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TekSIP Changelog

What's new in TekSIP 4.2.1.0

Sep 24, 2023
  • E-mail alerting feature is added.
  • RFC 5090 support is added with default draft-sterman-aaa-sip-00.txt method for RADIUS authentication.
  • HTTP REST API (JSON).

New in TekSIP 4.1.2.1 (Aug 8, 2019)

  • Route hunting [Requires SP license]

New in TekSIP 4.1.2.0 (Jul 12, 2019)

  • Route hunting [Requires SP license]

New in TekSIP 4.0.8.0 (Aug 10, 2018)

  • Routing has two sub sections; Destinations and Routes. You can have multiple route entries for a number prefix.

New in TekSIP 4.0.7.0 (Aug 10, 2018)

  • Media Encryption option in route profiles for outgoing calls when RTP proxy is enabled

New in TekSIP 4.0.6.0 (Aug 10, 2018)

  • IPv6 Support

New in TekSIP 4.0.4.0 (Feb 16, 2018)

  • IP Filters option is added to settings

New in TekSIP 4.0.3.0 (Feb 16, 2018)

  • Route capacity options is added to route profiles

New in TekSIP 4.0.2.0 (Feb 16, 2018)

  • SRTP <-> RTP Interworking in SP edition

New in TekSIP 4.0.1.0 (Jun 8, 2017)

  • .NET Framework 4.6.1 upgrade

New in TekSIP 3.9.0.0 (Mar 15, 2017)

  • You can specify a routing entry for a prefix which routes calls to a dynamically registered endpoint (Extension) in SP edition

New in TekSIP 3.8.1.0 (Oct 31, 2016)

  • A new routing entry type added. You can specify a routing entry for a prefix which routes calls to a dynamically registered endpoint (Extension) in SP edition

New in TekSIP 3.8.0.0 (Aug 22, 2016)

  • TekSIP can act as a WebRTC media proxy for SIP based WebRTC softphones. This enables WebRTC softphones to make calls to and accept calls from legacy SIP systems. Media proxy feature is available in SP edition of TekSIP and currently only audio supported

New in TekSIP 3.7.0.0 (Aug 22, 2016)

  • You can set “Private IP Addresses as Local” option to make TekSIP assume that all private IP address blocks, 10.0.0.0/8, 172.16.0.0/12 and 192.168.0.0/16 as local. This is useful if you use mixed private IP block behind a NAT gateway
  • Built in STUN server is enabled by default and cannot be disabled

New in TekSIP 3.6.9.0 (Jun 11, 2015)

  • TekSIP can act as SMPP Gateway. Instant messages sent by registered SIP endpoints can be sent as SMS through an SMPP gateway and received SMS' can be routed to registered SIP endpoints as SIP messages.

New in TekSIP 3.6.7.0 (Apr 30, 2015)

  • TekSIP can play a notification message when authorized time for a call is about to expire, Mid-Call Announcement

New in TekSIP 3.6.4.0 (Jan 30, 2015)

  • RFC 7118 WebSocket protocol support

New in TekSIP 3.6.1.0 (Nov 17, 2014)

  • Restriction of call recording in extension level
  • Restriction of dialing external destination in extension level

New in TekSIP 3.5.9.0 (Nov 17, 2014)

  • Music on Hold feature

New in TekSIP 3.5.8.0 (Nov 17, 2014)

  • Call pickup function

New in TekSIP 3.5.7.0 (Jun 20, 2014)

