Tenor (tm) VoIP MultiPath Switch/Gateway and Call Relay Products
P105-19-00 Release Notes

This document lists features and enhancements, as well as resolved and open inconsistencies, for the following products running software version P105-19-00:

• Tenor DX VoIP MultiPath Switch/Gateway
• Tenor AX VoIP MultiPath Switch/Gateway
• Tenor AS VoIP MultiPath Switch/Gateway
• Tenor AF VoIP MultiPath Switch/Gateway
• Tenor BX VoIP MultiPath Switch/Gateway
• Tenor CMS VoIP MultiPath Switch/Gateway
• Tenor Call Relay 60
• Tenor Call Relay SP
• Tenor Gatekeeper+

Interoperability

The Tenor DX, Tenor AX, Tenor AS, Tenor AF, and Tenor BX running P105-19-00 interoperate with Tenor Configuration Manager, CM105-10-00.

Tenor Monitor v2-0-2 interoperates with Tenor DX, AX, and AS.

See for all required files and firmware update instructions for each product.

Open Inconsistencies in Release P105-19-00

2354 Memory mapping error may occur after reboot (Tenor DX only)

Rarely, when a Tenor reboots, a PCI memory mapping error causes an exception. When this happens, the Tenor resets a second time and comes up properly. This applies to Tenor DX4120 and Tenor DX8120 only.

245 Windows XP file explorer doesn't work well with the Tenor for FTP

When using Win XP's file explorer (explorer.exe), you may not be able to FTP all the unzipped system and help files to a Tenor. We recommend running FTP from the DOS prompt or using Internet Explorer.

538 G.726 codecs do not work

None of the ADPCM (G.726) Codecs work with the Tenor. This problem only applies to codecs for voice calls and not for FoIP and MoIP.

1050 Specific database changes need a reset to take effect

When you change the CDR password or IP address, the unit requires a reset.

1134 Disconnect Supervision Options only working for 2 cycles (Tenor AX/AS)

The Disconnect Supervision Options (# of on/off intervals per cadence cycle) is only working when "2" is selected. An entry of "4" will still false answer if the ringback is followed by a busy tone. The workaround is to set it to 2. (Note: The default value was changed to 2.)

1862 When receiving malformed SIP message, Tenor does not send back message

When the Tenor receives any SIP message that cannot be decoded, it does not send back the "400 Bad Request" message.

1973 Pass Through Caller ID does not work

Pass Through Caller ID does not work. A work around would be to disable progress tones in the LCRG.

1987 ToneBasedSupervision not working on transferred call

On a transferred call, the tones were not heard on the second call.

2214 MaxForward may produce unexpected results

When using the MaxForward feature, unexpected results may happen. For example, the Tenor being called may not use its own Max Forward configuration for returning messages, but rather the Max Forward configured in the calling Tenor.

2247 UserAgent parameters do not accept blank value

To un-set any of the UserAgent parameters, the change command with blank value does not work. As a work around, you can put an empty string character in single quotes following the command, using the following format:

Instead of entering command: change 1 PrimaryPassWord

Enter command: change 1 PrimaryPassWord ' '

2341 Remote NAT does not work on SIP calls

The RemoteNAT feature does not work on SIP calls.

3646 Certain IVR parameters have no effect

The configuration parameter settings for both IVRCardNumberLength and IVRAccountLength are ignored and have no effect. These parameters are contained in both the TCRG and LCRG.

New Features Introduced in P105-19-00

1834 CNAM support

See CLI documentation for details.

1961 IVR Interdigit Timeout configuration

There are two new IVR inter-digit timer configuration options that available in the var_config.cfg file: ivrpinfdt d (controls the First Digit Timer, where d is 0 - 6, default 24 seconds) and ivrpinidt d (controls the InterDigit Timer, where d is 0 - 60, default 5 seconds).

See the Command Reference for additional information about the var_config.cfg file.

2500 Caller ID Type for British Telecom now supported (Tenor AS, Tenor AX only)

The Caller ID call type detection for British Telecom is now supported. This option, 6 - FSK, Rev Battery is available under Quintum-CASSignalingGroup-line.

2631 End of Digit configuration options

A new configuration parameter ivreod is available through the var_config.cfg file, which enables you to change the IVR end of digit for the IVR account number, card number and pin number. Valid options: * or # (default).

See the Command Reference for additional information about the var_config.cfg file.

2635 RADIUS redundancy enhancements

In the event that the primary RADIUS server becomes unreachable, the Tenor would start sending RADIUS records to the secondary RARIUD server, and then continue to send records even when the primary becomes reachable again.

A new Feature, FailoverRetryInterval (available under RadiusInfo), allows you to configure the number of calls the Tenor should try before checking the Primary again. If the Primary Server responds, the Tenor will switch back to primary. You can set the value from 0 - 5000, in increments of 50 (default value = 0). For example, setting the option to FailoverRetryInterval 10, will configure the Tenor to wait 10 calls before trying the Primary Server again.

2754 PPPoE fields expanded

The PPPoE username (PPPOEUserName) and password (PPPOEPassword) fields have been expanded to support 64 characters.

2855 Tenor allows route advance on some cause codes

In most cases, if a call fails, the Tenor will attempt to reroute. A new parameter, NoRouteAdvOnCause in var_config.cfg enables you to configure a cause code to not trigger a route advance.

To use the NoRouteAdvOnCause feature, set NoRouteAdvOnCause 34 in var_config.cfg file (or any other Cause Code which the Tenor typically re-routes like 27). This will prevent route advance on Cause code 27.

See the Command Reference for additional information about the var_config.cfg file.

2871 Operator assisted call treatment on FGD is now configurable

Currently, on FGD calls, any Operator assisted call (those starting with a 0 or similar) is hard-coded to normalize to 7005554141. There is a new parameter in var_config.cfg that can be configured so the call routes normally: FGDOperCallsReroute 1 (default, re-routes the call to 7005554141), or FGDOperCallsReroute 0 (passes along digits untouched).

See the Command Reference for additional information about the var_config.cfg file.

2895 New country support added to progress tones

New country support for progress tones are now configurable for the following: Brazil (11) and India (12). Configuration options are available through ProgressToneCountry (under Dial Plan).

2899 New parameters for SIP

New configuration parameters have been added to enable you to configure a public phone number to have a + at the start, and also for the URI to contain "user=phone". The two new options are available in the var_config.cfg file:

SipPrependPlus will add a "+" to any number except a private number in the SIP signaling message. Valid options: 0 (do not add a '+') or 1 (add "+").

SipUserPhone will add "user=phone" to SIP URI parameter list. Valid options: 1 (adds "user=phone" to SIP URI) or 0 (does not add "user=phone").

See the Command Reference for additional information about the var_config.cfg file.

2944/2933 New configuration options in SS7SG

Three new configuration options have been included to SS7SG: DefaultANIScreenInd, DefaultANIPresentationInd, and RelayANI.

DefaultANIScreenInd: Allows you to set the type of Screening Indicator messaging sent by the Tenor for this signaling group.

DefaultANIPresentationInd: This command relate to the presentation indicator (CLIR bit) in the Calling Party Information Element in the outbound H.323SETUP message. This bit tells the telco to suppress callerid. When the CLIR bit is set, the display of callerid is set to restricted presentation.

RelayANI: This controls whether the ANI of a call outbound on this routing group has a value that is relayed or suppressed.

For configuration options available through the CLI or Configuration Manager GUI for the SS7SG new features (available under SS7SG), see the Command Reference Guide at .

2959 Support for Hong Kong in SS7

New configuration options are available to support Hong Kong in SS7:

International/national indicator is configurable through the var_config.cfg file: internationalInd 1. Valid entries: 1, international or 0, national (default)

Under SS7SG, set the protocol to 5, which indicates Hong Kong.

See the Command Reference for additional information about the var_config.cfg file.

2962 Added FSK based MWI functionality (FSK VMWI) (Analog only)

If the Tenor receives a SIP notification to indicate that there is voice mail, an FSK message is sent to the line(s) so that the phone(s) can provide visual message waiting indication to the user(s). After all voice mail messages have been heard, and the Tenor receives an appropriate SIP notification, a corresponding FSK message should be sent to the line(s).

To configure, a new option MWIType is available in the CASSG. Options are listed below:

0 - Stutter only (default) 
1 - SMDF FSK + Stutter
2 - MMDF FSK + Stutter
3 - Euro FSK + Stutter
4 - Neon only
5 - LED only

3013 SIP Diversion header support

When a call is redirected, the diversion header retains the origin of where the call started.

