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Tenor VoIP MultiPath

Switch/Gateway Products
P103-08-00 Release Notes
This document lists features and enhancements, as well as resolved and open inconsistencies, for
the following products running software version P103-08-00:

Tenor DX VoIP MultiPath Tenor AX VoIP MultiPath

Switch/Gateway
Tenor DX 2008/ Tenor DX 2016
Tenor DX 2024/ Tenor DX 2030
Tenor DX 2048/ Tenor DX 2060
Tenor DX 4048/ Tenor DX 4096
Tenor DX 4060/ Tenor DX 4120
Tenor DX 6120/ Tenor DX 8120
Switch/Gateway
Tenor AXM0800/AXM1600/AXM2400
Tenor AXT0800/AXT1600/AXT2400/
AXT4800
Tenor AXG0800/AXG1600/AXG2400
Tenor AXE0800/AXE1600/AXE2400

Tenor AS VoIP MultiPath Tenor VoIP Carrier MultiPath Switch

Switch/Gateway (CMS)
Tenor ASM200/ASM400 Tenor CMS – 2 slot Chassis
Tenor ASG200/ASG400 Tenor CMS – 8 slot Chassis

Tenor BX VoIP MultiPath

Switch/Gateway
Tenor BX204
Tenor BX408
Tenor BX816

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1.0 Interoperability
Note: Quintum Software Version P103-08-00 is compatible with Broadsoft's products, and many
features are fully-functional, but it is not the version that has been validated by Broadsoft against
its Test Plan, to achieve its recently announced certification. That version of firmware will be
available soon.

The Tenor DX, Tenor AX, Tenor AS, Tenor BX, and Tenor CMS running P103-08-00
interoperate with Tenor Configuration Manager, CM103-07-02.

Tenor Monitor v2-0-2 interoperates with Tenor DX, AX, and AS. We do not recommend you use
the Tenor Monitor on a fully loaded Tenor CMS or Tenor CR-SP system.

Release P103-08-00 system code does not interoperate with Tenor CMS Series 1 hardware (System
Controller, DSP, T1/E1 cards).

When upgrading your unit, we recommend you install all .jbc files. There is a specific .jbc file for
each product, (i.e., tnrax.jbc is required for the Tenor AX product). When all .jbc files are
installed, the Tenor will then automatically select the required file for your unit. Also, the
dxdspb.bin file will be required only when installing the Tenor DX with 120-channel DSP card.
See http://www.quintum.com/support for all required files. Select the correct zip file for your
product and this will be taken care of for you.

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2.0 New Features/Enhancements
The following section describes new features/enhancements that apply to all Tenor products; if a
feature is applicable to a specific product, it is noted.

NOTE: Many new features include new configuration fields, which are available from the
Configuration GUI and the CLI. This section includes the names of the fields, as well as brief
definitions. For more specific information, see the Command Reference Guide, which can be found
on the User Documentation CD you received with the unit or at
http://www.quintum.com/support.

64 Additions/Changes to Accounting Types

Two new accounting types have been added to the AccountingType option (available under
RadiusInfo User Server): one start/one stop accounting per call and two start/two stop accounting
per call. In addition, the UpdateAccountingType and UpdateAccountingInterval options
(available under RadisInfo User Server) must be populated except when the AccountingType is
disabled.

459 Added ability to reset the unit to factory defaults from console
In order to reset the password without logging in, or if you have forgotten the password, the unit
can now be reset to factory defaults from the console.

654/1155/1681/1955/2050 DialPlanCountry and ProgressToneCountry added/updated

A new field, DialPlanCountry option (available under DialPlan) has been added to select the
country (or set of dial plan rules) where the Tenor is located. This automatically configures the
Tenor with the international dial plan for the specified country. This option includes: 0 - Generic
CCITT (default), 1 - US/NANPA 7-digit local numbers, 2 - US/NANPA 10-digit local numbers, 3
– China, 4 – Taiwan, 5 – UK, 6 – Australia, 99 - Custom - User Program Dial Plan (UPDP), 255 -
No dial plan/None. In addition, ProgressToneCountry options include: 0 – US, 1 – China, 2 –
Taiwan, 3 – Japan, 4 – Poland, 5 – Yugoslavia, 6 - UK, 7 – Australia, 8 - Korea, 9 - South
Africa, and 10- Israel.

This field must be set to Custom (99) in order to activate the User Program Dial Plan.

