail. use salt with md5
• call api from server console: route commands from console to api if needed (so we can use also the api from the mmanage server console)
• reguser from users and devices
• candidate ip's should be public
• esg_relay_tcp msocketgroup_tunnel_tcp  msocketgroup_relay_tcp
• use separate socket instead of esg_other_tcp_server
• nsert udp/tcp relay only if caller/called is not on the same lan
• for webrtc to webrtc calls use the same procedure (a=candidate:1004 1 tcp), just create a separate socket for the called (tcprelaysock2)
• check if need tcp candidate when we use udp candidate
• connected and receiving packets on tcprelay sock, but doesn't forward. check EVENT,xxxx checkfixroute
• add external simmetric port
• tunnel related fixes (check outbound audio over tcp and silencesupress; renilmohan@gmail.com)
• no stun is needed if media is from sip server in recv invite (invite from ip is same as media ip)
• reinvite with private media ip if no incoming audio with stun enabled (rtp stun public address found as)
• auto turn to tunneling only if not LAN voip server address
• handle SIP/2.0 423 Interval Too Brief (auto increase register ival)
• store equipment field for sip servers
• stringex free list garbage collect
• node setting for tb_settings
• disk io statistics to dashboard, sql reports
• maxdailycredit for users and devices
• thread local storage can't be used if string is deleted from another thread!
• introduce strictrtp setting for router forwarded port networks
• block too many not connected calls from trial servers
• set the resellerbilling to 1 if def is -1 and there are reseller users with valid childs
• disable lcr if billing is not set  
• remove api ERROR from the logs: api sending answer ERROR: no user parameter NORETRY  
• enable dynamic daily credit limits by default: Dynamic credit/spent limits
• don't lookup dir/spid for internsational calls
• limit message count after BYE and/or after state finishing
• baseobjects, tsqueue memory cheks
• add memory checks
• improved string performance 
• packetloss statistics
• STR_MEMORYPOOL also for char data
• switch to the new fast concurrent hash map
• add crc protection for all objects passed via TSQueue
• contact line normalization
• auto dtmf conversion when neccessary
• set pcmu/pcma codec only if recording needed
• various performance related improvements (mserver optimizations and speedup)
• replace TList with TListEx if threads might be involved
• replace heavily used TStringListEx with KSL especially if threads might be involved
• optimize thread sleep times (dynamic TSLEEPTIME based on usage)
• stringex optimisation. don't pass by value. strtointex; avoid memory copy whenever possible
• auto finetune mssql based on expected load, settings and hardware config
• tsqueue with multiple list trick (increaze size only at consumer thread)

• SIP stack upgrade to v.4.2 with many encasements including better support for text messaging and SMS relay
• Full support for WebRTC including DTLS/SRTP RTP relay and built-in STUN and TURN servers
• New PBX features such as conference rooms and voicemail
• Click to call and call me features
• Better softphone integration including features like P2P, callback, sign-up, payments and many others
• Updated server configuration wizard with network/nat/localization auto-detection
• Intelligent RTP offloading capabilities (peer to peer media whenever possible)
• Auto codec transcoding when necessary
• Auto SSL certificate for HTTPS/WSS
• Improved built-in VoIP tunneling and encryption (optional module)
• Security enhancements
• Improved voip admin client
• A new release for our customizable softphones
• Our VoIP server hosting service (SaaS ) was also refreshed with all the new features
• A special edition focusing on small business needs (company local PBX) is also available: https://www.mizu-voip.com/Software/WindowsPBX.aspx
• A robust SIP load balancer capable for millions of calls on a cheap hardware
• wsuser handle needregister=false and proxyserver
• sip upper authentication with sip username/password
• cache console login string at ServerData
• fix router routerequestevent ret with error
• speedup wsload answer
• tcp/udp turn with insertservercandidate=1 with firewall blocking between nat's
• turn with turnmultipleuppersock set to -1 (both udp and tcp turn)
• remove turnmutex (no more shared variables) 
• don't answer stun for local (not routed) rtp sockets
• add more fields for tb_loadstatistics: dbload
• fixed auto currency conversion  
• use the Error-Info: header
• mserver respond for OPTIONS as described in https://en.wikipedia.org/wiki/Cross-origin_resource_sharing
• check sql injecections also in QuotedStrEx; filter out also xssl attack
• check file transfer no exec right for stored files
• webportal and others log to logs subfolder if present
• don't write cdr record if caller was not authenticated and 0 len duration  
• max bandwidth mbits in msocket (rate limiter)
• newuser private/public api