3265 MAC address displayed at boot up

During the boot up process for the Tenor, the MAC address is now displayed (e.g., "Port A MAC Address: 00:30:E1:xx:xx:xx").

2520 Enhancements to cmd ether, including DNS from DHCP, PPPoE

Several improvements to the cmd ether command. Now shows DNS servers derived from DHCP, PPPoE, etc.

2848 Do not allow Call Waiting on a fax or modem call

Will block call waiting if the active call is a modem or fax call. Note, this only applies if this call is using FoIP or MoIP. It will not apply if the call is G.711 for example as the Tenor will not know it is a data call in that case.

2997 CID translation enhancement

If "cidt2 1" is configured in the var_config.cfg file the following will be allowed in CID translation tables:

- Wild card * The pattern field requires an explicit terminating * to specify a prefix pattern.

(example) 1732* --> 1732+ any length of digit

- Matching any number '.' is supported to match any character. '.' may be specified in the replacement field and will be replaced with the corresponding digit matched by '.' in the pattern.

(example) pattern replacement ....... 07....... (any 7 digit number gets a 07 in front) 0..... 8..... (any 6 digit number starting with 0 gets the 0 replaced with an 8)

- Support a digit * & #

• and # characters are supported with the caveat that terminating * always

specifies 0 or more following digits.

See the Command Reference for additional information about the var_config.cfg file.

3141 Now support SIP alias names

The use of alias names (as opposed to just numbers) is now supported in DNChannelMap. Inbound calls with alias names will now be matched and properly routed if configured in DNChannelMap tables. A new column has been added for this.

Also, a new configurable item in SIPSignalingGroup, "UseFromContact", has been added to control whether the Directory Number (DN), or alias name, is used in outbound calls in the from and contact fields. Default is to use the DN (same as previous behavior).

3234 Request to select ANI from a large pool of ANIs for outbound calls

Tenor can now use a list of ANIs to use for outbound calls. The list of ANIs to use should be put in a file called "anilist.txt" in the /cfg directory. One ANI per line in the file. RelayANI should be set to 99 in the outbound ISDNSignalingGroup or CASSignalingGroup.

This is limited to 4000 15 digit ANIs in the CMS. 120 in DX. This is supported in ISDN or R2.

3235 Provide support for DTMF Caller ID before first ring

A new callerid type was added for analog gateways to support callerid before the first ring. This is somewhat common in China and Taiwan.

CallerIDGeneration 7 for FXS, CallerIDDetection 7 for FXO.

3426 Second dial tone feature for IP PBXs

There are times where one would like to provide a second dial tone after dialing an outbound prefix (such as 9), emulating common PBX functionality. One can now configure one or more single digit prefixes that will provide this. See the UPDP documentation for P105 for more details.

3452 Add Caller ID Translation Table to IPRG

Caller ID translation has now been added to IPRG for outbound IP calls.

3453 Ability to Drop Calls based on ANI Length

The ability to block calls based on ANI length has been added. See P105 Command documentation for MinANILength for more details.

3180 Implement Modem over IP in non-proprietary way for G.711

The ability to detect modems and switch to G.711 in a standards-based way has been added. This give the Tenor the ability to provide Modem over IP services with other non-Quintum devices.

modemByPass in the LCRG must be 2 and the faxModemCoding in IPRG should be 8 for A-law G.711 or 9 for u-law G.711.

2298 Multi-Tenant IVR Enhancements

Multi-Tenant IVR which is accessible through var_config.cfg has been extended to include the following functionaility;

• Pre-authentication type 2 in IVR type 2 and type 3
• Multi-session in IVR type 3.

The new ivr var_config.cfg entry format is as follows;

"ivr <accessNumber> <ivrType> <customerServiceNumber> <preauthenType> <welcomePrompt> <playBalance> <playTime> <multisession>"

Where

• accessNumber = DID number received from the Central Office
  Valid Values: DID number or an asterisk('*'). An asterisk will match any DID number
• ivrType = Type of IVR application
  Valid Values: 2 (prepaid), 3 (postpaid)
• CustomerServiceNumber = Number to which to transfer the call if the credit balance is exceeded
  Valid values: customer service number, or dash ('-') if not used
• preauthenType = Caller authentication method
  Valid values: 1 (ANI), 2 (DNIS)
• welcomePrompt = Opening greeting
  Valid values: 1=welcome1.wav, 2=welcome2.wav, ..., 10=welcom10.wav
• playBalance = Playout credit balance to the caller
  Valid Values: 0 (disabled), 1 (enabled) - must be 0 for IVR type 2, and 1 for IVR type 3
• playTime = Playout maximum call duration for the destination entered by the caller
  Valid Values: 0 (disabled), 1 (enabled) - must be disabled for IVR type 3
• multiSession = Allows caller to call multiple destinations without re-authentication
  Valid Values: ** or ##

See the Command Reference for additional information about the var_config.cfg file.

2393 Allow Digital Tenors to send Alert w/PI=8 on receipt of 183Progress instead of Progress

Some PSTN switches will not cut through the voice path on Progress Indicator =8 in a progress message. This caused issues when receiving a 183 progress from SIP. Cut through can be established if sent in an alerting message. See P105 command documentation for ForceAlertOn183

3174 Add ability to set community string in traps in UI - SNMPTrapCommunity

A new command, SNMPTrapCommunity in the MasterChassis prompt, was added to allow for sending a community string in SNMP traps. If not set, previous behavior of sending "trapcommunity" will be used.

3390 Nortel dial plan enhancements

Functionality was added in H.323 to support Nortel's proprietary Type of Number coding using partyalias numbering. See P105 command documentation for UserProgrammableDialPlan for details.

3405 Show received dial tone level in "cmd test o" command

A feature has been added to show the audio level of the received dial tone when running the cmd test o command. The reported level WILL be affected by the rxgain settings in cassg-line. This feature can be used to remotely determine the audio level of the PSTN line, and used to set proper gain/loss levels for the line.

Resolved Inconsistencies/Changes in P105-19-00 GA Release not included in P104 Patch Releases

2765 Error Message In CLI for UPDP needs to be changed

Corrected misleading error message when configuring UPDP.

2866 Tenor AS: Generates wrong alarm report when logged in from GUI

Tenor was reporting a failed login alarm when logging in from Configuration Manager, even though the login was successful. This has been corrected.

2926 Unattended transfer with SessionTimer crashes

Under certain, fairly rare, circumstances the Tenor could crash during a long unattended transfer. This has been resolved.

3031 One way INFO digit transmission for transferred calls

When digitrelaysip is configured for INFO (5) there were times where the digit is only heard in one direction. This has been resolved.

3150 Tone based disconnect supervision fails for calls to IVRs

With some very persistent IVRs, auto-attendants, etc. that do not stop talking it is possible that a call might hang indefinitely as the speech will make it difficult for the Tenor to detect disconnect tone from the calling side.

A new var_config.cfg variable was added to adjust the sensitivity of the detection. If this is an issue, contact support for guidance.

See the Command Reference for additional information about the var_config.cfg file.

3354 Hostname is truncated in 'From' and 'Referred-By' Headers

There were situations where the hostname was truncated in the 'from' and 'referred-by' headers in SIP. For example, quintum.com might show as quint. This has been resolved.

3382 Modified tone detection algorithm

The 2100 Hz tone with phase reversal detection algorithm from both PCM and IP sides now remain on for the duration of voice and G.711 fax calls. Detection of Fax CNG tone is now disabled by default. Also see enhancement 3269 below.

3269 Fax CNG tone detection on VoIP call origination side

The call discrimination algorithm has been modified to optionally include the detection of CNG fax tones which initiates a transition to fax mode on the VoIP origination side. Detection of fax CNG tone on the VoIP origination side may now be enabled by including the line "enableCNGdetection 1" in var_config.cfg.

See the Command Reference for additional information about the var_config.cfg file.

3395 When Authentication and PRACK are enabled, Canceling of call

When proxy authenticated the call and PRACK is enabled, caller is unable to cancel the call. Upon hanging up of the phone, there was no CANCEL sent out and called party kept on ringing. This has been resolved.

3422 Disable Echo Canceller when calling Modem Pools

There are certain types of older modems that do not generate 2100hz answer, instead using 2250hz or other tones. In modern use, this often occurs with credit card services. Tenor cannot detect and act upon these tones, so the echo cancellors are not turned off, causing intermittent failures.

