CpIP Voice Software 4.2 Release Notes

18 May 2006

Please also read the release notes for the previous major release.

This release adds CpIP Voice Soft Switch support for:

  1. CpIP Voice Soft Phone
  2. CpIP Voice Cellular Gateway
    1. Initiating call-back from a missed voice call
    2. From incoming text messages:
      1. Initiating call-back, with optional automatic call-forward
      2. Caller-ID registration (multiple per account)
      3. Reload coupon
  3. TP 260 Digital Voice Gateway

CpIP Voice Software 4.1 Release Notes

29 June 2005

Please also read the release notes for the previous service release and the release notes for the previous major release.

The CpIP Voice Open Gatekeeper has been withdrawn and replaced by with CpIP Voice Switch which features greater stability.

The CpIP Voice Switch does consume more CPU power than the Open Gatekeeper; upgrading users whose server platforms are close to performance limits should take note.

All previous capabilities provided by the Open Gatekeeper are now provided by the Switch which also adds the following new features:

  1. Improved H.323 protocol support (including v4 features such as multipleCalls) and configurability
  2. Improved number translation configurability
  3. Support for registering to H.323 gatekeepers
  4. Support for relaying RTP through the Switch
  5. Support for some configurations where the voice gateway or client is behind a Network Address Translation (NAT) router.
  6. Support for presenting IVR services to H.323 gateways and client devices
  7. Support for Callback, where the Switch calls both the subscriber and his callee.
  8. Support for the new CpIP Voice E-media gateway (see below)
  9. Support for SIP protocol (RFC 3261), including
    1. Registered clients, authenticated by Challenge/Response
    2. Peer Gateways using either SIP/UDP or SIP/TCP, authenticated by IP address
    3. Protocol translation between Inbound and Outbound call legs, e.g. H.323 calling SIP and vice versa.

The CpIP Voice Gateway software has been removed in this release, as it has been replaced by the CpIP Voice E-media gateway which features:

  1. reduces cost and increases reliability by eliminating the hard disc (Microsoft® Windows® is not required)
  2. simplifies network management by being centrally managed through the Switch
  3. leverages the Switch's improved H.323 support, including Fast Start.
  4. supports all CpIP Voice Analogue PCI interface boards except for the CP3200 and CP3205
  5. supports Fax Relay to third-party gateways that comply with the T.38 recommendation.

In addition, this release includes the following new features.

  1. Support for Origination (Subscriber and Prepaid) billing through a Reseller tier.
  2. A new CDR field (CPRATEMOD) logs the telephony access DNIS (origination calls), call termination prefix (termination calls) as well as the technology-prefix which may be material in calculating the charged and/or reconciliation rates.
  3. Prefix digits of an originating caller's ANI (caller-id) may now be used in call rate selection from
    Routing and Rating Manager -> Task-> Multiple Calling Rates

CpIP Voice Software 4.01 Release Notes

15 March 2004

Please also read the release notes for the previous release.

CpIP Voice Software 4.0 Release Notes

14 February 2002

Please also read the release notes for the previous full release, its' service releases, as well as the release notes for the previous pre-release.

This release includes the following new features which are described in the product documentation:

  1. Support for Prepaid Calling Card Reload Coupons
  2. RDM by DNIS or user-id feature allows an operator to provide out-of-country toll or toll-free numbers to business subscribers.
  3. Default Login By Trunk can simplify the billing of some one-stage dialling scenarios
  4. Sliding Scale rating allows rate to change as the call duration increases
  5. The proportion of outbound VoIP traffic sent to different destination gateways can be controlled by using different load factor settings in otherwise-equal out-rules.
  6. The proportion of inbound VoIP traffic sent to different trunks or terminated with different carrier access prefixes can be controlled by using different load factor settings in otherwise-equal in-rules.
  7. Matching of in-rules and out-rules to the relevant E.164 numbers can now be further restricted by specifying the minimum and maximum number of digits to be matched.
  8. User can register/deregister his Caller-ID using #4 in the IVR submenu. Registration will by-pass initial password entry on next phone-call.
  9. New Voice Manager->General->DNIS Properties configuration enables different initial language and user-id prefix to be set on a per-DNIS basis. This replaces the DNIS User-ID Prefix Digits feature which is now deprecated and will be removed in a future release.

CpIP Voice Software 3.9 Release Notes

14 February 2002

Please also read the release notes for the previous major release and the release notes for its most recent service release.

