WINDOWS THIN-CLIENT VOIP RECORDER
This recorder passively monitors all Ethernet VoIP traffic and silently records it. It will then upload each call recording directly to your personal CallN account via a secure 2048-bit SSL encrypted connection.
It records all SIP, H.323, Cisco SCCP, MGCP, NEC iSIP, LG iPECS, Broadsoft, Mitel, Avaya, Toshiba, Panasonic and ShoreTel calls.
LATEST VERSION: 1.32.0 – 19 JANUARY, 2021 (24.5MB)
Windows 10, 8.1, 8, 7 SP1, Vista and Windows Server 2008/R2, 2012/R2, 2016 (32-bit / 64-bit)
WHAT’S NEW IN THIS VERSION
Fixed – Built with .NET framework 4.8. Only support Win 7 SP1 and later operating systems.
New Feature – Sonus SIPREC call recording interface.
Fixed – Excluded detection of inband audio DTMF for NECISIP protocol which resolved an issue with passed keypresses not being correctly detected.
Fixed – Limit removal of duplicates by IP address to Panasonic handsets only. This resolved an issue with handsets not being identified and calls being discarded on LG iPECS.
Fixed – SIPREC Cisco include Extension Number into call metadata and detect call metadata from late TAPI for calls already in progress.
Fixed – Mitel event handler stops randomly.
New Feature – Support Mitel pool event.
New Feature – Support Mitel OIG Advanced License.
Fixed – SIPREC starts recording in middle of calls.
Fixed – Some missing calls in Avaya
Fixed – TAPI comes late
CHANGE LOG HISTORY
25 JUNE, 2019
- New Feature – Detect Cisco CUCM call metadata from TAPI driver.
- New Feature – Netsapiens call recording interface.
- New Feature – Added SIPREC endpoint ability to be set to local ip address.
- New Feature – If you create a registry key HKEY_LOCAL_MACHINESOFTWARECallNVoIPRecordingClientMaximumCallLengthSeconds as DWORD then any call exceeding this duration will be terminated and saved.
- Fixed – Call not recorded when RTP first matched via absolute address/port and then media changed to address not defined in SDP.
- Fixed – Remote linux executing ‘sudo command -v tcpdump’ via bash shell e.g. ‘sudo bash -c “command -v tcpdump”‘ to fix compatibility issues with some distributions.
- Fixed – Panasonic MGCP use ‘X-P/cctl(04)’ to detect hangup.
- Fixed – Shoretel phone numbers and call direction for new handset model.
- Fixed – Cisco CallManager active SIPREC listener will change call/destination parties when a full Australian phone number in international format is detected as the destination phone number.
- Fixed – Detect non-spec Asterisk SIP INFO application/dtmf-relay event content. e.g. ‘Signal=10 Duration=160’.
- Fixed – SIP 488 Not Acceptable Here Status Message received in response to SIP INFO dtmf-relay was tearing down call not just for INVITE message.
- Fixed – SIP INFO application/dtmf-relay was sometimes associated with the wrong call party.
- Fixed – SIP auto-answer calls which automatically change direction have issue with subsequent update SIP info and name direction.
- Fixed – Enhance registration process for newer Panasonic handsets. This would cause handset direction to be incorrect and a lot of ‘Anonymous’ calls.
- Changed – Set the default pcap filter to ‘not (greater 10240 or port 445 or port 139 or port 69)’. This will ignore packets larger than 10KB as well as SMB and TFTP packets which will help reduce traffic to be processed. e.g. when file servers are part of the capture stream.
- Changed – Changed STUN server for public ip detection to ‘stun1.l.google.com’.
MAR 23RD, 2018
- New Feature – Ability for remote capture from linux servers via ssh and the tcpdump utility.
- Changed – Limit the incoming packet queue length to 1GB per capture stream/adapter instead of the previous 100,000 packets.
- Fixed – Ignore SIP NOTIFY with appearance-uri when the phone number is empty. (e.g. don’t overwrite an existing phone number with this empty value).
- Fixed – LG iPECS call pickup with ** was not recorded without an existing call being present.
- Fixed – Log errors when archiving directly to a UNC path.
- Fixed – Possible race condition on servers with very very slow HDD write speeds resulting in ‘XML file is empty’ exception while uploading calls.
- Fixed – Sometimes for Avaya an outbound call dialled digits were included in the ‘caller keypresses’ metadata. Executing keypress rules etc.
- Fixed – Allow LG iPECS messages to be interpreted from TCP packets and not just the standard UDP ones.
- Fixed – When using local archive don’t add recording if the call audio is not uploaded to hosted account. e.g. handset disabled or recording is too short.
OCT 27TH, 2017
- New Feature – Monitor hard drives configured as the logging and archive paths to alarm when unavailable.
- Fixed – SIP status OK 200 to INVITE without an SDP would not capture call connected time and treat call as unanswered upon upload.
- Fixed – SIP Broadsoft *98 pickup for Anonymous call now switching parties and retrieving ‘Anonymous’.
- Fixed – Recorded calls with an 8 byte encrypted audio attachment containing a 0 byte payload are unable to be uploaded and block queue.
- Fixed – Shoretel, added other party phone detection at cursor position 35,173.
- Fixed – Remove possibility of upload thread permanently stopping upon any hard drive read failure.
- Fixed – Detect call pick-up type call on LG iPECS and reverse call parties.
- Fixed – Due to possible clipping resulting in less accurate transcriptions, revert ‘When normalising only count audio signal as a peak when the given percentile is below 99.9%. Average volume increase by about 4db’.
- Changed – When the log path (was app path) is less than 2GB (was 5GB) automatically turn off pcap logging when enabled.