As a workaround, one can put a hopoff number pattern with the credit card modem pool number, and set TON = 99. When that number is called over an FXO line, the echo cancellors will be disabled. We hope to have a better fix at some point.

3438 SS7-related database update time

Some SS7 enabled CMSs were taking up to 20 minutes to boot. This has been resolved.

3439 MaxTalkTime in CAS outgoing side does not work

MaxTalkTime values less than -1 (outbound) were not working on CAS types, only in ISDN. This has been resolved.

3440 Request ability to disable SIP registration completely

If one wishes to disable SIP registration completely, they may now set RegisterExpiryTime to -1 in the SIPSignalingGroup.

3442 CMS - SS7 Echo cancellation for India SS7

For SS7 applications the Echo control Device Indicator bit will now be set in the Nature of Connection Indicators parameter. This will improve echo cancellor behavior, especially in India.

3445 Upon config/IP change Tenor corrupts the RMSS registration

In some circumstances when configuration (gateway->Description) or DHCP IP address is changed. Tenor sent corrupted registration messages to RMSS, and after 90 seconds. It would disappear from RMSS registered tenor list. This has been resolved.

3449 Radius username field is broken in case of Multisession call

A change made in P104-12-15 broke RADIUS on multisession calls, improperly populating the username field. This has been resolved.

3458 Need better match between FXO port and PSTN line to eliminate echo

A new variable has been added to better match PSTN line characteristics. See LocalLoopType in the P105 documentation for more details.

3466 Analog Only; Line Gain Default Values

A new loss plan (gain/loss values) has been implemented. This only applies to new units, and after a factory default. It will not apply to an upgrade.

IPRG:

 RxGain = 0
 TxGain = 0

CASSG-line

 TxGain = 0
 RxGain = 6

CASSG-phone

 TxGain = 0
 RxGain = -6

3471 SIP RemotePartyId not working for SS7 configuration

We now support sending RemotePartyID from SIP calls hopping off on an SS7 trunk.

3474 calling party number is not right for some SIP calls

If a SIP alias was on a SIP call, hopping off on SS7, it was sent as-is. This is an error. It will now only sending calling number only if it is a number, not an alias.

3480 Radius return code 51 sets a credit timer to unlimited

If RADIUS sent a return-code 51, credit-time was ignored. This has been resolved.

3487 PrivateDN length used instead of UPDP mind maxd

When UPDP is enabled, PrivateDNLength should be ignored. It was not being ignored, it now will be.

3489 AutoSwitch number always is translated to 7 digit local number

There were issues with 10 digit US dialing when an autoswitch number was in the same area code. This has been resolved.

3502 PRI decoder has problem to handle message larger than 128 bytes

PRI logging was having problems with ISDN packets larger than 128 bytes. This did not cause any functional problems, just logging problems. This has been resolved.

3504 Analog Only; Failure To Detect Loop Drop On FXO Ports

There were intermittent problems on forward disconnect lines where the drop was not detected. This has been resolved.

3505 Allow AliasName in transfers

If an incoming refer-to header had an aliasName, Tenor was stripping the alpha characters before sending the new Invite. This has been resolved.

3274 SSRC validation for incoming RTP streams

In order to prevent an old stream and new stream being mixed, this occurs especially on music-on-hold systems, rtp will now be rejected if the SSRC does not match.

3333 Hardware Registration takes too long

Some efficiency was added to the boot process, speeding up system boot up.

3622 H.323 Fast Start Open Logical Channel (OLC) is not encoded correctly

Call setup problem between inbound SIP, and outbound H.323 call legs in Call Relay products. H.323 outbound call connect does not encode codec information in the fast start OLC message when the inbound SIP call includes SDP information only in the 200 OK message. This has been resolved.

3625 Incoming DASS2 to outgoing SIP calls may receive no ring back tone

Incoming DASS2 calls may not receive ring back tone when calls are being terminated to a SIP endpoint which sends a 183 Progress message without a 180 Ringing message. A timeout and subsequent call disconnect occurs after 20 seconds. This has been resolved.

1405 DHCP doesn’t work with Cisco PIX

There were intermittent issues obtaining DHCP from Cisco PIX firewalls. This has been resolved.

2032 SIP Proxy/Outbound Proxy Failover algorithm clean-up

Several cleanup items to the proxy and outbound proxy failover mechanisms have been implemented.

2119 For New Zealand testing change the twist for DTMF generation

Change the DTMF power levels to -11.4 and -9.6 in order to fix the twist for New Zealand. Twist is the variation between the 2 DTMF frequencies. This should provide better DTMF performance in New Zealand.

2134 AX48; Reset at "memPartAlloc: block too big- 78136"

It was possible, though difficult to reproduce, to crash an AX48 when submitting after creating a large number of TCRGs, LCRGs and Channel Groups. This has been resolved.

2533 - Need configuration to allow suppression of Session-GUID in SIP messages

Interoperability problems could occur with certain SIP gateways, such as AudioCodes and 3Com if the "session-guid" tag was included in a SIP message. The devices would choke on it, and not parse beyond the session-guid line. session-guid can now be suppressed by including the line "SIPSessionGUID 0" in var_config.cfg.

See the Command Reference for additional information about the var_config.cfg file.

2573 CR60 crashes in H.323 to SIP call when EnableSignalingOnly is set to 1

Call Relay could crash if SIP to H.323 transcoding was occurring, and the enablesignalingonly parameter was enabled. This has been resolved.

2577 FreeBSD and various Linux OSs are unable to telnet to Tenors

Several Unix and Linux based telnet clients were unable to telnet in to a Tenor. This was system dependent, but was seen on many new Linux and BSD systems. The Tenor was disconnecting the session immediately after prompting for the user id. This has been resolved.

2624 AnalogInterface-line name shows "10-Port DSP Card"

Minor display issue was showing a 10 port DSP card at the Analog Interface prompt. This has been resolved.

2649 Account Blocked gives incorrect Prompt

There were situations when return code 7 in RADIUS would play the "technical difficulties" error message. It will now play "account expired".

2751 BX: License upgrade does not show correct number of DIs

After a license upgrade on a BX it would sometimes show an incorrect number of digital interfaces. It required tech support intervention to resolve. Now, a simple reboot after the license upgrade will resolve it permanently.

2786 Credentials encode failure

Tenor did not support the fairly rare "qop" tag in SIP used to select authentication procedures. This has been resolved.

2789 DHCP: endless IP request in network trouble case

A Tenor configured for DHCP needed to be reset if there were serious network problems. A reset is no longer required, the Tenor will quietly attempt to reestablish DHCP communications.

2791 Caller ID detection failures when using busy out and/or reverse battery

Caller ID detection would fail on several callerid types if the port was busied out, then restored. This has been resolved.

2802 Do not include Require: timer in 200Ok when not supported remotely

When Tenor received an INVITE without timers information and had SessionTimer set to required, Tenor included Required: timer header in 200OK. This did not create s problem in Quintum gateways but it might confuse non-Quintum gateways. It is now coded as "supported" rather than "required".

2823 CR-60 Only: RFC2833 digits coming in for H.323 outbound calls causes reset

On Call Relay 60 (does not apply to Call Relay SP) and the system is used to convert from SIP to H.323, it was possible to crash the system if RFC2833 digits were coming in on the SIP leg. This has been resolved.

2849 CR60 - Changes submitted not saved to flash

Under certain circumstances changes made on a CR60 were not being saved to flash (would not be saved across a reboot). This has been resolved.

2868 Via/DNS problem

Responses to SIP messages were going back to the IP address we received them from, as opposed to using a DNS name in the via header. This has been resolved.

2879 - SIP: Support for Nortel CS1000 Voice Mail (Redirect)

Support was added to interoperate properly with a Nortel CS1000 Voicemail server using redirect.

2964 'SipServerInFromHeader' configuration item in Subscribe not being observed

The 'SipServerInFromHeader' configuration item was not being observed when the Tenor sends a Subscribe message. It now is being observed in Subscribe.

2972 BX- SIP Hold Call

Call hold was not working properly if the Reinvite was responded to without a phone number in the contact header. This case will now be handled.

2976/2977 CallerIDType 3rd column help inaccurate

Several corrections were made to 3rd column (CLI) help for calleridtype.

2995 SessionExpires has bogus value after a transfer

The sessionExpires tag had random and incorrect values after a transfer. This has been resolved.

3008 Configured ToS/DiffServ value not being set in RTP packets

Under certain circumstances the TOS values were not being set in RTP as configured. This has been resolved.