• on subsequent error for cdr thread spawn new thread
• auto codec convert on 488 Not acceptable here
• a bounch of new pbx features for small business
• dial a group (for to all numbers to group)
• optimize for fast lookup (cache all after mainlogic fully started)  
• auto detect nat ip handling from source address; auto nat handling allow ip (check from where it is registering); autodetect if rtp routing is required (auto call a public ivr and check incoming rtp)
• detect if the 2 peers are behind the same NAT and don't route media in this case (if they have the same from IP)  
• rtpwritefirst  ...only for private ip's  ..not to servers
• don't auto allocate common ports such as 3389 (rdp) and others for any purpose
• don't accept nonce from client for authentication
• data path (no write access to bin path); test mmanage (and mserver) on restrictive account (no write access to program files); write rights to program folder (recheck all apps to use a writeable data folder); test mmanage (and mserver) on restrictive account (no write access to program files)
• don't normalizedports for existing servers
• generate stunaddress according to mainaportudp
• enable "TLS" if "AutoSSL" is selected
• auto ssl: ask for email and don't popup domain missing on wizard start
• tlsproxy autodetect listenipex (like in mmanage config wizard). also pass listenipex from config    
• add max call time and ring time options
• remove mobile prefixes from config wizard. add "block not billed calls"
• expected load finetune already running server (make sure to don't break anything)
• don't handle bindig request for channles -> route it to other end... so need a separate udp socket for the channels
• insert our own candidate and forward the media
• maintain a list of turn objects associated with mainudpsocket
• auto delete turnchannels not used for more then 15 minutes
• fix no audio for webrtc to webrtc in firefox
• handle candidate in endpoint (send media there)
• handle websocket recv too much (should call process again)
• voicerecupload implemented also on server side
• speex codec convert on SIP/2.0 488 Not Acceptable Here
• fix webrtc chat not working: webrtc to SIP, webrtc to webrtc
• don't send keepalive multiple time to same ip:port
• storeonlinestatus not called on subsequent register
• when checking if user is offline, check also huser status
• load server configuration and check coherence
• check if mainaport is free. also might check if ports are reacheable behind nat.
• on server launch, make sure that it was actually started and running correctly; load and show last errors, warnings from startup 
• STUN_ATTRIBUTE_XOR_PEER_ADDRESS port mismatch
• refresh lifetime can be omitted. handle zero unsubscribe
• the REQUESTED-TRANSPORT must be UDP otherwise reject with 442 (Unsupported Transport Protocol) 
• reject if STUN_ATTRIBUTE_REQUESTED_TRANSPORT is not UDP    442 (Unsupported Transport Address)
• implement turn touple: transid, channel number, permission, peer address
• implement turn service
• stun allocations REQUESTED-TRANSPORT LIFETIME  DONT-FRAGMENT  EVEN-PORT RESERVATION-TOKEN; answer: XOR-RELAYED-
• ADDRESS (bind address), LIFETIME, RESERVATION-TOKEN, XOR-MAPPED-ADDRESS (client tcp address)
• bugfix: Invalid column name 'maxdailycredit'
• routing numportcache
• mlbhandling, numport, directions
• remove block satellite calls if forward (tunnel or webrtc)
• improved tls proxy high load behavior
• Via processing for proxies is described in Section 16.6 Item 8 and Section 16.7 Item 3.
• X-MSL: 0  ...keep only highest
• send all chat messages without the to tag (also subscribes, etc)
• don't set fromip/fromport on msg rec (because addr might be wrong)
• transportip/port mismatch (port from recvport but ip from orig db transpip)
• asyncDecreaseCredit  ...don't set if too big value and is traffic sender
• don't disconnect websocket from api (for wsuser)
• set parentid to -1 where not valid daily task
• common.binpath+config.CFGSOUNDSDIR
• config.voicebackupdir
• smartblacklist, optimize prefixlookup
• auto close mmanage after some idle time
• remove v_dialplan upgrade  (create procedure ...) from the sp upgrade
• number portability: link to global config
• set strongdigestauth to 0 if tunnel or gateway
• CopyDirectoryEx improvements
• send presence in sip message or webrtc message sendim/SendMessage()
• increase webrtc priority if address is set
• spid import tool
• registry FastSendDatagramThreshold set to 1400 (default is 1024)
• 3GB option; change internal mem checks to allow 4 GB (or 3 GB on 32 bit systems)
• fastreg module: a very fast sip registrar
• encrypt all passwords in global config
• secure websocket (tlsproxy)
• public universal API accessible via UDP/TCP/HTTP
• mlb: one click test, hsl->msl, long lists, meassure
• BRS form allow to change fields
• brs/lcr lower priority entry
• inverse billing after caller number prefix. remote-party-id
• major upgrade for tunneling and encryption modules