AUG 23RD, 2017
- New Feature – Automatically retrieve setting from cloud service for audiobitrate.
- Fixed – High CPU usage on service when username/password not entered but machine rebooted after install.
- Fixed – Avaya 1600 series handset now recording correctly.
- Changed – When normalising only count audio signal as a peak when the given percentile is below 99.9%. Average volume increase by about 4db.
MAY 2ND, 2017
- Fixed – When building audio from RTP and a packets sequence number is sequential yet the time is unexpected and the packet doesn’t have mark flag enabled then don’t insert filler silence. Causes audio sync issue when decoding traffic from comvergence.com.au.
- Fixed – Fix for unknown phone number calls on NEC phone systems.
- Fixed – Not writing fragmented packets to tracefile log when ‘Write PCAP All’ was enabled.
- Fixed – Additional Avaya screen update message type 0xe3.
- Fixed – On initial install when entering an account/username an error is displayed “the authority / host could not be parsed”.
- Fixed – Better CLI detection on Avaya 9608 and 4610 handsets.
- Fixed – SIP handsets receiving multiple OKs to a single REGISTER will use existing user-agent if not provided.
- Fixed – Possible problem with SIP INFO message if it contains application/dtmf-relay with more than 1 keypress in ‘Signal=’ body content.
- Changed – Renamed temporary task request folder from ServerRequests to RecordingNodeTaskand also temporary task request folder from ServerResults to RecordingNodeTaskUpload.
- New Feature – NEC speed dial codes now supported.
- New Feature – Toshiba phone systems now supported.
- New Feature – Automatically detect handset registrations via the Cisco CallManager SIP PUBLISH message containing XML attachment.
- New Feature – Use the new CallN REST API v2 calls. This adds telephone platform and unanswered call capture.
JANUARY 20TH, 2017
- New Feature – If pcap tracing is turned on and the free hard drive space drops below 5GB then disable the trace.
- New Feature – Ignore detection of Bluetooth interface adapters, these can no longer be recorded.
- Fixed – Read the optional P-Preferred-Identity SIP Header for INVITE message for the proper caller party. As seen on NEC SIP trunk gateway device.
- Fixed – Cisco SCCP update tracked handset address from CallState On-Hold and Connected states. Previously could have caused calls transferred to a ringgroup to not be recorded.
OCTOBER 21ST, 2016
- Fixed – Allow whitespace characters in handset name registration for Avaya.
- Fixed – Correctly identify Panasonic handset MAC address.
- Fixed – Possible RTP sync issue after ‘Conn:’ from hold/dial screen sequence on Avaya.
- Fixed – Action Avaya disconnect message type 0x84 when received by handset and not just when sent by handset.
- Fixed – Avaya ‘Call:’ will now record if there was never a ‘Connected:’ state. These calls are just dial tone though.
- Fixed – Possible error log line written when decoding malformed SIP content section. Usually detected from backup software copying pcap files across LAN.
- Fixed – PCAP Syslog timestamp messages now use zero mac address and zero ip address to remove problem with some Bluetooth adapter traces.
- Changed – PCAP Syslog timestamp interval is now every 30 seconds.
23 AUGUST 2016
- New Feature – Add a syslog entry to the pcap trace files every 60 seconds indicating local time and machine name. This will help with aligning debug traces.
- New Feature – Record calls from Cisco Unified Communication Manager when acting as an active SIP endpoint.
- Fixed – Avaya audio sync issue on some outbound calls due to incorrect RTP timestamp sent by Avaya.
- Fixed – When an adapter experiences unexpected capture stop then restart the capture automatically. e.g. Hyper-V guest machine save state.
- Fixed – Handle change to MIME types for NEC notify messages.
- Fixed – Shoretel fix for incorrect phone numbers occaisionally being displayed.
- Fixed – MiTel internal call phone numbers correctly identified.
- Fixed – LG iPECS conference calls now handled correctly.
- Fixed – Shoretel fix for anonymous registrations.
29 JULY 2016
- Changed – SIP registrations are no longer detected upon request but on positive acknowledgment.
- Changed – Removed the Promiscuous Mode setting and default to always on. It is very rare to see a LAN adapter that is no longer compatible and can cause problems when disabled.
- Fixed – Added a service dependency to npf which is the winpcap driver service. Possibly on fast machines the CallN service could initially start faster than the winpcap driver.
- Fixed – Bug introduced previously for Cisco SCCP CallInfoV2 version 20 message.
25 JULY 2016
- Changed – Log when an ethernet adapter is inserted or removed.
- Changed – Remove soft restart when WireShark is closed. This is no longer necessary.
- Fixed – Detect auto-answer call setup for SIP handsets which indicates an auto dialler/desktop toolbar resulting in reversal of the caller/destination parties. Includes Aastra, Grandstream, Linksys, Polycom, Snom, Yealink and Cisco.
- Fixed – Use RTCP Goodbye and OpenLogicalChannel as an audio sync event for Avaya h.323 calls. Because of the multiple stereo/mono/stero/mono rtp changes.
- Fixed – Possible reliability with administrative remote node task retrieval stopping.
- Fixed – Panasonic detecting handset registrations with incorrectly included ‘Incoming’ in name.
5 JULY 2016
- New Feature – Update handset registration with name if the original registration didn’t include a name but a successfully recorded call does.
- Changed – Server request tasks now use new RecordingNodeTask REST api which removes 1 minute polling frequency in favour of 1 second resolution blocking.
- Changed – Limit the packet processing queue to 100,000 packets/adapter and then purge. This allows for recovery on networks processing a lot of backup software traffic.