724 Remote NAT feature added

The Remote NAT feature allows you to set the Tenor to ignore the RTP (Real-time Transport
Protocol) IP:port information in the signaling, and instead uses the received IP header (in the RTP
packet) to derive the destination.

A new command, RemoteNAT, has been added to the Gateway prompt. This new command
enables or disables the Remote NAT feature.

893 IVR type 2 with Pin

IVRType 2 requires the card number and plays a credit amount and time, and IVRType 3 requires
account and pin, and does not play anything. A new feature has been added for a prepaid server
with card number and pin, including a prompt for credit amount and time. For this purpose, the
IVRPinNumLength (available through TCRG and LCRG) is available when using IVRType 2.

1065 DigitRelaySIP includes two new options

Two new options have been added to DigitRelaySIP (available under IPRG): RFC 2833 + SIP
INFO (processes both RFC 2833 and SIP INFO) and SIP INFO (processes only SIP INFO).

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1066/1084 Ability to accept calls only from a configured proxy
The ability to accept calls only from a configured proxy has been added to the Tenor through
AllowOnlyProxyCalls (available through SIPSG).

1072 DNS Resolution

You will now be able to enter a url (i.e., www.quintum.com) instead of the IP address (i.e.,
192.168.2.55) for the Primary SIP Server.

1082 Max number of forwards enabled

A new field, maxforwards (for CLI) and maximum forwards (for Configuration Manager), has
been added to SIPSG. This command sets a maximum number of forwards for a call to prevent it
from being looped.

1169 Turkish IVR supported

Turkish IVR is now supported. You can set IVR to Turkish by setting IVRLanguage to 9. This
can be configured through the Command Line Interface (CLI) only; it is not available through the
Configuration Manager.

1238 Answer Tone feature added

A new feature has been added to the Tenor to support AnswerDisconnectTone on connect and
disconnect, which enables an answer tone using DTMF C-tone. Through AnswerDisconnectTone,
you are able to set whether the tone should be disabled, on answer only, on disconnect only, or both
answer and disconnect. In addition, there is a new command AnswerToneDuration which sets
the number of milliseconds to play the AnswerTone, when that feature is enabled.

1335 SIP codec negotiations improved

Codec negotiations have been improved when using SIP.

1406/1204 Added H323ID to CDR

Added H323ID, trunkgroupin and trunkgroupout fields to CDR formats 199/99.

1433 IVR Type 11 support added

Support has been added for IVRType 11, which supports Prepaid and Post paid calling card at the
same time.

1470 Detection/Generation of callerid with name

Support has been added for the detection/generation of callerid with the name in the IE. To
incorporate this feature, you must include the configuration information as part of the
var_config.cfg file. If you are not familiar with the var_config.cfg mechanism, please go to
Quintum’s support site at http://www.quintum.com/support and enter the search word
var_config.cfg in the search box.

The configuration options that should be included in the var_config.cfg file are as follows:
display-ie followed by 0 (No DisplayIE, default), 1 (Passthru DisplayIE to IP or PSTN Intf), or 2
Passthru w/User Defined Text (string to follow) to IP or PSTN Intf.

For example, type the following:

display-ie 1

1471/1676 Ability to set different RelayANI option

Added a new value to RelayANI in the ISDNSG and IPRG configuration menu options. When
set, this option means that ANI is relayed, but uses the TON and NPI values from DefaultANI
TON and DefaultANI NPI (both fields are located in the ISDNSG and IPRG menu options).

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1562/1795 SS7 support added
Support for the SS7 protocol has been added. Two new configuration groups have been added:
SS7 Gateway and SS7 Signaling group. Each of these new groups have corresponding
configuration fields.

1557 Screening Indicator is now configurable

The screening indicator, available in IPRG and ISDNSG, is now configurable. A new field,
DefaultANIScreenInd, along with a number of configuration options, are available to support the
new feature.

1579 When IP address changes, Tenor does not need a reset

For PPPoE only, when the Tenor has a dynamic IP address, the IP address can be changed by the
ISP at anytime. Once the IP is changed, the Tenor will not reset immediately following the change.

1619 ANI category on R2 into RADIUS messaging

For incoming R2 calls, this ANI Category will now go out in the RADIUS authentication packet
VSA 1. The attribute value will be in the format:
h323-ivr-out=ANI_INFO:<n>
where <n> is the ANI Category digits.

1637 Added ability to separate channels out when IP Extension is configured

You are now able to use some channels for IPExt and other channels for a normal call.