• server load-balancer and failover re-design
• Send RTCP BYE
• extra data for ep at separate class pointer
• keep rtp_ssrc between fileplayer sessions
• fix tunnel related issues 
• delete unneded files from distributables
• ios tcp tunnel to alternatelocaport
• msg queue check should depend on simultaneous call count
• add msg/sec stat for statdump
• blockselfcall allowselfcall  ...to allow call self mobile
• review free ammount check at v_getprice
• websockets (parse and forward from port 80) and embedded webserver 
• skipped media packet on client call not connet ....if we remove this, ring is working fine
• dont send with rtpwritefirst if rtp already receiving
• cdr comment field
• create rtp from main thread only (not on rtp receive)
• check file name on http (remove special chars)
• cer and msi, asp, aspx mime types
• msocket mutext optimisation
• fix server stability (quickrtp/quickforward and others) 
• detect if selftest is not running
• selftest should detect if no incoming messages (quickly)
• warn if too much time spent on ep->timer
• improved ReleaseLocks
• mlist lazy lock
• fix pack is not called after delete for socketprocessers and others 
• implement rating in APIEX
• tests with constant high load
• tls proxy cpu limit
• statistics by equipment
• add tcp to contact header
• upper server rewrite on server side: rewrite incoming / outgoing sip message: ip/condition/rewritefrom/rewriteto
• video stream routing improvements
• tunnel connectivity fixes (sometime doesn't connect with simple udp)
• fix too many query get and release
• p2ppeerpublicport and others for unpaused sockets
• IVR rerouting
• fine-tune HasOverlappedIoCompleted
• fix msockdata ovl pending on dtor. mem leak.
• better socket stats at call end (and retest with udp)
• check tcp sockets (same reconnect count as for udp)
• cfg_extraprotect 2
• display warning on readonly writeonly sockets: sending on wrong socket
• auto increase packet per frame on high server load

• All cumulative updates since 2015 January
• Kernel space RTP routing
• General security review and a separate documentation for security related topics
• Other improvements include fast registrar optimizations, auto-tuning depending on server load and various micro optimizations.
• new MizuManage
• WebRTC support
• API over websocket and json
• fast route path
• a new mailer client
• smart blacklist
• MNP improvements
• new tunneling module
• built-in webserver
• RTMP/Flash support
• improved IVR and billing
• The new "universal" API for the Mizutech VoIP server is available now to help you with any VoIP integration tasks. Works over UDP, TCP, HTTP, SIP and SMS
• X-CDDT: send only if p2p path not found yet (p2p rtp path final found)
• fastencryptkey
• don't set alternatelocalports to autoprovisioning encv2
• HTTPReq ...use also via udp via our server if our server and encryption is used (the server should
• cache the answers if no query parameters)
• RTP sent 338 (p2p: 0 sdp)  add also rec: (recvfromport)
• no more presence requests if no response for subscribe
• win 8.1 issues (copy files to system32)
• chat and presence with and without tunnel (quick signaling routing) 
• handle from config wizard: standalone tunnel server should not set port limits
• userautoaddwithowner: wether server should auto create resellers. -1=auto,0=no,1=yes, 3=must  
• max capacity specifiy also from mmanage branding
• from server auto add upper server, set also the ccenddate
• oldalternatelocalports. listen on old udp/tcp sockets. env2 variable in msocket should also consider this
• onlyoldtunnel variable to msocket (oldalternateserverudp) and endpoint (route from TCPTransport)
• server to handle both old and new tunnel
• check if rtp is routed after session progess (don't wait for connected)
• shared did full implementation
• fix SIP/2.0 400 Malformed topmost Via header
• report server failed backup
• set senddailyemail = 1 for new owners
• disable calls between users if different domain (different company)
• check tunnel max line limit (alert treshold)
• use pipe when possible for db connection
• CursorLocation new TMSQuery  Provider=SQLOLEDB
• pooling / singleconnection per thread; MaxPoolSize and singleconnection per thread
• auto convert subqueries to left joins
• fix disconnecting because forced: no media detected clnt 
• auto refresh dns lookup cache 
• fix rtp port range bug
• fix RecordThread: buffer overflow
• fix socket bandwidth limit reached
• add mutex for tsqueue
• Common::GetCPUTime() 
• tests and improvements with slow db connection 
• fix missing rtp sent statistics in the cdr
• GetFreeLocalSDPPort is slow. avoid it
• disable rtcp, fileplay and maintanance on high query quee length / high load
• disable video rtp routing on high load
• don't block requests with Max-Forwards: 0 (gts OPTION)
• sipsever upper auth load username/password from the normal username/pwd fields if proxy authentication is not set
• set reroute to 0 if only 1 routing entry
• time-zone adjutements