- Changed – Now support iPECS UCP decoding.
- Fixed – LGiPECS correctly match audio streams for remote NATed handsets.
- Fixed – SIP REGISTER messages which don’t include an Expires: header will no longer assume unregister but a non expiring register. e.g. latest MagicJack.
- Fixed – Cater correctly for the same VoIP stream captured over multiple ethernet adapters.
- Fixed – Interpretation of variant Call end messages on LGiPECS.
- Fixed – Avaya various fixes.
- Fixed – Bug in RTP packet silence padding based on packet arrival time.
- Fixed – Pause/Resume dtmf sequence only resuming the keypress party and not other party on Cisco SCCP.
6 JUNE 2016
- New Feature – Decode Cisco proprietary SIP NOTIFY messages containing an ‘application/kpml-response+xml’ content.
- New Feature – Decode LG iPECS SDP packets for digital handsets.
2 JUNE 2016
- New Feature – Detect Symantec endpoint protection firewall and alert.
- New Feature – Allow user to acknowledge alarms in system try so they are not repeated.
- Fixed – Avaya outgoing calls incorrectly detected with ‘Call:’ screen update call control message.
- Fixed – Avaya various fixes.
- Fixed – Cisco Callmanager now lookup KeypressEvent using CallID before endpoint IP address. This corrects behavious on gateway mirroring.
23 MAY 2016
- New Feature – Service control library refactored and includes system event log write with stack on service start failure.
- New Feature – Increase SCM timeout from the default 30 seconds to 5 minutes. This allows the service to start on EXTREMELY slow laptops.
- New Feature – Monitor when a wireshark process is started and stopped. When found, restart the CallN service to make sure it has exclusive use of the network card(s).
- Fixed – Large changes to Avaya proprietary screen scrape call control messages.
23 APRIL 2016
- Fixed – SIP decoding of SDP failing when Content-Type: header was the compact c: variant.
- Fixed – RFC 2833 using RTP payload type 97 as well as existing 101.
- Fixed – Issue misinterpreting length of RTP silence.
- Fixed – Cisco Callmanager was applying DTMF codes to the incorrect call party.
21 APRIL 2016
- New Feature – Command line archivequeueupload.exe which will allow re-uploading of archive files.
- New Feature – Added argument for waitforexit on remove server request dos command.
- Changed – Option for Record RAW rtp removed from tray gui.
- New Feature – Added support for ShoreTel Softphone.
- Fixed – SIP decoding failing when Content-Type and Content-Length headers where not at the bottom. Was introduced in 1.25.0.
- Fixed – Multiple issues decoding Panasonic MGCP.
- Fixed – Shoretel issue relating to Unknown callers when handsets receive a call while already on a call.
6 APRIL 2016
- New Feature – Alert of any newer CallN software download available via the system tray balloon popup.
- New Feature – SIPREC configuration settings are managable via the tray application.
- New Feature – Increase SCM timeout from the default 30 seconds to 5 minutes. This allows the service to start on EXTREMELY slow laptops.
- New Feature – Added support for ShoreTel Softphone.
- Fixed – Pause/Resume keypress detection was not working for LG IPECS.
- Fixed – Detect correct caller and destination phone numbers and names on Cisco Callmanager message header version 22.
- Fixed – Correctly display ShoreTel calls with blocked Caller ID.
1 MARCH 2016
- New Feature – Decode fragmented IPv4 UDP/TCP packets sent accross multiple IP frames.
- New Feature – Broadsoft SIPREC endpoint recording ability.
- New Feature – Support for Panasonic MGCP messages.
- Fixed – Multiple fixes for decoding of NEC SV-9000 messages.
- Fixed – Unable to retrieve SID from a windows 2003 primary domain controller, causing the CallN service not to start.
- Fixed – Possible service start problems retrieving cpu core count or last boot time on a ..very.. slow server.
- Fixed – When installed and the cut down domain/username/password dialog is shown, limit credentials verify to api retrieve 1 call to stop timeout on large existing accounts.
- Fixed – Added further support for ShoreTel MGCP protocols
- Fixed – Fix to handling of Cisco Callmanager SCCP messages for DTMF codes
- Fixed – Added further support for LG IPECS messaging
11 NOVEMBER 2015
- New Feature – Detect phone numbers for incoming/outgoing calls made upon ShoreTel ver 14.2 using proprietary MGCP extension U package.
- Fixed – For LG IPECS, possibly audio sync issue on outbound calls which was introduced in 1.24.0.
2 NOVEMBER 2015
- Changed – For LG IPECS, always decode keypress via inband proprietary messages and not via audio. This method is much more reliable.
- Changed – Removed the keep / discard configurable keypress action configuration. This has been moved to the website.
- Fixed – For LG IPECS, clear keypress and audio once callinfo outboundexternal message is detected. This removes redundant ring tone within audio as well as the outbound phone number being collected as part of keypress events.
21 SEPTEMBER 2015
- New Feature – Include DTMF keypress detection results to the CallN cloud service when uploading a call. A much more advanced keypress action feature is now available via the website.
- New Feature – Added NEC iSIP decoding. Handle the SIP message variations from the SL-1100 and SV-9000 series.
- Changed – Removed the Identify call Tab as this feature is now available in the website.
- Changed – Removed the Audio Tab, which includes normalise volume, trim leading silence and stereo output. These are now the default.
- Changed – Removed the LDAP Tab, this was an unused feature
27 AUGUST 2015
- New Feature – Detect and Ignore duplicate packets. This will clean up captures and improve speed where a port mirror captures packets twice.