3021 File descriptor leak - cannot access file system

On certain versions of earlier code, if var_config.cfg file existed, it was possible over some time to get to a point where the flash file system could not be opened. Configuration changes would not be saved; one could not ftp to the system, etc. This has been resolved.

See the Command Reference for additional information about the var_config.cfg file.

3026 Tenor crashes when setting SNMPTrapIP2

Tenor could crash if SNMPTrapIP2 was set. This has been resolved.

3128 Wrong progress tone sent by Tenor/CMS

If progresstone 1 is set in TCRG, and calls hopping off to an ISDN trunk got rejected with a cause code in progress or call proceeding messages, ring back was sent to the IP caller. Busy will now be sent.

3129 CMS only: SNTP failing on local timeservers

CMS could fail to retrieve time from an NTP server if the response came back too fast (such as on a timeserver on the same network). This has been resolved.

3149 - GK/VoIP Call licensing checks off by 1

It was found during testing that under certain circumstances the license check would only allow 1 less than the licensed amount for GK and VoIP calls parameters. This has been resolved.

3185 Event log contains SNMP syMFree exceptions

Several misleading exceptions were being shown in event logging. These incorrect messages are now suppressed.

3187 SIP: don't close/open DSP channels on Re-INVITE

During a reinvite the DSP was briefly closed then reopened, causing a brief dropout. This caused no issues for voice, but could cause issues to modems. This will no longer occur.

3194 Clean up error logging message "No echo during 30 sec 0"

A PPPoE related error message "No echo during 30 sec 0" has been changed to "PPPoE: No pppoe packet for 30 seconds. DSL link or Server may be down." to make it clearer what is occurring.

3207 PPPoE : tNetTask < LOG_ERR, queue send error

This error could occur even if PPPOE is not enabled, but some PPPoE traffic was seen on the network. It is now suppressed to avoid causing confusion.

3208 SessionTimerExpires must always be greater then MinSE

A configuration check was added to ensure SessionTimerExpires is always be grater then MinSE.

3214 If RequestURI is different from the destination report ANI as the destination number

When a SIP call is forwarded by the destination, the destination number (To field) and the ANI (From field) are left unchanged. This causes problems in billing if both calls (the original and the forwarded) pass through a Call Relay, as the Pulse's billing is based on the ANI. The forwarded call should be billed to the original destination. Tenor will now change the original ANI to the original destination number, if the original destination number is different from Request URI.

3222 Call failing because IP does not match configured ProxyIP

Legitimate calls from the proxy could fail if "AllowOnlyProxyCalls" was set to 1. These situations were resolved.

3249 - PPPoE : send PADT with the source MAC address 00 in Tenor reset

Tenor was improperly sending a MAC address of all zeros after reboot in the PADT message. This could cause some PPPoE servers problems. Tenor now sends proper MAC address in PADT.

3251 - CRSP crash when SIP supplementary service messages received

Call Relay could crash if SIP supplementary services messages were received. Call Relay does not support SIP supplementary services and will now ignore these messages, rather than crash.

3252 Enhancement for Session Timers

Some SIP servers were improperly coding Session Timer values in mixed case. The code has now been made more liberal in accepting mixed case refresher tag coding.

3309 Analog Tenors: Disconnect Supervision: Extend ON Time Recognition to 1.25 Seconds

Some PBX's and PSTN switches, most notably in India, send disconnect tones longer than 1 second. Tenor can now be configured to detect tones up to 1.25 seconds in duration.

3332 For Unattended Transfer, * and # are being sent as E and F respectively

For unattended transfers (did not apply to attended transfers) if the number included a * or #, it would be routed, in error, as an E or F. This has been resolved.

3338 Implement a debug command to flush the ARP table

2 new CLI commands have been implemented to manipulate the ARP table. "debug arp" will display the table. "debug arpflush" will flush it.

3349 Tenor crashes when db files Integrity Check failed

Tenor could crash if the database files were corrupted (usually when corrupted by manual editing). This has been resolved.

3363 - Empty ADD DN command caused reset

When working with a Static Route and "add dn" (with no value) was entered the system could crash. This has been resolved.

3376 DTMF from H.323 to SIP broken in CR for non default Payload Type

When Call Relay is converting from H.323 to SIP, and the sipdigitrelayPayloadType was set to anything other than the default 101, the Call Relay would still use type 101 instead of the configured payload type. This has been resolved.

3378 FXO-FXO port mapping

Port extension (port mapping) with an FXO line at both ends did not work properly. This has been resolved.

3386 Japanese Ring Back Tone is changed sometimes

Ring back for Japan would work for a while, then sometimes it would be distorted, requiring a reboot to resolve. This has been resolved.

3399 Fax redundancy settings over 1 could cause reset on 120 port DSP card

This applies only to CMS and DX with a 120 port DSP card. If T38HSDataRedundancy is set to over 1 and the system incurs heavy load, the system could crash. This has been resolved.

3402 With intercom enabled, cannot set 2 routing prefixes to null

A restriction preventing multiple prefixes to be set to null (blank) when intercom has been enabled has been removed. One can now set several of IPRoutePrefix, PSTNRoutePrefix, and MultipathPrefix to blank/null.

3406 Adjust FXO port RxGain to compensate for 1.8 dB calibration error

A slight calibration error caused the input audio level from a FXO/PSTN line to be reduced by 1.8db. This has been corrected.

3416 Calling name feature cause QSIG protocol to send out FACILITY message

If the ISDN protocol was set to QSIG, orientation user and the relaycallingname is set to 2 a FACILITY was sent out with the calling name. This is not correct behavior, and often caused corrupt calling party info to be sent. This has been resolved and will no longer send out a FACILITY for calling name in this situation.

3418 - ACK message has incorrect branch

There were situations where a SIP ACK was sent from Tenor with a VIA that contains a branch that is incorrect per RFC3261. Though there were no known interoperability issues caused by this, it has been corrected.

3419 Tenor detecting reflection when EchoCanceller is off

When connecting to a Super G3 fax the echo cancellor is turned off, intermittently causing echoes of fax CNG to be detected. This could cause multiple re-invites and cause call failures. This has been resolved.

3420 New IP address by PPPoE reset does not send RRQ

If a new IP address was received via PPPoE an RRQ (H.323 registration) message was not sent. This caused a significant delay in re-registration and calls would not be sent by the GK during this time. A gratuitous RRQ will now be sent immediately to re-register the Tenor with the new IP address.

3428 - One multisession key (*/#) is input by double key

A bug introduced in later P104 versions may cause multisession keys to be recognized twice. This has been resolved.

2068 Invalid Call-ID in Invite caused the system to freeze

An invalid Call-ID in Invite caused the system to freeze. This has been fixed.

2158 Tenor did not get dynamic IP address

With DHCP enabled, the Tenor was unable to get an IP address from a router functioning as the DHCP server. This has been fixed.

2315 Ethernet interface was disabled if DHCP discover failed

If the Tenor attempted to renew its lease, and the DHCP server did not respond, it gave up after about 2 minutes. After it gave up, the Ethernet interface was disabled. Although the ei show command may have displayed a valid IP, and the siprd show command may have shown a valid default gateway, when you tried to ping an IP on what appeared to be the valid subnet, a system error appeared. This has been fixed.

2611 Codec G723 5.3 and 6.3 kbps connection led to 1 way voice

If one end of a call was set to 5.3 KBPS and other end was set to .3 KBPS G.723, the call ended up with one way voice. This has been fixed.

2622 No IP Address in Contact field caused Tenor to freeze

When a SIP call was placed from a soft phone to a Tenor without the local IP address being configured on the soft phone, the Tenor froze. This has been fixed.

2641 Could not hear ring back tone when set to ITU SS7

In SS7SG, if the protocol was set to 1 (ITU), the calling party could not hear ring back tone. This has been fixed.

2733 Could not reconnect after accepting a Call

When a Tenor accepted a Call Waiting call while on hold from the remote side, after finishing the Call Waiting call, it could not connect back to the original call. This has been fixed.

2767 Normalized Number was incorrect in UPDP

Using UPDP, the number pattern for Type 4 was not normalized correctly. This has been fixed.

2771 Too many digits in Pattern (UPDP) resulted in Tenor hanging

Using UPDP, when adding a pattern of 16 or larger, you should have received an error message, but when you did a submit, the Tenor would hang instead. This applied to CLI only (not Configuration Manager GUI). This has been fixed.

2790 Invalid pass thru id caused the system to go down

When a pass thru trunk was associated in the TCRG and LCRG, if the pass thru id in LCRG did not match the id in the TCRG, the system would go down. This has been fixed.