  • Call from multi-homed PC based Lync clients to TekSIP registered IP phones establish without audio (Version 3.5.7).
  • IP Phone PnP provisioning based on SUBSCRIBE / NOTIFY methods for Yealink and GrandStream IP phones. Fanvil, Snom and Aastra IP phones are also supported but not tested (Version 3.5.7).
  • SDP Manipulations (Version 3.5.6). Commercial editions only.
  • Lync Proxy (Version 3.5.6). Commercial editions only.
  • NTLM Authentication method support for upstream registration (Version 3.5.5).
  • RADIUS Disconnect Request (RFC 5176) (Version 3.5.1).
  • RADIUS Interim Update (Version 3.5.1).
  • TCP stack is redesigned (Version 3.4.9).
  • Built-in STUN server has been added (Version 3.4).
  • You can specify a FQDN (DynDNS address etc) as an external address; TekSIP will query FQDN every minute for possible IP address changes. (Version 3.4).
  • Prefix removal option has been added to route entries (Version 3.4).
  • TLS support been added (Version 3.3.2).
  • HTTP Interface has been added. (Version 3.3).
  • New performance monitors added. Please see manual for details (Version 3.2).
  • Web monitoring feature added (Version 3.2).
  • Endpoint registrations can be saved before re-start and restored after re-start (Version 3.2).
  • An option to keep endpoint passwords in clear text has been added, Settings / Authentication / Encrypt Passwords (Version 3.1).
  • Active sessions will be dropped when TekSIP is restarted (Version 3.1).
  • Specific user agents can be banned through Settings / Services / Banned SIP User Agents parameters. You can define more than one user agent by concatenating with semicolon “;” (Version 3.1).
  • You can select transport protocol which TekSIP uses using Settings / Services / SIP Transport (Version 3.1).
  • TekSIP monitors failed registration and call attempts from suspicious endpoints and blacklists them (Version 3.1). GUI Monitoring is available only in commercial editions.
  • TCP protocol support enhanced and protocol option has been added for routing entries (Version 3.0).
  • Log files are kept in \Logs directory and rotated daily (Version 2.9).
  • Version 2.9 introduces routing table enhancements; TekSIP can also register itself to provider SIP server if needed. You can receive incoming calls with registration. You can specify a separate domain name if needed for routing entries. TekSIP automatically reloads routing table if any update is made in the routing table; service re-start is not needed anymore in version 2.9. You have to use new TekSIP.mdb which comes with the distribution since table structure has been changed for SIP routes.
  • Version 2.8 introduces RTP proxy and audio recording features. RTP recording can be performed only for G.711 A and mu law calls. TekSIP will reject calls which does not use G.711 A and mu law codecs if audio recording is enabled.
  • You can limit maximum session duration by setting Settings / Services / Max. Session Duration parameter. Default value is 0 which disables limiting. You can set its value up to 24 hours in 1 hour steps.
  • Version 2.6 presents Presence Server ability (SIP/SIMPLE based).
  • Version 2.5 can re-direct calls to a voice mail system (like TekIVR) when a user is unavailable. Please see manual for details.
  • Version 2.4 can disconnect SIP sessions through TekSIP Manager’s Active Sessions tab.
  • Version 2.3 can detect external IP address if domain name entered in FQDN format. If TekSIP detects external IP address via this method, TekSIP automatically refreshes external IP address information querying SIP domain name. Refresh period is 300 seconds.
  • Version 2.2 introduces RADIUS Client functionality.
  • TekSIP does not authenticate inbound calls to registered endpoints from known gateways listed in routing tab in version 2.2.
  • This freeware version allows maximum 5 SIP endpoints registration and 5 simultaneous SIP sessions (Calls).
  • Endpoint definitions and routing table are stored in TekSIP.mdb.
  • If you change or remove listened IP address in your Windows Network configuration, TekSIP automatically selects first available IPv4 address in your network settings.
  • You may need to re-start TekSIP after first run if TekSIP is located behind an UPnP supported NAT gateway.

New in TekSIP 3.5.1.0 (Dec 27, 2013)

  • RADIUS Disconnect Request (RFC 5176)
  • RADIUS Interim Update

New in TekSIP 3.4.9.0 (Dec 27, 2013)

  • TCP stack is redesigned

New in TekSIP 3.4.6.0 (Mar 2, 2013)

  • Built-in STUN server has been added

New in TekSIP 3.4.4.0 (Dec 28, 2012)

  • HTTP interface improved

New in TekSIP 3.4.0.0 (Jul 18, 2012)

  • You can specify a FQDN (DynDNS address etc) as an external address; TekSIP will query FQDN every minute for possible IP address changes.
  • Prefix removal option has been added to route entries

New in TekSIP 3.3.2.0 (Mar 28, 2012)

  • TLS support been added.

New in TekSIP 3.3.0.0 (Nov 29, 2011)

  • HTTP Interface has been added

New in TekSIP 3.2.0.0 (Mar 30, 2011)

  • New performance monitors added. Please see manual for details
  • Web monitoring feature added
  • Endpoint registrations can be saved before re-start and restored after re-start

New in TekSIP 3.1.0.0 (Dec 7, 2010)

  • Specific user agents can be banned through Settings / Services / Banned SIP User Agents parameters. You can define more than one user agent by concatenating with semicolon “;”

New in TekSIP 3.0.0.0 (Nov 3, 2010)

  • TCP protocol support enhanced and protocol option has been added for routing

New in TekSIP 2.9.0.0 (Jul 7, 2010)

  • Introduces routing table enhancements; TekSIP can also register itself to provider SIP server if needed. You can receive incoming calls with registration. You can specify a separate domain name if needed for routing entries. TekSIP automatically reloads routing table if any update is made in the routing table; service re-start is not needed anymore in version 2.9. You have to use new TekSIP.mdb which comes with the distribution since table structure has been changed for SIP routes.

New in TekSIP 2.6.0.0 (Feb 12, 2010)

  • Presents Presence Server ability (SIP/SIMPLE based)

New in TekSIP 2.5.0.0 (Feb 12, 2010)

  • Can re-direct calls to a voice mail (like TekIVR) when a user is unavailable

New in TekSIP 2.4.0.0 (Feb 12, 2010)

  • Can disconnect SIP sessions through TekSIP Manager’s Active Sessions tab.

New in TekSIP 2.3.0.0 (Feb 12, 2010)

  • Can detect external IP address if domain name entered in FQDN format. If TekSIP detects external IP address via this method, TekSIP automatically refreshes external IP address information querying SIP domain name. Refresh period is 300 seconds.

New in TekSIP 2.2.0.0 (Feb 12, 2010)

  • Introduces RADIUS Client functionality.