- New Feature – Handset Registrations survive restart as persisted xml file. This will help with LG iPECS and MiNET as handset registration is not regular.
- New Feature – Detect additional SCCP message types 0x107 is ConnectionStatisticsReq and 0x023 is ConnectionStatisticsRes. They could occur in clear down bundles which would inhibit clear-down.
- New Feature – Added L16 codec decoding which was shown in a ShoreTel system.
- New Feature – Added local status alarm system so that more than one concurrent alarm can be displayed by the tray application.
- New Feature – Alarm when no LAN adapters are found within the computer rather than just logging.
- New Feature – Alarm when the PCAP driver is not installed on the computer rather than just logging.
- New Feature – Increase service start-up speed, by not requiring all of the IP stack to have started.
- New Feature – When the CallN software client is installed for the first time on a new machine, show a cut-down settings window containing only domain/username/password as well as verify before allowing to continue.
- New Feature – Recording for native LG iPECS protocol.
- New Feature – Rework of MiTEL MiNET call recording including transfers. NOTE: No longer turn on ‘Record Raw RTP streams’ for this to work.
- New Feature – Added automatic handset detection on MiTEL systems.
- New Feature – Performance improvements on networks with a large amount of SMB traffic.
- New Feature – Tested on a Windows 10 pro installation.
- Changed – Minimum call length removed as it is now configurable via the Web portal.
- Changed – Removed “Ignore internal calls” as well as “Ignore Duplicates” as these are now handled in the Web Portal.
- Fixed – Correctly process Cisco Callmanager SCCP messages with Header version ‘CM7 type 0x11’.
- Fixed – Service would not start if the CPU information was unable to be determined.
19 JUNE 2015
- Fixed – Only decode SIP messages when complete body complies with rfc3261 UTF-8 characters.
- New Feature – Detect phone numbers for MiTEL calls via SAC messages when encrypted MiNET protocol is used. NOTE: Must turn on ‘Record Raw RTP streams’ for this to work.
29 MAY 2015
- Fixed – Correctly process Cisco Callmanager sccp messages with Header version ‘CM7 type A'(0x12) and ‘CM7 type C'(0x14).
- Fixed – Limit incoming buffer to 100,000 packets.
- Fixed – SIP REGISTER is invalid if not all characters within the FromURL are displayable.
27 APRIL 2015
- New Feature – Decode g.722.1c 32Khz 48Kbit audio from Polycom vx1500 video calls.
- New Feature – Added dynamic handset detection for Cisco SCCP using the DynCallInfoMessage message.
- New Feature – Include timezonestandardname with recording node updates.
- New Feature – Log the windows machine name on start-up. New Feature – Always log to the log file when the pcap filename changes when tracing is enabled.
- New Feature – Add trace logging of SCCP Register and Unregister events.
- Changed – Changed service start type from Delayed-Automatic back to normal Automatic.
- Changed – Optimised recording node update for registered handsets only including upon change to save a tiny bit of bandwidth.
- Fixed – First 20ms audio packet associated with wrong call party within RTP only calls.
26 MARCH 2015
- New Feature – Detect available handsets via SIP registration and report to CallN cloud for automatic provisioning.
9 FEBRUARY 2015
- New Feature – Setting to control the amount of concurrent upload threads. Help fix issue with small upload bandwidth, e.g. 5KBytes/sec.
13 JANUARY 2015
- Fixed – Possible leak of empty temp files with multiple segment audio recordings in c:windowstemp on a 32-bit machine causing stop uploading once 64K files exist.
4 NOVEMBER 2014
- New Feature – Additional information uploaded to cloud includes voip call id as well as mac address/media endpoint for both caller and destination. This allows hot standby recording points in a High Availability environment without creating duplicate recordings in the portal.
- Changed – Upon service shut-down save incomplete calls rather than just disposing of them.
24 JUNE 2014
- Changed – Upgrade to Installshield 2014. Latest version.
- Fixed – RTP no-activity timeout increased to 5 minutes from 60 seconds. Problem with recording terminated when callers placed on hold for longer than the 60 seconds.
- Fixed – Possible audio-sync issue when placing caller on hold when overlay audio tone is disabled.
15 MAY 2014
- New Feature – Detect possible outgoing name from optional P-Asserted contact url provided by Broadsoft switch.
24 APRIL 2014
- Fixed – System tray icon would flash balloon repeatedly on error.
- Fixed – System tray was not opening options form on initial install.
- Changed – Upgrade to WinPcap v4.1.3. Latest version.
- New Feature – Officially tested with Windows Server 2012 R2 and Windows 8.1.
- New Feature – Log if detected running within a virtual machine.
5 MARCH 2014
- Fixed – Possible problem when creating archive files larger than 4GB.
- Fixed – Missing audio if RTP sequence ID reset without change of Synchronization ID. Only seen with newest MagicJack plus.
- Fixed – Issues when a call recording file has more than 7 audio segments which uses multiple combine actions.
15 JANUARY 2014
- New Feature – Handle SIP CANCEL message. Not necessary, by looks cleaner in the logs.
- Changed – Capture SIP OPTIONS message in VoIP packet trace.
- Fixed – Allow receiving of RTP audio on sender RTP send port as well as the SDP port definition. e.g. unexpected symetrical RTP.
- Fixed – When an xml index file needs to be appended to a zip archive file because of rollover and that zip file is corrupt then just rename the index file and don’t try again.
21 DECEMBER 2013
- Fixed – If an archive zip file is corrupt, then begin a new file.
- Fixed – If an archive temp file is corrupt, then remove it and don’t try again.
- Fixed – If a call recording file has more than 7 audio segments, then use multiple combine actions.