2798 Detected Caller ID on Trunk Circuit Ignored

For certain PSTN providers, the Caller ID was ignored and the call was sent with a setup message but no caller information. This has been fixed.

2847 In pass through mode, NI2 switch doesn't pass Network Specific Facility (NSF) correctly

This has been corrected.

2869 When SIP INVITE message contained Non-Numeric User ID, Tenor did not signal to phone correctly

When a SIP INVITE message contained Non-Numeric User ID, Tenor did not signal to phone correctly. As a result, the phone did not display the calling number correctly, displayed a wrong number, or blank digits, such as "_ _ _-_ _ _-_ _ _ _". The results would vary by phone manufacturer. This has been fixed.

2889 Inaccurate message reported

When in masterchassis, if a status command was issued, an inaccurate message was reported back: "Chassis Temperature : Hot" (or 'normal' or 'warm'). This has been fixed.

2913 For SIP, RegisterExpiryTime set to 0 still continued retries after one attempt

When the RegisterExpiresTimer was set to 0, and the REGISTER was never responded to by Proxy/Registrar, the Tenor kept sending REGISTER and retries until the Proxy/Registrar responded. This has been fixed so the Tenor stops retrying after one cycle.

2928 Incomplete Transfer caused Tenor to freeze

Incomplete Attended Transfer with session timer caused the Tenor to freeze. This has been fixed.

2930 Tenor did not submit changes if reboot occurred too soon after submit

When a Tenor was rebooted (through debug reboot) too soon after a submit, any submitted changes were not correctly submitted to the Tenor. This has been fixed.

2940 Call with empty contact list not processed correctly

Upon receipt of a call that has an empty contact list, the Tenor was not processing the call correctly. This has been fixed.

2970 Tenor not responding to H.245 End Session message

Tenor now responds to H.245 End Session messages received from peer. This is an interoperability enhancement which helps gateways which wait for an explicit End Session acknowledgement prior to call disconnect.

2988 When using BRI, no voice on 2nd channel

When using BRI, and the 2nd channel was selected, there was no voice. This has been fixed.

3009/1561 Tenor did not provide local ring back

The Tenor was not providing local ring back. This has been fixed.

3029 SendRemotePartyID configuration was not working completely

When the Tenor was terminating a SIP call, the SendRemotePartyID configuration was not working completely.

3051 Community string not sent in SNMP traps

For SNMP traps, in the "Community" field, the Tenor sent the string "trap community" as opposed to the configured community string. It is now configurable in the field "SNMPTrapCommunity" under the MasterChassis prompt.

3069 Call failure when using an external Gatekeeper

There were issues when a call is routed to an external Gatekeeper and routed back to the same Tenor for termination. We were improperly sending 127.0.0.1 (loopback) as the RTP destination. This caused issues. This has been resolved.

3093 In certain scenario, using G.711 codec would freeze the Tenor (Analog only)

If a SIP call using G.711 loops back to the same Tenor, it would result in a crash. That behavior has been resolved.

3107 Call from SS7 could not be routed by Call Routing Server

A call could not be routed by the Routing Server if it came from SS7. This has been fixed. trunk.

3201 Packet Saver uneven bandwidth utilization (Call Relay only)

The Tenor Call Relay showed uneven bandwidth utilization when Packet Saver was used. The size of the Packet Saver packet was much smaller on the transmit side than on the receive side. This has been fixed.

3215 AllowOnlyProxyCalls was set to 0 after deleting DB

When the db files were deleted on a Tenor, then the Tenor was rebooted and came back up, the AllowOnlyProxyCalls was set to 0 (zero). It should have been set to 1. This has been fixed.

3246 Tenor reset if media was looped back and RemoteNAT was enabled (all units except CMS)

If a circuit call came in on the Tenor and went out over IP (SIP or H.323), but it was redirected back to the same Tenor (causing the media packets to be looped back within the Tenor), the system reset. This has been fixed.

3296 IVR Multisession may not have worked before call was connected

If a terminating endpoint did not support fast-start (H.323 calls) or did not send 183Progress with SDP (SIP calls) the Tenor did not allocate resources properly. This has been fixed.

3307 In certain scenarios, the Tenor's internal resources are deallocated/allocated unnecessarily

In certain scenarios, the Tenor's internal (DSP) resources are deallocated/allocated unnecessarily, which produces undesired results. This has been fixed.

3337 1-second initial audio is dropped for calls

Some of the calls coming into a Tenor over a circuit interface (PRI) and destined for an IP phone, miss the first (approximately) one second of audio. This has been fixed.

Other Changes Since P104-12-00 GA Release

2618 SIP Call now sends 481

If an endpoint is in the Ringing state and receives an Invite with Replaces header, the Tenor will now reject the 2nd incoming call with a 481 and continue to Ring the 1st call.

2886 Tenor allows static DNS entries as IP address

The Tenor now allows static DNS entries to be formatted as IP address.

3110 480 response converted to cause code 18 instead of 41

Based on RFC-3398, the 480 response (Temporarily unavailable) has been converted to ISDN cause code 18 (No user responding).

3060 Ignorednis configuration changes

Ignorednis configuration (available through the var_config.cfg) applies to LCRG only, and only when ForceRoutingNum is configured. The default is now 1 (changed from 0).

See the Command Reference for additional information about the var_config.cfg file.

Changes From P104-12-00 Introduced In P104 Patch Releases

P104-12-16

2837 Use DN for Authorization in Register

Adds an option to use the Directory number from DNChannelMap in the from and to in REGISTER messages, instead of username.

This is enabled by setting SIPUseDNinRegister 1 in var_config.cfg

See the Command Reference for additional information about the var_config.cfg file.

3359 Incoming Fax Call Causes DSP Crash If FaxRelay Is Disabled

There were situations where the DSP may crash if a fax call is received and FaxRelay is disabled. This would not likely affect general operation, as the DSPs recover on their own. This has been resolved.

3435 CH: SIP Mica/Moca can cause memory leak

There was a memory buffer leak if MaximumOutboundCallsAllowed or MaximumInboundCallsAllowed were set to anything other than -1 (default). This would likely take a long time to reproduce, but would be an issue on busy systems. This has been resolved.

P104-12-15

3283 Media Stream Loopback functionality not supported in Call Relay

Prior to this, the Call Relay products did not support sending RTP to itself. It now will.

3397 Gateway could crash if FaxRelay redundancy enabled

Under heavy load with certain codec configurations, and FaxRelay redundancy configured, it was possible to cause the gateway to become unstable. This has been resolved.

3404 System upgrade may cause file corruption

It was possible to corrupt the file system if the system is updated, and rebooted, within approximately 1 minute of a reboot. This was seen in lab testing, and unlikely to be seen in the field due to the rapid operation necessary. It has been resolved.

3408 Incorrect User-Name Value in Start Accounting

User-Name in the Start-Accounting message did not have the right value when IVRType 9 and IVRAuthType 1 is used in the IPRG. The value of the User-Name is now the same in all the radius messages: Start-Accounting, Authentication, Authorization and Stop-Accounting.

3410 CR60 crashes while trying send RTCP and packet saver packets

RTCP used with PacketSaver is causing crash in CR60 due to different memory access requirements. The code will be changed to accommodate CR60 memory access requirements. CRSP was not affected, and CR60 is now working fine.

P104-12-14

2578 Implement Uniform Transfer Feature

"Uniform transfer" allows for the same operation to perform either a attended or unattended transfer. In order to obtain "Uniform transfer", the admin will configure in lcrg or tcrg:

1) Unattended Transfer Keystroke = HU AND 2) Attended Transfer Keystroke = HU

Currently, the default for Unattended Transfer is 90.

Uniform transfer has been implemented as recommended in draft-ietf-sipping-cc-transfer-07.txt, section 7.6.

Here is how it will work:

The user will have call 1 up. When a transfer is desired, they hit 'flashhook' and dial the phone number. After the number is dialed they have a choice: 1) hangup - in this case they do not know the result of the transfer. 2) wait for call 2 to connect and then hangup to perform a regular attended transfer.

If they choose to hangup before call 2 is connected, then although the phone has been hung up, the call stays in progress.

If call 2 gets connected, then a Refer w/replaces is sent. (then we disconnect ourselves from both sides).

If call 2 fails, we disconnect the original call.

If call 2 does not answer, after 1 minute the Tenor will disconnect both calls. During this time the user CAN pick up the phone and make other calls - the in-progress call has become a logical entity.