CpIP Voice Software 3.4.3 Release Notes

23 July 2002

Please also read the release notes for the previous service release.

This is a service release to addresses issues in the previous release.

  1. Fix problem with G.729 codec negotiation with some versions of third-party gateway firmware.
  2. Fix problem with in-bound technology prefix containing a hash (pound) character being misinterpreted by Open Gatekeeper.
  3. Implement out-rule technology prefix in the Voice Gateway (Centralised edition and above only). Note that this means that you may no longer share the out-rule between the Voice Gateway and the Open Gatekeeper, since the rule will now arrive at the Open Gatekeeper including the prefix, and will thus need to be matched an out-rule that includes that prefix.
  4. If a Special Information Tone is detected when placing a call to PSTN with tone detection enabled, the call will not be charged. If Voice answer supervision is selected this means that the forward voice path will never be cut through for that call.
  5. Implement Voice Manager->General->Maximum VoIP Calls % for digital gateways. The current implementation counts IVR sessions as VoIP, therefore the percentage needs to be set to a higher value than expected.
  6. Resolve issue where FXS can cause sensitive PBXs to falsely detect ring upon call end
  7. Require a minimum of 5 seconds of airtime credit before placing a call
  8. For RDM calls that do not meet the above 5 second condition, play the insufficient credit IVR message
  9. New DSP firmware for the analogue PCI gateway features improved echo cancellation.

CpIP Voice Software 3.3.5 Release Notes

7 August 2002

This is a Service Release in the 3.3.x series of software releases for customers who are not upgrading to the 3.4.x series.

Please also read the release notes for the previous service release in this series.

This Service Release addresses affects the Voice Gateway for PCI bus only. The specific goals of this release are:

  1. Incorporate improvements and bug fixes to the Voice Gateway software from Release 3.4.2
  2. Any new features added in Release 3.4 or later, that are configured from User-Interface components which are new in the Admin Client of that release are expressly not supported in this service release.
  3. Group III Fax over IP which was added in Release 3.4 is expressly supported in this service release.
  4. Interoperation between a 3.3.5 Gateway and a 3.4.1 or higher Gateway via a 3.4.1 or higher Open Gatekeeper in transit mode is expressly supported.

CpIP Voice Software 3.4.2 Release Notes

7 August 2002

Please also read the release notes for the previous full release.

This is a service release to addresses issues in the previous service release.

This service release addresses the following issues with the Voice Gateway for PCI bus:

  1. Natural pronunciation of numbers: hundreds followed by no tens or units would be pronounced with an extra and.
  2. Inability to receive audio (one way voice) when interoperating with a third-party VoIP implementation that uses different UDP ports for the forward and reverse RTP audio streams.
  3. Error placing a follow-on call after calling through a protocol conversion (interoperability) open gatekeeper
  4. Incorrect gains between 3.4.1 analogue gateway and 3.3.* digital gateway

In addition, the Open Gatekeeper has been updated to address issues concerning interoperability with certain third party client implementations.

CpIP Voice Software 3.4.1 Release Notes

12 June 2002

Please also read the release notes for the previous release.

Highlights of the release

  • With Voice activated answer supervision, the PCI analogue gateway is now more interoperable with third-party gateways. Please read release notes below for more detailed information on this feature.
  • Beta Release of the Radius Server component of CpIP Voice billing software for third party gateways.
    Please note that as this software is still in BETA, the radius server executable will expire after the end of Calendar Year 2002. You will need to upgrade the executable before it expires to continue using the software after that date.
    Please also note that as beta software, it is licensed under additional terms as stated in our Beta Software License which differs materially from our Standard Software License.
  • In addition, several issues arising from the previous release have been addressed.