- Fixed – Possibe annoying ‘Arithmetic operation resulted in an overflow’ message in logfile.
- Fixed – Handle diagnostic file upload when request has be prematurely removed from server.
- Changed – Seperate Archive logfile.
7 DECEMBER 2013
- New Feature – Format phone numbers via regular expressions.
- New Feature – Capture extra stats in heartbeat message, uploadspending.
- Changed – When the caller or destination name is the same as the phone number, disregard the name.
- Changed – Archiving files require a lot less disk I/O to construct zip files.
- Fixed – Problem introduced when using RawRTP.
3 DECEMBER 2013
- New Feature – Installation signed with a certificate.
- Changed – CPU usage decreased by approx 8%.
- Fixed – When installing on Windows 2008 64-bit, don’t allow installation with less than service pack 2 as this is the minimum required for vc++2010 sp1 runtime.
- Fixed – Possible audio combine bug with avaya h.323 facility openlogical channel change source for destination party.
- Fixed – Installer, when upgrading from a previous version or repair then make sure the service and tray application are stopped first to avoid a possible reboot.
- Fixed – Allow malformed additional body at end of cisco sccp messages. Symptom was some calls with unknown cli.
- Fixed – Use invarient culture for dates in temporary queue files, as possible issues with some non english languages.
- Fixed – Much lower CPU usage when packet logging is turned on.
- Fixed – VLAN wrapper decoding was broken somewhere along the line.
- Fixed – When using optional pause/resume via keypad, padded silence didn’t follow the configured rule.
- Changed – Remove avaya h.323 UserUser message containing nonStandard identifier 0x07 as hangup detection as it is unreliable.
- Changed – Updated lame encoder to v3.99.5 and include a native 64-bit version.
- Changed – Updated normalize executable to lower CPU usage and reduce possible clipping.
20 NOVEMBER 2013
- New Feature – Ability to configure two digit sequences for pause/record etc.
- Fixed – Audio overlay tone not working due to audio template file location.
- Fixed – Record Avaya h.323 calls that are initated with rtp stream from unexpected gateway ip.
- Fixed – Instantly clear baloontip on service tray icon instantly when a fault is rectified and there is no desktop mouse interaction.
5 NOVEMBER 2013
- New Feature – Local archive. Create local zip file(s) containing all upoaded calls for legal or regulatory compliance.
- New Feature – Keypad control recording with pause/resume and tone overlay.
- New Feature – Keypad identify calls buy adding notes.
- New Feature – Native 64-bit application and associated assemblies when a 64-bit operating system is detected.
- New Feature – Display (paused) next to upload queue count on system tray balloon popup based on upload time constraint.
- Fixed – Main application service would not start correctly when a malformed lan filter query was entered.
- Fixed – Installer now has prerequisite of vc++2010SP1 runtime and not vc++2010 which could create a false error if vc++2010SP1 is already installed.
- Fixed – On the Mitel VoIP switch, don’t use the SIP INVITE P-Asserted-Identity field to determine the destination phone number.
- Fixed – Lower cpu usage when call parties have more than one rtp source.
20 OCTOBER 2013
- New Feature – Use gzip compression when uploading recordings using the new REST api. Approx 5% better upload speed of these mp3 files.
- Changed – Diagnostic file upload also now uses the new REST api.
- Changed – Better descriptive names of the detected network adapters.
- Fixed – Monitoring network adapter also provides last VoIP packet time and last packet time for notifications.
12 OCTOBER 2013
- New Feature – Ability to disable the recording of internal calls. Avaya pbx only.
- Fixed – Only hangup detect via h.323 facility OpenLogicalChannel nullData message when no session exists.
- Fixed – System tray total calls today count reset at midnight for Windows XP.
- Fixed – Monitor network adapter for unplug event under Windows XP.
6 OCTOBER 2013
- New Feature – Ability to configure mono audio file creation.
- New Feature – Add all monitored adapter status(s) to the heartbeat messages. This will allow remote monitoring for media disconnect or no VoIP traffic.
- New Feature – Track h.323 registration messages for phone number substitution when rtcp is unavailable.
- Fixed – Non technical error descriptions for system tray baloon notification. Was introducded in 1.12.8.
- Fixed – Hangup detect via h.323 facility OpenLogicalChannel nullData message.
- Changed – Also track avaya h.323 handset display update with 0x11 and 0x12 messages.
24 SEPTEMBER 2013
- Changed – Use new optimized REST api and multithreaded calls which improves upload bandwidth usage by approx 450%.
- Changed – When capturing all packets in the trace file, ignore HTTPS as this is just the software uploading.
- Fixed – Lower memory requirement when converting g.729 audio recordings.
- Fixed – Possible memory leak crash when decoding malformed rtp packet.
24 AUGUST 2013
- New Feature – Display cpu speed in log.
- Fixed – Limit remote trace uploading to 20MB of ram usage.
- Fixed – Faster shutdown when uploading large trace files.
12 AUGUST 2013
- Fixed – Allow RTCP Goodbye to indicate hangup for Raw and h.323 calls.
- Fixed – Remove Avaya h.323 0x07 facility message from indicating hangup, it was unreliable.
- Fixed – Add g.729a support in h.323 facility openlogical channel message on Avaya.
- Fixed – Allow Avaya call setup from incoming caller phone number message without handset state event message first.
- Fixed – Simplify Avaya h.323 incoming/outgoing direction detection. Allow 0x05 facility off-hook state change before display change to indicate outgoing.
- Fixed – Re-sync RTP audio streams on SIP re-invite message.