2648 Toll Free Feature is Not Working

Using Calling Card IVR functionality if the RADIUS server responded with a Return Code 13 (Toll Free Call) the call was disconnected. It will now continue the call, with no prompt for PIN or account number.

3028 Channels not being freed

There were situations, especially with SIP forked calls with many forks, where channels may not be freed. This situation has been removed.

3048 Progress tone doesn't work in some analog circuit switch call

Analog to analog calls, such as FXS to FXS, FXO to FXO, and IVR calls switched to an FXS port were not getting progress tones (such as ring back and busy). This has been resolved (portions resolved in bug 2571).

3141 Support for SIP alias names

Added support for the use of SIP aliases to be used in authentication and routing.

For example, DNCM could have an entry with DN of 18475551212 and AliasName of john_smith associated with a single entry in DNCHannelMap.

There are two new configuration items:

1) A new column in DNChannelMap called "AliasName". The default will be blank.

2) "UseFromContact" in SIPSignalingGroup. This controls which value the Tenor will use (DN or AliasName) for outbound IP calls (in From and Contact). The default will be DN.

On incoming calls, the 'username' portion of the To header has always been compared against the DN's in DNCM (or huntldn). This will still happen, but will additionally be checked against the 'AliasName' in DNCM. If a match is found, the call will be directed to the line/channel that matches.

In many situations it will be necessary to Register the AliasName, not the DN. To do this, the Tenor must configure it's user agent Contact field with the AliasName, not the DN.

3327 SuppServ: Provide a way to hang up a ringing '2nd call'

Assume that a call is up and the User hits hookflash, dials a number and then hears ringing. Currently, if the User wants to hang up the 2nd ringing call before it's answered the only way to do so is to wait 60 seconds. Then call 2 is disconnected and the User is returned to the original call.

Now, the second call can be terminated using hookflash. While the 2nd call is ringing, if the User hits 'hookflash', the ringing call will be disconnected and the original call will resume.

3361 Australian Progress Tone not working

When ProgressToneCountry was set to Australia, US progress tones were played. This has been resolved.

3380 PTE= 2 on Tcrg also presents LDN routes

When PTE is set to 2 on tcrg, order in which routes should be tried is as follows:

• Try IP first.
• Try Hopoffs second
• If the above fails, try passthrough to LCRGs with matching PTID.

Previously it was also finding LDN/DNChannelMap routes. It will no longer find LDN/DNCM routes, just passthrough routes.

3389 Switch back to voice mode from fax feature needs some improvements

The feature "fall back to voice mode after fax completes" was designed with the assumption that voice is always negotiated before Tenor switches to the fax mode.

With H.323 slow start it is possible to directly negotiate T.38 fax without ever going to the voice mode. In that case, when fax complete event is received, tenor should not try to switch to voice mode as no voice codec is negotiated. This is now the case.

P104-12-13

3098 Access number to be added to the CDR in CDR type 99/199

The Access Number (DID on original call leg) is now included in CDRs in the last column. CDRType 99 or 199 are required for this.

3296 All; IVR Multisession May Not Work Before Call Is Connected

If a terminating endpoint does not support faststart (H.323 calls) or does not send 183Progress with SDP (SIP calls) the Tenor does allocate a DSP until it receives the codec information usually in the Connect message. During this period, it can not collect multisession digits. A DSP has now been allocated to the call if multisession is enabled, and as a result this has been resolved.

3333 Hardware Registration takes too long

Some efficiencies were added to the boot process, speeding up system boot up.

3347 Some calls through a DX have large amount of static

There were issues with bursts of static on DX calls. This was fairly rare, but could be duplicated using certain call patterns. This has been resolved.

3362 BX occasionally boots as DX

Some Tenor BXs were occasionally booting up as DXs, causing all kinds of issues. This has been resolved.

3370 Add support for 1 second delays when using Pulse dial

One can now use comma(s) in the Hopoff replacement field to allow for delays in the dial sequence. Each comma introduces a 1 second delay.

P104-12-12

3093 FXS port to FXS port G.711 "put back" SIP calls crash AX

It was possible to crash a Tenor using G.711 and SIP calls looping back to the same box (for example FXS to FXS). This has been resolved.

3099 RoHS support for CMS

Support was added for new RoHS compliant cards for CMS and Call Relay SP. This was mostly to address display issues in the UI for show -v, etc.

3239 MaxTalkTime is not working if radius sends credit-time

The "MaxTalkTime" value in IPRG was not working if credit-time was being returned by RADIUS. Credit-time was taking precedence. This posed a major problem if the RADIUS server was returning very large times, as many do. A new algorithm was implemented using the shortest of credit-time and MaxTalkTime if both are configured.

3322 SIP: Session timer doesn't disconnect call, if destination gateway reboots before Re-INVITE starts

Several problems were found with session timers in a fairly rare scenario of the far end rebooting before re-invite starts. These have been resolved.

3330 CMS: DSP Status Light always red

The DSP Status light on CMS DS1 card was always red, regardless of error condition. This has been resolved.

3345 Allow code to ignore license when allocating DSP's for Caller ID

There was a problem where the license was being exceeded by DSPs allocated for CallerID. This has been resolved.

P104-12-11

2707 Busy out for T1 & E1

The digital equivalent of the analog "busy out" feature has been added.

It is implemented in var_config.cfg:

gkbusyout "counter"

Where "counter" is a threshold value of RRQ timeout. When RRQ timeout is reached to this counter, Frame loss alarm will be generated on T1 or E1 to protect from incoming T1/E1 call.

See the Command Reference for additional information about the var_config.cfg file.

2835 All; StopAcctID Parameter is Ignored When Start-Accounting is Used

The parameter StopAcctID in TCRG and LCRG decides the value of Use-Account when RADIUS Stop-Accounting is generated. This is the case for all the calls that do not get authenticated. However, if Start-Accounting is used then User-Account always contained the calling-party-number, irrespective of the StopAcctID setting. This has been resolved.

2850 CH crashed when passed FACILITY message to ISDN

The Tenor was crashing when a FACILITY message was received on a passthrough call. This has been resolved.

2872 Need way to route CAS (FGD) calls with no DNIS

In some Feature Group D scenarios it was possible a Tenor may receive no DNIS. Tenor would not be able to route the call, so would reject it. We added a var_config.cfg option to assign a DNIS to these calls. This only applies to Digital CAS calls, mostly applicable to FGD.

var_config.cfg entry:

CASNoDNISRerouteto xxxyyyzzzz

Where xxxyyyzzzz is the (E.164) number to forward to. It will use this number as the DNIS.

See the Command Reference for additional information about the var_config.cfg file.

2891 Caller Name from Facility IE in SIP invite

The ability to receive Caller ID with name in a Facility IE from ISDN, and put it into a SIP Invite message (in the 'From' field) has been added.

2939 Support for h323-ivr-in value=available-funds:14.72 for Porta

Support for a Porta Billing VSA "available-funds" has been added. In Calling Card IVR scenarios we will now recognize and act upon this VSA syntax.

For example:

h323-ivr-in value=available-funds:14.72

2996 Change Default Value of AniInfo for Outbound FGD Calls

This is a specialized feature for a specific customer requirement. This feature, when enabled, will take the first two digits of ANI coming from a SIP INVITE, use those two digits as the ANI II digits in an outbound FGD call, replacing the 00 now used as the hard-coded default. Those two digits should be stripped from the ANI.

This is controlled by a var_config.cfg "FGDreplaceANIinfo". Default is 0 (just use the 00 default), 1 will enable always stripping the two digits and using them as ANI Info digits.

See the Command Reference for additional information about the var_config.cfg file.

3005 Terminating pound (#) in IVR with no dial plan makes dialed number private

There were issues with Tenors configured with DialPlanCountry = 255 (no Dialplan) and customers typing # in an IVR session to terminate their number. Call was routed as a "private" number, and ultimately failed. This has been resolved.

3017 CNAME support for NI2 switch

Support for obtaining Caller ID with name on an NI2 switch (where the name is sent in a Facility message) was added.

3018 SS7 supports original called number, redirecting number and redirection number

CMS running SS7 was not passing through original called number, redirecting number and redirection number from incoming PSTN line to outgoing PSTN line. This has been resolved.

3022 When SIP call doesn't support RFC2833 DTMF Detection is disabled, affecting Supplementary Services & IVR

If a SIP endpoint did not support RFC2833 (DTMF detection) this negatively impacted supplementary services. With this fix, a DSP will now be assigned to monitor the channel for DTMF, allowing SIP supplementary services to work properly.