    Specific changes in this release

    1. Analogue (PCI) gateways now implement voice activated answer supervision. This is the default setting for new installations. Previously, line polarity reversal was required from the PSTN for terminating calls from third-party gateways. Nevertheless, line polarity reversal answer supervision is preferred over voice activated answer supervision especially on wireless mobile lines. The detector is susceptible to wireless interference noises or unstable tone signal levels.
    2. Voice Manager->Gateway->Charge Notification: now supports configurable DTMF beep
    3. New option IVR configuration option (Centralised license and above only): Voice Manager->Gateway->Play Balance on Login
    4. IVR now supports natural pronunciation of numbers using the (optional) 11 through 19, 20 through 90, 100, 1000, and 1000000 .723 IVR files.
    5. Digital Gateways now support the use of trunk-only AG-4000 boards for PCM switched ('hairpinned') calls using a VoIP AG-4000 board for IVR resources (connected together over H.100)
    6. The PostCDR and DeleteCDR executables have been phased out, being replaced by a higher performance stored procedure which is scheduled from the Database Manager utility.
    7. A bug preventing Postpaid Subscriber bills from being generated after a workstation's OS timezone is changed has been resolved.
    8. Account Manager->Accounts->Settings allows a technology prefix to be added to calls made from User IDs of a particular prefix (Centralised license and above only, applies only to calls originating from a CpIP Voice Gateway)
    9. The Open Gatekeeper will optionally prepend a technology prefix configured on a per-outrule basis.
    10. Digital Gateway operators using ISDN protocols on T1 lines who are upgrading from a previous version of CpIP Voice, please take note:
      Previous versions of CpIP Voice defaulted the Line Coding parameter to AMI and changing it required manual editing of the AG.CFG file (located in the NMS\AG\CFG subdirectory of your installation directory)
      This release changes the default to the more standard B8ZS encoding, allowing the choice to be made via a dialogue presented during the software setup procedure.
      Please make sure that you choose the correct value. If you are unsure what your current configuration is running with, you are advised to check for the LINECODE= entries in your current AG.CFG file before upgrading.

    CpIP Voice Software 3.4 Release Notes

    3 Dec 2001

    Please also read the release notes for the previous release.

    Highlights of the release

  • This focus of this release is to provide a platform for interoperability with other VoIP implementations. This is primarily achieved by protocol translation through the Open Gatekeeper software in Release 3.4, and is only available for CpIP Voice for the PCI bus. Please refer to the Open Gatekeeper documentation for more details.
  • For VoIP and FoIP calls between CpIP Voice gateways, we continue to use a proprietary RTP protocol to gain superior voice and fax quality on Internet paths that suffer from packet loss.
  • In addition, Group 3 Facsimile (T.38 Fax Relay) support has been added to the Voice Gateway for the PCI bus (both analogue and digital) however it does not interoperate with the Fax implementation in the Voice Gateway for the ISA bus, nor with any other third-party VoIP implementation.
  • The CpIP Voice for the ISA bus feature set has been frozen at the 3.3 level. Some of the features that are detailed in the next section will thus only apply to CpIP Voice for PCI because they are implemented in per-architecture software. They will be marked with a (P).
  • Support for upgrading from Version 2.x and earlier has been withdrawn in this release. Please upgrade via version 3.3.