17 JULY 2013
- New Feature – Decoding of the SIP protocol if it uses TCP packet encapsulation instead of the commonly used UDP.
- New Feature – Decoding of TZPK packet encapsulation of UPD packets.
- New Feature – Ability to enable adding incomplete calls to the total value displayed in system tray rollover.
- Fixed – No ability to send heartbeat or respond to server commands if the local machine performance counters are corrupt.
12 JUNE 2013
- New Feature – Added setting to allow calls that are optionally recorded to only include audio after keypad digit sequence pressed.
27 MAY 2013
- New Feature – Combine calls when Telstra customers dial *98 call pickup shortcode.
- New Feature – Shoretel MGCP proprietary protocol recording.
- Added – Included H.225 IE types 0x1c facility and 0x08 cause.
- Added – New setting ‘Location’ to add search parameter for multiple offices.
- Fixed – H.225 IE Calling and Called number handle optional extended numbering plan byte.
- Fixed – Raw RTP streams without synchronous port numbers.
- Fixed – Don’t always restart the network card monitoring on all setting changes.
- Fixed – Possible upload queue hang when conversion utility writes more than 2K to error stream. Noted when converting bad speex files.
- Fixed – Possible remote directory hang when listing more than 2K to console stream.
26 FEBRUARY 2013
- Changed – Upgrade to SharpPcap library v4.2.0. Latest code base.
23 FEBRUARY 2013
- Fixed – Increase upload speed with concurrent connections.
10 FEBRUARY 2013
- New Feature – Allow configuration of LAN packet capture filter.
24 JANUARY 2013
- Fixed – Visual c++ 2010 dependency as a result of Streamcoders v2.0.424 update.
- Fixed – Detect optional SIP SDP in Status 180 Ringing.
18 JANUARY 2013
- Fixed – Detection of caller phone number of incoming calls to Avaya h.323 under certain conditions.
- Fixed – Call disconnect detection to Avaya h.323 under certain conditions.
- Fixed – Sharpziplib not deployed, which is used for optional zipping of upload ethernet traces.
- Fixed – Remote server commands fix current working directory.
8 JANUARY 2013
- New Feature – LDAP integration for looking up call party names via phone number.
- New Feature – Allow callN staff to remotely request ethernet trace files be uploaded for analysis.
- Changed – Upgrade to Streamcoders v2.0.424. Latest version.
- Fixed – Updated version of g.721 decoding.
- Fixed – Internal RTP timestamp logging for uint32.
- Fixed – Speed up the perceived load time of the system tray application options form.
- Fixed – Some system tray application right-click menu items were missing after the core service was restarted.
10 NOVEMBER 2012
- Fixed – Unable to detect h.323 calls on certain versions of Avaya switch. Problem was introduced sometime after version 1.6.0 of callN software.
2 OCTOBER 2012
- New Feature – Added current computer time to server ping which will allow future automated notification e-mails informing of clock-sync problems.
- Changed – Enable auto-recover and delayed auto-start for the callN service.
- Fixed – Possible out of memory exception and shutdown when trying to decode h.323 traffic.
7 AUGUST 2012
- Fixed – Can’t connect to server to upload recordings on some non-english operating systems.
- Fixed – Version 5 of some h.323 messaging. e.g. Avaya switch.
19 JULY 2012
- Changed – Upgrade to WinPcap v4.1.2. Latest version.
- Changed – Dynamically include network adapters that are added/removed after startup. e.g Wireless USB dongles. Check every 30 seconds.
26 JUNE 2012
- New Feature – Upload calls with multiple parallel connections for greater speed.
- New Feature – Add additional machine information with ping heartbeats.
- Changed – Lower CPU requirement and faster packet processing.
- Fixed – Increase network adapter interface kernel buffer to 16MB to stop possible packet drop with saturated 1GB LAN connection.
- Fixed – Possible stop of upload queue until reboot after windows machine experiences very low system resources.
- Fixed – Don’t insert silence because of RTP time jump when previous sequence packet is a marker. Possible audio sync issue.
13 JUNE 2012
- New Feature – Compatible with Aastra MX-One h.323 pbx.
- New Feature – Allow non-english unicode call party names for Cisco CallManager calls.
- Changed – Don’t try to decode inband DTMF for CallManager calls, as handsets always pass via outband messaging.
- Fixed – Recover when anti-virus locks files in the temporary upload queue.
- Fixed – Unable to upload calls on certain non-english windows installs.
- Fixed – Cisco Callmanager call transfered and then recalled now includes second conversation leg.
31 MAY 2012
- Fixed – Include applicable silence filler within large gaps of single RTP sequence identity ranges.
- Fixed – Some versions of callmanager would cause 0x14a to place called phone number as the caller name.
- Fixed – Don’t try to decode inband DTMF if rfc2833 is detected in SIP SDP.
- Fixed – Lower RAM requirements for system tray application.
21 MAY 2012
- Changed – Upgrade to SharpPcap library v4.1.0. Latest code base.
- Changed – Upgrade to PacketDotNet library v0.12.0. Latest code base.
- Changed – Lower CPU requirement and much faster packet processing.
- Fixed – Upgrading rather than re-installing could install incompatible service/tray versions.
Note: There is now a dependency on Microsft .NET framework 4.0.
28 DECEMBER 2011
- Fixed – Variation of SCCP call info 0x14a message reserved id 0. Symptom was caller/destination phone number was not detected which resulted in all calls anonymous.
20 NOVEMBER 2011
- New Feature – Open system tray settings on first install.
- Changed – Allow raw RTP stream recording feature to be disabled via system tray.
- Fixed – Variation of SCCP media endpoint messages for Cisco UC540. Symptom was calls not recorded.