3041 AS -> CRSP Slow Start calls Fail

Call Relay only. There were issues when an inbound H.323 endpoint used slow-start, and outbound call was SIP. Codecs were not negotiated properly, resulting in no audio. This has been resolved.

3051 community string not being sent in SNMP traps

There was a problem with SNMP traps not properly sending the community string in outbound traps. It was sending "trap community" (a hardcoded default) instead. This has been resolved, outbound traps will now properly send the community string if it is defined in the var_config.cfg variable "SNMPTrapCommunity".

See the Command Reference for additional information about the var_config.cfg file.

3055 SIP: Provisionals without SDP causes incorrect codec negotiation in Call Relay

If Call Relay received 180 or 183 from terminating gateway without SDP, it sent the first configured codec to the originating gateway instead of waiting for SDP in 200 connect. This has been resolved.

3060 LCRG forced routing and Ignorednis changes

The ignorednis feature, mostly used to provide a "hotline" feature on FXS ports has been changed.

• ignorednis no longer applies to TCRG
• ignorednis is now enabled by default if "ForcedRoutingNumber" is configured in a LCRG.

Bottom line, if "ForcedRoutingNumber" is configured in an LCRG, anytime a phone on FXS port) is picked up, the call will be immediately routed using the value configured in "ForcedRoutingNumber". This effectively creates a "hotline" phone.

3069 Call failure when using an external Gatekeeper

There were issues when a call is routed to an external Gatekeeper and routed back to the same Tenor for termination. We were improperly sending 127.0.0.1 (loopback) as the RTP destination. This caused issues. This has been resolved.

3106 Delay login prompt until hardware registration is finished

During boot up, on the console port, the login prompt was presented prior to full system initialization. This could cause annoying issues. The system will now wait for proper initialization before presenting a login prompt.

3107 SS7 can not use Call Routing Server

CMS using SS7 was not able to work with the Call Routing Server. This has been resolved.

3142 Incorrect data in ISDN Setup CNAM for DMS switch

There were some formatting issues sending the Caller Name IE parameter when using the Nortel DMS100. This has been resolved.

3151 Outbound Access level does not properly work with SIP

If OutboundAccessLevel was used with SIP calls, there were issues with re-routing calls upon failure. This has been resolved.

3172 PPPoE Termination Procedure Change

When PPPoE server terminates a PPPoE link, the server sent a PADT to Tenor. Tenor had 2 problems. It kept sending PADT response repeatedly. It also put tags in the PADT, in violation of RFC2516. These have been resolved.

3193 Tenor watchdogs when radius receives corrupted access accept packet

Tenor could crash when it received AccessAccept packet with attribute length = 0. This has been resolved.

3196 DHCP task crashed in case of one DNS server configuration

The Tenor could crash when only one DNS server was returned by a DHCP server. Tenor now works properly with this condition.

3215 AllowOnlyProxyCalls set to 0 after deleting DB

In a fairly rare case of deleting the configuration via ftp, and deleting all but ipconfig.txt, AllowOnlyProxyCalls was set to 0, instead of the correct value of 1. This has been resolved.

3253 BX MFG test broken - second card

The MFG test was failing on a BX with 8 ports. This has been resolved.

3265 Small Tenors Only; Print MAC address in boot up scroll

The MAC address will now be printed on the console during boot.

3271 Unwanted "," at end of the column help

When show is typed in a CLI prompt, an unnecessary comma was printed at the end of the third column help field. This has been removed. Minor display issue.

3277 With Pulse dialing line side mute never gets undone

There were conditions using pulse dial on outbound analog where the line would not be unmuted when dialing is complete. The muting is done to prevent pulses being passed back to the caller, but was not being unmuted. This has been resolved.

3282 Busy out call setup fail by seizure timeout

When the busyout feature was utilized, and a call failed to route via VoIP, it was possible that the attempt on the FXO port would fail to seize the line. This has been resolved.

3301 CRSP crashing on receipt of various H.323/Q.931 messages containing H.245 control messages

It was possible to crash the Call Relay SP if an H.323 Alerting, Connect or Facility message arrived with tunneled H.245 Control Messages. This has been resolved.

P104-12-10

2450 Analog MFG Noise Test - NOT RUN AXM1600

Analog "noise tests" were not being run on AXM1600. This has been resolved.

P104-12-09

3006 new framer driver for Infineon version 2.2

Support for new Infineon framer chip.

3071 New serial number prefixes for RoHS boards

Serial number prefixes were incremented for RoHS compliant systems.

P104-12-08

3049 Flash Write fails when hardware related changes made and power rebooted

Certain configuration changes related to hardware (such as online/offline) may not be saved, so will be reverted upon reboot. This problem was introduced in P104-12-04 and resolved in P104-12-08.

P104-12-07

3042 Tenor crashes when changing HND replacement pattern to more than 15 characters

A crash may result when making a change to an existing HND replacement pattern in excess of 15 characters. This problem exists in all previous P104 code, and is resolved in P104-12-07.

3045 Dropped calls due to Session Timer not cleaned up properly

A memory leak may result from the use of the SIP Session Timer functions. This has been resolved.

P104-12-06

2837 Change reversed

Bug 2837 added in P104-12-03 was removed in this release as it would cause problems to other customers.

P104-12-05

3039 BX only - SIP FAX issue -Incorrect T38MaxBitRate encoded

The SIP T38MaxBitRate SDP parameter was being mis-coded on the BX product, causing fax problems. This has been corrected.

P104-12-04

2804 Provide a temporary solution for Telnet Linemode Problem

Certain Linux, BSD and other Unix-like OSs were having trouble telnetting to the Tenor. The symptom is the session is terminated immediately after the login prompt is displayed.

This was especially the case when Gnome Terminal was used. The Tenor telnet server was not properly decoding eol when it was set to 255 (0xFF) during linemode negotiation. As a temporary solution, linemode will not be negotiated. A more permanent fix is forthcoming, likely in P105 code.

3010 exception "can not send bye"

An exception "can not send bye" was being printed in certain call flows in test. Though it is not clear if this would cause any field problems, it may cause a memory leak which eventually could definitely cause odd and hard to identify problems. The memory leak was corrected.

3030 System Test scripts leads to buffer pool depletion and DX would reboot.

Certain stress tests in system test could exhaust the allocated memory buffer pools. Though it is unlikely these conditions would exist in the real-world, more memory buffers were allocated as a precaution.

P104-12-03

2942 Japanese Caller ID Could Cause Disconnect

If Japanese caller id was enabled and a passthrough call was answered on the first ring the call was disconnected. This only applied to Japanese Caller ID and only with certain telephones. This has been resolved.

1952 BX : NTT PBX doesn't need called party IE in some cases

In certain cases some NTT PBXs will have problems if the DNIS (CallED Party) is sent. To overcome this a var_config.cfg parameter was added to disable CallED Party on IP calls.

NTTrelayDNIS 0 will disable the sending of CallED Party on IP calls on the BX in NTT protocol. The default is to send CallED Party.

See the Command Reference for additional information about the var_config.cfg file.

2745 TCRG Passthrough 2 not presenting IP route

In previous P104 builds, where PassthroughEnable 2 was configured, IP routes were not being checked, and all calls were only sent to matching LCRG routes. This has been resolved.

2749 IPRG parameters not taken in count for Supp Services

DigitRelaySIP, SIPDigitRelayPayloadType, PacketSaverEnabled in IPRG were not being observed on transferred calls. This has been resolved.

2812 ALL; Being able to reset unit from CLI and FTP reliably

In very rare circumstances the system will not reboot when commanded to. To address these rare cases a more direct method was added to forcibly reboot the system. FTP into the unit from a command line ftp client and execute this command "get resetnow.sys". The system should reboot.

2843 SIP Secondary Server: Registration Request to Secondary always picks Primary User Name

In circumstances where both a primary and secondary proxy has been configured, and the primary fails, the system was still using the username from the primary. This was not an issue if the username was the same for both (as it usually is), but would cause a failure if they were different. This has been resolved.

2844 SIP: UA: Adding more UAs uses consecutive increments for Listen Port Number

Documentation said that new User Agents will increment the listen port by 2. That is incorrect, it increments by 1. Doc has been corrected.