    Specific changes in this release

    1. Simplified channel gain settings for Digital gateways: Values of 0dB for input and output gain are now default, and correspond to a setting of +12dB/-12dB for Input/Output gain for analogue peer in previous versions. A side effect is that the digital gateway IVR gain is now -12dB lower, previously it was +12dB too high. Consequently, we've made the IVR output gain adjustable.
    2. (P) As part of the interoperability rationalisation, support for authenticated gateway access from Microsoft® Netmeeting 2.11 has been withdrawn.
    3. (P) Codecs supported in this release are
      1. G.723.1 6.3 kbps
      2. G.729A 8.0 kbps
      3. G.711 u-law 64 kbps
      4. G.711 A-law 64 kbps
      However, the following restrictions apply:
      • Only G.723.1 is supported when interoperating with previous versions of CpIP Voice
      • G.729A is an extra-cost option
        • For digital gateways, the option is a software option
        • For analogue gateways, the option is a manufacturing option on the voice card itself
      • T.38 Fax Relay is only supported when the voice codec is G.723.1
    4. (P) FXS will generate busy tone upon off-hook if outgoing VoIP is barred
    5. (P) Installation of ISDN gateways no longer requires the second (inconsistent) protocol configuration screen. However, configuring ISDN DCE ports will now require configuration from the Voice Gateway Manager.
    6. HTML applications have been included in the Database Server Components to assist in
      1. Establishing automated on-line backup servers
      2. Export and Scheduled Import of Out-rules, primarily for scheduling changes to subscriber rates and destination gateway preferences.
    7. A new Taskpad includes wizards to simplify basic configuration of new dialled destinations. Detailed configuration as well as destination changes and removals still requires the use of the Gatekeeper and Account Manager applications.
    8. New database installations will include a pre-installed support quickdial *21415232 to call the CpIP Voice Support Center in Malaysia. Please refer to the documentation for our business hours.
    9. The rule that Billing periods must be generated in consecutive order without gaps in the periods is now enforced.
    10. Changes to logging of Call Detail Records:
      1. Intranet edition: CDRs displayed in the Account Manager now display the correct start/end time.
      2. Packet loss statistics will be logged for calls which did not advance to chargeable status.
      3. CDRs now include a GUID field which can match up the separate CDRs belonging to the Outbound, Transit (through Open Gatekeeper) and Inbound legs of the call. However, there is no application support for this feature.
    11. The requirement that the database server must be upgraded before the voice gateways is now made explicit: Voice gateways will refuse to run if the Database Server has not been upgraded to the required level.
    12. The requirement that gateways must not be installed from copies of the same hard disc installation image has been relaxed: You can set string values of 1 to 25 in HKLM\Software\CPM\DBFE\General\CloneID for each of up to 25 copies of the same gateway.
    13. The account manager's settings screen for adding new UserID prefixes can now set two new properties on the UserID:
      1. Disable PIN change
      2. Reset Expiry date after first Logon
      Note that you can apply these properties to previously installed prefixes by deleting them and re-adding them.
    14. New in/out rule classes. Note that older gateways will mis-interpret these rules so do not configure them into your database until you have upgraded all your gateways or restrict the rules only to apply to new gateways.
      1. Terminal out-rules A call routed via a terminal out-rule will not fall back to the next preference out-rule if it does not succeed.
      2. (P) No Auth inrules The VoIP Auth In direction setting no longer has effect in PCI gateways. Instead, you need to configure a Normal in-rule if authentication is required and configure a No Auth in-rule if authentication is not required.
    15. (P) detection of the #0# follow-on call sequence has been improved
    16. (P) If an E.164 number was dialled, it can be re-dialled by entering **
    17. (P) In-rules may restrict calls to particular sets of ports (hunt groups) whose membership is configured under Channel control in the Voice Gateway Manager.
    18. (P) Out-rules and in-rules may now reference a Call Profile which can set gain adjustments and Codec preferences on a per-destination basis. The Call Profiles themselves are configured from the Call Parameter Profile page selected from the Task menu option in the Gatekeeper application. The Codec selection setting in the Voice Gateway Manager has no effect in PCI gateways and thus has been disabled.
    19. Calls routed through the Open Gatekeeper can have their gains adjusted through the Indirect Gateway Call Profiles page selected from the Task menu option in the Gatekeeper application. The codec selection in that case has no effect, with the in/out rule Call Parameter taking priority.
    20. Other Voice Gateway Manager changes:
      1. The disappearing LED problem should now be resolved.
      2. The alert page search start date now defaults to 30 days ago.
      3. After individual channel setup, no longer need to click on the tick after Apply.
      4. (P) Channel control now includes dial suffix.
    21. (P) Fix for DTMF relay being garbled when network delay between originating and terminating gateways is greater than than the time between DTMF digits being pressed/detected.
    22. (P) Fix for failure to originate call to second preference gateway when first preference gateway is busy (all ports in use).
    23. Add facility to disable individual out-rules (set rule type to 'Disabled') to gatekeeper. The system will treat these rules as if there were not present at all.

    CpIP Voice Software 3.3.4 Release Notes

    31 Oct 2001

    This is a Service Release addressing the following single specific issue for Digital gateways:

    1. This fix enables call attempts to be retried on temporary out of service channels (Red LED showing on voice manager) when all other green channels are already in use. Previously, channels which go out of service may not be returned into in-service state.

    Please also read the release notes for the previous release.

    CpIP Voice Software 3.3.3 Release Notes

    15 Oct 2001

    Please also read the release notes for the previous release.

    This is a Service Pack to address specific issues in the software for both Digital and Analogue Voice Gateways for the PCI bus.