7 OCTOBER 2011
- Fixed – With certain network traffic types, very short calls are detected from raw RTP streams and uploaded if the minimum length is configured as 0 seconds. Now a minimum of 3 seconds is applied to raw RTP calls.
2 OCTOBER 2011
- New Feature – Allow the client disconnect notification. This is configured via the web portal.
28 SEPTEMBER 2011
- Fixed – Upload could fail with ‘The magic number in GZip header is not correct’.
13 AUGUST 2011
- New Feature – Allow recording of RTP streams that are not initiated via SIP/SCCP or H.323. This can be useful to record protocols such as google voice.
- Changed – Install c++ runtime from web rather than within setup.exe to decrease size.
5 JULY 2011
- New Feature – Inband RTP DTMF decoding.
- Changed – Approx 20% lower CPU requirement when recording a g.711 codec.
- Fixed – RFC2833 DTMF was broken in a past release.
24 JUNE 2011
- Changed – Slightly lower CPU requirement and full utilization of all available CPU cores.
- Fixed – When resume from sleep (e.g. laptop), calls were no longer recorded.
- Fixed – Clean up ghost tray icon on uninstall.
- An example deployment is 200 concurrent calls using a dual core i3 2.26Ghz and 1GB memory.
16 MAY 2011
- Changed – ‘Total Calls Today’ count in the system tray application now represents only calls successfuly recorded and queued for upload. Previously is also incorporated short calls, incomplete calls or duplicate start messages.
7 APRIL 2011
- New Feature – When a lot of messages are queued for upload, upload the oldest file first.
- New Feature – Audio is compressed when uploading via web services for a saving of about 5% of the bandwidth.
2 APRIL 2011
- New Feature – Added support for the g.722 codec.
- New Feature – Use extension display name within 0x14a SCCP message for calls recorded using Cisco CallManager.
- New Feature – Initial support for Avaya H.323 IP Office handsets.
- Fixed – Calls were sometimes not recorded for extensions of a ‘Ring All Group’ when using Cisco CallManager.
- Fixed – Don’t halt installer if trying to stop tray application fails.
- Fixed – Close the pcap trace file when adapter is restarted. Previously if tracing was set to diabled from enabled the trace file is still locked.
- Fixed – Include VoIP traffic provided within VLAN frames.
- Fixed – Cisco SCCP calls would stop recording if they were put on hold for longer than 1 minute.
22 JANUARY 2011
- New Feature – Handle SIP calls that change Call-ID during call. Use the From: SIP Tag to align session if available.
4 JANUARY 2011
- New Feature – Add ability to direct recordings directly to a reseller account and have them match a client via phone number.
- Fixed – The ‘Dont show tray’ setting was stored per user rather than as a global, resulting in this setting had to be made per user.
28 DECEMBER 2010
- New Feature – Add ‘back off’ time when uploading if there are issues. 1min, 3min and then 5min.
- Changed – Log when calls are dropped on shutdown.
- Changed – Don’t restart service for settings that can be changed on the fly. e.g. enable pcap tracing.
- Fixed – Setting ‘Keypad Criteria’ in tray application back to ‘All’ would not save.
14 NOVEMBER 2010
- Changed – Use SharpPcap library v3.3.0. Latest code base.
- Fixed – When checking the ‘Promiscuous Mode’ or ‘Upload Time’ from the tray application, the Apply button would not be enabled.
- Fixed – Remove invalid characters from the voip packet trace log filename. e.g. some adapters can have a in the description.
28 SEPTEMBER 2010
- New Feature – Allow ‘Ignore Duplicates’ to be configurable via the system tray application.
- Fixed – Changing ‘Upload Time’ from a configured value back to ‘Anytime’ would not save.
- Changed – When multiple RTP streams are detected for a single endpoint, give precedence to the one that matches both source port and destination within the RTP packet.
- Changed – Don’t record calls where either caller or destination is less than the minimum record time. Previously this was calculated as the maximum of either.
- Fixed – Cisco SCCP would not record call if OpenReceiveChannel/OpenReceiveChannel/StartMedia messages don’t contain the correct conference-id.
- Fixed – Accept G.729 RTP packets with a multiple 30 byte payload.
24 AUGUST 2010
- New Feature – Can select which Network adapters to record. Rather than just all.
- New Feature – Can conditionally record calls based on a keypress sequence during call.
- New Feature – Trim leading silence from recordings with levels below -40db.
- New Feature – Display calls in progress count in system tray icon tooltip.
- Fixed – Missing audio leg after hold/transfer on cisco sccp when media endpoint different to proxy.
- Fixed – Other possible one-way audio with cisco sccp under certain abnormal condtions.
- Fixed – Pcap trace log files were not rolling over to new file, instead the same file again.
- Fixed – Pcap trace log files were rolling at 105MB instead of 100MB.
- Fixed – Possible ‘collection has been modified’ error in the logfile when a call is complete.
- Fixed – Handle StartMediaTransmissionAck message for cisco sccp correctly.
- Fixed – Cisco sccp increase RTP timeout to 5 minutes once the call is placed on hold.
14 JULY 2010
- New Feature – Right click context menu item ‘Enable / Disable’ from system tray. To temporarily disable call recording.
- Fixed – There was a problem in translation of a windows permission group name.
- Changed – Silent install for WinPcap 4.1.1.
- Changed – Use SharpPcap library v3.1.0. Latest code base.
- Fixed – Would not run correctly on windows installed with non-english as the default language.
23 JUNE 2010
- Changed – Don’t end sccp cisco call on CloseReceiveChannel or StopMediaTransmission message. One-way audio may still be in progress past this point.