2854 Several SIP responses do not map to cause codes properly, 500 501 502 503 422

When ISDN calls are sent via SIP, SIP Response Codes must be mapped to appropriate ISDN cause codes. This was not being done correctly for several SIP response codes, resulting in unnecessary re-route call attempts and other oddities. The following were changed to map correctly:

• 422 ClientErrorSessionIntervalTooSmall to cause 125
• 502 ServerErrorBadGateway to cause 27
• 500 ServerErrorInternalServerError to cause 111
• 501 ServerErrorNotImplemented to cause 29
• 503 ServerErrorServiceUnavailable to cause 88

2890 Cannot save to text db - any configuration change lost on reboot

If the system database and the actual hardware installed do not match, writes to the database will not happen, causing configuration changes to be lost upon reboot. Though this is unlikely to happen in the field, it was corrected anyway.

This does NOT apply to CMS or Call Relay SP.

2887 Implement Brazil and India Progress Tones

Support was added for the progress tones for India and Brazil.

• 11 Brazil
• 12 India

2905 CMS; Channels hangs under heavy call rate (E1-R2)

Certain signaling conditions, and heavy call load, can cause us to miss state changes in R2, potentially causing channels in hung state. This has been resolved with more frequent polling of AB bit states.

2914 All analog - change fwd disconnect trigger

Forward disconnect functions were made a bit more sensitive to allow for better detection.

2934 Ring trip broken with some SLIC revisions

Certain analog FXS port versions will not always detect the off-hook condition when answered during ringing. This was somewhat rare. This has been resolved.

2936 SIP, BYE authentication fails with Vonage proxy

It was found that some proxies will attempt to authenticate BYE messages. We had problems with this on parallel forked calls. This has been resolved.

2946 DX: Card type mismatch for 6 DS1 Tenor DX.

A "Card type mismatch" alarm was being generated for 6 DS1 Tenors. This was causing a series of problems, including the inability to save configuration changes. This has been resolved.

2965 Taiwan Static Dial plan

03nxxxxxx, 05nxxxxxx, 06nxxxxxx, 07nxxxxxx, 08nxxxxxx (where n=2~9, x=0~9) patterns were not working in the Taiwan static dial plan. This has been resolved.

2999 Call Waiting causes crash after first call disconnects

B calls A. While B and A are talking, C calls A. While the second call is in ringing state, B hangs up. When A tries to produce the Call Waiting tone after B hangs up, A crashes. This has been resolved.

P104-12-02

2753 DX resets when working with Routing Server

DX may reset when using the Routing Server. This has been resolved.

2857 Memory leaking on SIP Redirect call

Under heavy load, with a specific home-grown proxy, Tenor would leak memory and eventually crash. The cause was determined and resolved.

2877 Add the version number info in reset.log file

To ease troubleshooting, the version of code running has been added to the reset.log file. Note, it will show the version running on boot, so this may be misleading if the software version was updated before rebooting.

2881 CMS crashes when it runs out of memory buffers

We found some conditions where the CMS will exhaust it's allocated memory buffers. Buffers were increased and the problem was resolved.

2885 DTMF payload values are reversed

In P104-12-00 there was a feature added to allow for asymetrical DTMF payload. This might occur during SDP negotiation. Though we added the functionality, the result was the values were reversed. This has been resolved, it is now working as designed.

P104-12-01

2398 Remote call id is not parsed properly

There was a problem where extremely high Call IDs (an internal call identifier) over 0x7FFFFFFF were not being parsed properly, possibly causing call drops. This was fairly rare, but has been resolved as it may occur.

2423 SIP Remote-Party-ID headers need to be supported

Some encoding/decoding fixes to Remote-Party-ID headers included in this build. Some issues remain with transcoding these between ISDN and SIP.

2734 Hung calls for AttendedTransferKeystroke configured to non 'HU'

If Tenor’s “AttendedTransferKeystroke” was configured to something other than ‘HU’ (ex. #22), hanging up the phone instead of entering the transfer command resulted in hung calls. This has been resolved.

2741 Refer-To header does not follow IPDialPlan configuration

‘Refer-To’ header in REFER message did not follow configured IPDialPlan. For example, if IPDialPlan was configured to delete first 4 outgoing digits and replace them with something else, ‘Refer-To’ header contained the original dialed digits without replacement. The transfer failed because number pattern was not matched by the transferred party. This has been resolved.

2762 CallID of SIP Register message is the same after a reset as before

The SIP CallID assignment algorithm was not random enough. The chances of the first registration attempt using a previously used CallID were very high, especially on AS/AF units with no real-time clock, as the CallID was based on time. The CallID algorithm is now considerably more random, not strictly time-based, making this almost impossible to happen.

2770 Transferred call is being aborted before Connected

In certain cases a transferred call might get Progress (alerting) while the call was technically connected, causing the call to be dropped. This situation is now being dealt with properly and the call is not dropped.

2772 SIP: ptime should be added only once per media attribute

The ptime attribute was being added once per codec, as opposed to once per media. This has been corrected and is now only added once per media.

2773 CMS,CR-SP; sr command or iprg command causes eventual crash

Certain configuration changes, especially those related to Static Routes and IPRGs could cause memory corruption over time, causing crashes. This has been resolved.

2779 The IrDA receiver on the CMS should be turned off

On some CMSs the IRDA port was enabled, causing spurious garbage on serial console terminal sessions. The port has now been completely disabled.

2781 When MICA exceeded, incoming SIP calls not allowed

If the MaximumInboundCallsAllowed (MICA) feature was enabled, and the total number of SIP calls exceeded the configured value, all future SIP calls using that IPRG were being rejected until the system was reset, or the value changed. This has been resolved, SIP calls now observe the MICA value properly, and do not accumulate causing call failures.

2784 Authentication with Asterisk broken

It appears that Asterisk requires a space before the word "nonce" in the authentication headers. The spec does not agree with them, but we believe that adding the space back is benign and will solve this problem, and break no one else. This was not a problem in P103 code (we had the space), but the space was removed in P104-12-00, breaking Asterisk. We now send the space character, and now authenticate fine with Asterisk servers.

2788 All files and folders disappear after "show -v" command

File handles were not being released after running the "show -v" command. Eventually, no more file handles were available. This would not affect general function, or cause a crash. But, it would cause some errors on commands, you would not be able to ftp to the tenor, and configuration changes would not be saved over a reboot, amongst other problems. This has been resolved.

2792 CMS/DX; OutboundCallDetect not working when media is not established on IP side

One way voice problems may occur when using the OutboundCallDetect feature set, if the originating gateway does not establish media prior to connect. This has been resolved.

2795 Voice not heard when BYE received early in transfer

There were situations where a failed unattended transfer (new call leg is rejected) that the original callers may lose audio. This has been resolved.

2801 Session Timer with Transfers causes one way digit transmission.

Problems with the session timer feature, and transferred calls could cause RFC2833 DTMF transfers to fail in one direction. These cases have been resolved.

2803 StaticChannelConnection disappeared after CMS reboot

In CMS changes to the StaticChannelConnection value were not being written to flash, so were lost upon reboot. This has been resolved.

2813 Tenor should generate an Alarm when the Proxy/Registrar is unavailable

A feature was added to generate an alarm, SNMP trap and exception if the proxy is not reachable.

2815 CR60/CRSP only: 2833 packets are not dropped when passed to H.323

When DTMF packets are converted from 2833 to H.245, original packets are not dropped. In most of the cases, since the other end is H.323 GW, these packets will be dropped. But in cases where the H.323 GW interpretes both H.245 and 2833 packets, it will generate two tones. This has been resolved, RFC2833 DTMF packets are extracted from the RTP stream when it has been converted to H.245.

2831 BX: DI law set back to A-law after reset

If the companding law was changed in a BX to mu-law (for Japan primarily) it was not saved in flash, and the change lost upon reboot. This has been resolved. Note, you must make the change once you have updated to this code for it to be written, so ensure the change is made and submitted after upgrading the system software to this version (or later).

2833 "disconnect" command: remove need for 0x prefix

The disconnect command no longer requires "0x" to be prepended to the call ID. Also, some cases where this command did not work were resolved. There will still be certain calls, especially some digital cas types, that may not get completely disconnected with this command, as the signaling does not allow it.

2853 ivrtype 4 call gets connected in invalid authentication

In ivrtype 4 used on an LCRG, when a call failed there was still a connect sent to the originating device. In some cases this could cause billing issues. This has been resolved.

2858 Change factory default for AllowOnlyProxyCalls

The factory default value for AllowOnlyProxyCalls has been changed to 1 (enabled). This will not affect upgrades, it will only change the factory default. It is recommended that all users change this value to 1 if all inbound calls are only coming from the configured proxy as this will add considerable security value.