    Changes affecting both Digital and Analogue

    1. Add outbound PSTN release guard time

    Changes affecting Analogue only

    1. Support has been added for voice on the following new hardware:
      FXO
      CP3202/CP3202A/CP3203
      FXS
      CP8202/CP8203
      These new models are functionally identical and have the following improvements:
      1. Certified for G3 Fax speeds above 4800bps
      2. Line current detection at 20mA (applies to FXO only, previous models required 28mA)
    2. Fix for channels going out of service permanently (LED goes red) where line current detection is not available
    3. Fix for channels going out of service permanently (LED goes yellow) randomly
    4. Improve call supervision tone detection

    Changes affecting Digital only

    1. Add ANI support for the Hong KONG IDA-M protocol
    2. Work around vocoder leak which caused spontaneous gateway reboot.
    3. Increase maximum input and output gain limit from +12dB to +16dB
    4. NMS System software updated to versions contemporaneous with Fusion version 3.1
    5. When the gateway CPU is under heavy load, voice frames could be internally lost between the AG4000 board and the PC, degrading voice quality. Ameliorated by using an updated firmware (ag4000.cor) file.

    CpIP Voice Software 3.3.2 Release Notes

    23 May 2001

    Please also read the release notes for the previous release.

    This is a minor release which resolves serious issues with the previous release, and is therefore a required upgrade for all Gateways using Analogue hardware for the PCI bus.

    Support has also been added for Voice on the following new hardware:

    1. CP3201 2-Port FXO card for the PCI bus
    2. CP8201 2-Port FXS card for the PCI bus

    Issues resolved in this release

    Analogue gateway for PCI bus cards
    1. Fix bug where calls originating from FXO ports on an ISA bus gateway would not disconnect when the caller hangs up.
    2. Update DSP firmware to correct echo cancellation problems.
    3. Improve detection of DTMF digits.
    4. Transmit DTMF into the line at the correct volume.
    5. False detection of tones could lead to disconnecting calls which are still in progress: fixed by making tone profiles more stringent.
    6. Remain on-hook between calls longer otherwise the PSTN or PBX may fail to detect the gateway going on-hook.
    7. Implement partial support for tuning the echo cancellation filter (see documentation).
    8. Recording IVR would sometimes result in garbled voice.
    Voice Gateway (all types)
    Fix bug where PC calls from Clients did not respect the Call Termination Prefix field in the corresponding Gatekeeper in-rule.
    Voice Gateway Manager
    Fix bug where the configuring the ISDN 5ESS protocol would result in an invalid, non-working configuration

    CpIP Voice Software 3.3 Release Notes

    27 February 2001

    Please also read the release notes for the previous release.

    Highlights of the release

    Support for Voice on the following new hardware:
    1. CP3200 2-Port FXO card for the PCI bus
    2. CP8200 2-Port FXS card for the PCI bus
    which has these improvements over the ISA version such as :
    1. Add precise call progress tone for local tones in addition to previous generic tones
    2. Reliable (out of band) transmission of DTMF digits
    3. Improved echo cancellation
    4. Proper handling of incoming/outgoing call collision (glare)
    5. Integrated hardware integrity testing
    6. Reduced CPU utilisation
    7. Flexible configuration to detect line current and/or dial tone before dialling
    8. Interoperability with Quicknet Internet SwitchBoard version 3.2
    9. Faster call setup (H.245 negotiation)
    10. Depending on the cpu board, up to 10 PCI cards are supported per chassis
    Improved software installation procedure.
    Installation of new voice gateways is particularly simplified as the installation wizard is able to configure all the mandatory configuration records.