- Changed – Default minimum recording length set to 10 seconds.
- Fixed – Installer would not launch WinPcap installer on Vista/Windows 7/Windows 2008 R2 unless the user has disabled UAC.
- Fixed – Don’t treat SIP invite messages with no SDP payload as malformed.
- Fixed – Calculate timespan correctly with multi frame payloads of RTP data.
- Fixed – Handle multiple concurrent RTP data streams to a single endpoint, which is more than can be played within the timespan.
- Fixed – Speed up system tray feedback on re-configure.
21 JUNE 2010
- Fixed – Change RTP codec mid call.
- Fixed – Session description in SIP ACK message followed.
- Fixed – Session description in SIP Porgress 183 Status message followed.
- Fixed – Follow multiple transfers and include silence offset between change of synchronization source id’s.
- Fixed – Pause for 2 seconds between re-configure as winpcap can sometimes fail opening devices again so quickly.
- Fixed – Handle same RTP sequence id used again after a SIP re-invite.
- Fixed – Remove calls from the upload queue that have malformed xml files.
- Fixed – Handle telephone numbers/names that cause errors without encoding to xml first.
- Fixed – Detect windows server r2 clients properly instead of reporting windows 7.
10 JUNE 2010
- Changed – Don’t apply the caller/destiantion name/phone number of successive SIP re-invite messages for a call currently in progress.
- Fixed – Detect possible silence between change of rtp synchronization source id’s.
- Fixed – Handle change of rtp audio codec within call. Usually at begining of call in association with second sip invite.
- Fixed – If the audio payload length is not standard for the codec, then don’t just add it as the result is out of sync for codec framing. In this case any g.729 packets that are not 20 bytes.
2 MARCH 2010
- New Feature – A ‘show in system tray’ option.
13 FEBRUARY 2010
- Changed – Use SharpPcap library v2.4.1. Latest code base.
- Fixed – Handle IPv6 packets correctly.
1 FEBRUARY 2010
- New Feature – Right click context menu item ‘Upload calls now’ from system tray during restricted upload time.
- Changed – Added 90, 120 and 240 to the ‘Minimum Call Length’ selection.
- Changed – Removed ACD login/logout tracking.
- Fixed – Changing settings causes the service to stop and not restart. Introduced in 1.2.7.
7 JANUARY 2010
- Changed – Use WinPcap library v4.1.1. This allows install on Windows 7 / 2008R2 without using ‘Troubleshoot compatability’ workaround.
- Changed – Use SharpPcap library v2.3.0. Latest code base.
7 NOVEMBER 2009
- New Feature – Correct RTP sync when a client not honoring agreed data rate. e.g. magicjack sending 7700 bytes/sec mu-law, instead of 8000.
- Fixed – Timeout on any SOAP request was set as unlimited, reduced to 10 minutes.
- New Feature – Decode RTCP control messages.
23 OCTOBER 2009
- New Feature – Restart service on crash in Service control manager.
- Changed – Don’t store caller or destination name if it is the literal ‘unknown’.
- Fixed – RTP sync issue when Synchronization Source(SSRC) identifier is non-sequentially recycled within the same call.
- Fixed – g.729 codec silence substition frame was noisy.
- Fixed – Decode shorthand notation within SIP messages. e.g. i: instead of Call-Id:
27 SEPTEMBER 2009
- New Feature – Added about box to show version.
- Fixed – Possible RTP sync issue with fast mono channel buffering at start of call.
25 SEPTEMBER 2009
- Changed – When pcap trace logging is enabled, break files into 100MB segments.
- Fixed – RTP sync issue when sequential serial number packets but Marked semaphore has timestamp discrepancy.
- Fixed – Handle Cisco skinny packet with multiple SCCP message segments and one is an unknown type.
- Fixed – Handle Cisco no answer with CallState OnHook instead of OnHookMessage.
- Fixed – Handle SIP status 601 Decline properly. Rather than just timing out with no activity.
- Fixed – Removed dependancy on .NET Framework 3.5 SP1. Add installshield check for .NET Framework 2.0.
3 SEPTEMBER 2009
- New Feature – Added support for not recording calls from web portal configurable contacts.
- Fixed – Support for the 0x14a CallInfoMessage variant on Cisco SCCP.
- Fixed – Handle restarted RTP timestamp without Mark semaphore.
- Fixed – Handle SIP status 487 cancelled properly, instead of 60 second no activity timeout.
- Fixed – Handle RTP timestamp reseed when synchronization source numbering not reseeded.
- Fixed – Timeout on any SOAP request was set as unlimited, reduced to 10 minutes.
20 AUGUST 2009
- Fixed – Upload in 256Kb chunks to resolve problems uploading 7 hour recording from some networks.
4 AUGUST 2009
- Changed – Now uses WinPcap v4.1 beta 5.
1 AUGUST 2009
- New Feature – Added ability to track ACD login/logout via DTMF and substitute the handset number with the agent number in recordings.
- Changed – Upload timeout changed from 30mins to 12hours. Fixed – Upload time not being followed.
- Fixed – Upload time doesn’t allow day boundary crossing. e.g. 11pm to 2am.
21 JULY 2009
- Fixed – Re-invite was clearing previous call audio, affecting supervisor barge-in.
12 MAY 2009
- New Feature – Added ability to enable/disable Promiscuous Mode for the ethernet connection. This will help when using a wireless card that doesn’t support this feature.
20 MARCH 2008
- Fixed – Apply button not enabled within tray application when Login Domain/Account/Password content was changed.
14 MARCH 2008
- Initial Release.