    Other changes include

    Database level security has been implemented
    Accessed from Database User Manager, User->Update User->Groups->Permission->Database.
    allows you to create database users with rights for:
    1. voice gateway operation only
    2. operating the gatekeeper application
    3. operator level access to the subscriber database
    4. manager level access to the subscriber database
    Call charging improvements
    1. consistent subscriber charge rate based upon best preference out-rule rate rather than actual call out-rule rate
    2. per-account markup applies to charge after per-out-rule markup is applied, rather than to the raw charge
    3. support multiple calling rates (e.g. special offers) based upon offer validity date and user ID range (ie prefix). Applicable to phone2phone calls on the Centralised ITSP edition only: gatekeeper Tasks->Multiple Rate Table
    4. Air-time announcements take account of call-rate and per-account markups
    Improved security
    The "Configure SQL access" tool no longer stores (encrypted) passwords into the registry, this functionality has been moved into the gateway configuration wizard so only passwords for database users with voice gateway (lowest) rights will be stored.
    New reduced cost editions with reduced feature sets
    1. Basic ITSP edition with reduced ITSP features (analogue only, airtime announcements, offer rates, merged phonecard ID/Password)
    2. Intranet edition which does not provide call charge facilities (e.g. call duration in Call Detail Record)
    Bug fixes, including:
    1. Gateway with same name as itself was treated as privileged even though not marked as such in gatekeeper.
    2. Voice Manager application was sometimes unable to locate its help file
    3. Speed dial function was non-functional in 3.2
    4. Digital gateway fails to redirect calls when it's full
    5. Database manager backup function
      1. did not work for databases without transactional log backups enabled
      2. could not disable transaction log backup enabled flag
    6. RTP priority fix -> Improved voice quality on busy Digital Voice Gateway
    7. Correct digital peer default gains for Digital Voice Gateway are input = 0 and output = 0
    8. Correct analogue peer default gains for Digital Voice Gateway are input = 11 and output = -11
    9. Fixed connection failure to SQL Server 2000 (NOT REGRESSION TESTED, THUS NOT SUPPORTED)
    10. Block and percentage markup was switched around in gatekeeper outrule form
    11. In Account Manager forms with Advanced Search, an invalid Advanced Search made it impossible to call up the form again.

    Known issues

    T1 ISDN gateway call termination issue.
    A T1 wink start or CAS trunk has 24 voice channels, whereas a T1 ISDN trunk has only 23 voice channels. Therefore, the board's logical channels 23, 47 and so on cannot be used and must be disabled by setting the VoIP Allow field to None in the Voice Gateway Manager. The trunk protocol and parameters of a T1 ISDN board are configured on logical channels 0, 24, 48, and so on.

    CpIP Voice Software 3.2 Release Notes

    11 August 2000

    Highlights of the release

    Version Compatibility
    1. Starting from the end of May 2001, CpIP Voice Software version 3.2 will not inter-operate with CpIP Voice Software version 2.2 and lower.
    2. Starting from the end of calendar year 2000, CpIP Voice Software version 3.2 will not inter-operate with CpIP Voice Software (Client/Server Edition) version 3.14 and lower.
    Database Integration
    The Microsoft Database Engine (MSDE) has been integrated into the release. This means that
    1. Microsoft SQL Server client utilities are no longer required for the installation of CpIP Voice Software, such functionality has been integrated.
    2. Small installations do not require the installation of Microsoft SQL Server if
      1. The installation is indeed small and requires only a modest amount of database engine power (typically no more than five workstations and voice gateways in total, and assuming low Voice Gateway port density).
      2. The operator does not require the extra functionality provided by Microsoft SQL Server.
      3. databases are no larger than 2Gbytes in size.
    Digital Voice Gateway Integration
    Digital and Analogue Voice Gateway software is now installed into the same installation folder, and accessed from the same start menu shortcut folder. Note that only one may be configured to operate at any one time.
    Multiple local domain support
    The Inter-gateway charging architecture has been changed so that the relevant account to be debited is associated with a domain trust rather than a domain. This means that multiple Voice Gateway domains can share a single database server.
    Centralised Voice Gateway Software Licensing
    The Voice Gateway Software is now licensed for operation by the centralised Software Licensing manager running on the database server. As such, the license file format has changed and the old VGWLIC.DAT per-gateway license file is gone, resulting in simpler maintenance.
    Documentation/Help files
    These have been put into their own installation sets and thus need to be installed separately.

    Other changes include

    Voice gateway IVR
    1. supports an optional talk time announcement, when configured in the voice account manager.
    2. has been relaxed so that where it is unambiguous, either * or # may be used.
    Database archival
    has been removed from the standard account manager client software to a database manager application that runs on the server.
    postcdr.exe
    An optional batch program is provided for the posting of call records from the live database to the billing database which can be run from the Task Scheduler (requires IE5 to be installed if your server runs Windows NT version 4). When run on a daily basis, this can speed up bill generation significantly.
    Paradox2SQL migration
    has been fixed so that the three day delay between steps 5 and 6 is no longer required.
    Bills and Statements
    Printed bills and statements now have a check digit appended to their reference numbers, calculated using the modulo-10 16 digit credit card algorithm, ignoring alphabetics and padding from the front with zeros (for purposes of calculation only).
    Analogue Voice Gateway
    Improved compatibility on PCs with 500Mhz or faster processors: Channel lock ups with driver error messages "...vcagain..." may occur on these PCs under previous versions.