1645 Enable/disable IVR multisession through var_config.cfg

This new feature allows you to enable/disable IVR multisession through the var_config.cfg file. If
you are not familiar with the var_config.cfg mechanism, please go to Quintum's support site at
http://www.quintum.com/support and enter the search word var_config.cfg in the search box.

The following option is needed in the var_config.cfg file:

ivr <accessnumber> 2, 3 or 6 (ivrtype) <customer service number>

<preauthentication> <balance> <time> <multisession key>

For example, ivr 18476669851 2 - 1 0 1 1 ##

1656 Multiple GWs behind firewall now supported

Multiple gateways behind a single firewall are now supported.

1660 Remote Tenor Manager Info feature added

A new Remote Tenor Manager feature enables you to configure the Remote Tenor Manager
application. This manager enables you to manage a Tenor behind a firewall; it facilitates the
communication between the management client and the Tenors being managed. As a result, it
performs the tunneling required to traverse the firewall.

A new configuration group RemoteTenorManager, had been added for configuration along with
options such as PrimaryServerIP, SecondaryServerIP, ConnectTimer, and Udp.

1671 Ability to send Public or Private Caller ID based on digits dialed

A new feature has been added to the Tenor to send Public or Private Caller ID based on digits
dialed. To enable this feature, you must include the configuration information as part of the
var_config.cfg file. If you are not familiar with the var_config.cfg mechanism, please go to

P/N 480-0071-00-03 Page 5

Quintum’s support site at http://www.quintum.com/support and enter the search word
var_config.cfg in the search box.

The varconfig format is, "selectani <enable/disable flag> <maxdigits>"

where: 0 (enable flag), 1 (disable flag)
maxdigits = any number within reason (WARNING there is no bounds checking here)

The default settings are feature enabled with maxdigits = 7.

1673 Ability to remove Quintum header from messages

You now have the ability to remove the Quintum header from messages. A new command,
IncludeQuintumHeader has been added to SIPSG for configuration, which can be disabled (0) or
enabled (1).

1677 Packetsaver is now working properly in all NAT scenarios

Changes have been made to the Tenor to work in all NAT scenarios (Tenor behind firewalls).

1685 SIP unregister ability added

For SIP, in order to prevent "confusion" on certain registrars, there is now support to "unregister"
all old information when changing and re-submitting any registered values. This is done by setting
the RegisterExpiryTime (via SIPSG) to 0.

1692 Added SIP Call Id to Radius messages

SIP Call Id has been added to RADIUS Stop-Accounting, Update-Accounting, and Start-
Accounting messages.

1694 Number of IPExtension Port mapping calls increased for Tenor DX and Tenor CMS
Tenors can now handle more than 62 port mapped calls. It is possible to program up to 120 port-
mapped IPE connections in a Tenor DX, and up to 960 in a CMS.

1695 SIP Number "Rollover" Routing

Support for the feature of “rollover” routing has been added to Tenor. When Line/Channel #1 is in
use (off hook), one of the other Line/Channels receives a second call to the main number
successfully. In addition, some of these channels will not receive these rollover calls (e.g. FAX
Machine, CEO, etc.)

To incorporate the “rollover” routing feature, you must include the configuration information as
part of the var_config.cfg file. If you are not familiar with the var_config.cfg mechanism, please
go to Quintum’s support site at http://www.quintum.com/support and enter the search word
var_config.cfg in the search box.

The configuration options that should be included in the var_config.cfg file are as follows:

InboundSIPSGMatching 0 = (default). Inbound SIP calls will only match the number before
being routed to an xCRG.

InboundSIPSGMatching 1 = If set to 1 using var_config.cfg, inbound SIP calls will match the
number (LDN) and as well as the SIPSG, before being routed to an xCRG.

1702 New ping ability

You are now able to ping for the domain server name as well as an IP address.

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1725 Payload now configurable
A new configuration item in IPRG, SIPDigitRelayPayloadType, will enable you to configure the
Payload Type to be sent out in its SDP (when the value is changed from 101).

1745 SIP username as the URI in authorization header

You are now able to configure the SIP username as the URI in the authorization header. Through
the UserNameAsURI configuration option (part of SIPSG), you are able to enable or disable
whether the user name in URI format.

1747 SIP number of Request Retransmits are configurable

Through a new command, RequestReTransmitCount, you can now configure the number of
Request Retransmits (3 through 18, default 6). This configuration item is available under SIPSG.

1780 Supplementary services supported

For CAS calls, the Tenor now supports supplementary services for the following: Hold/Off Hold,
Unattended Transfer, and Attended Transfer. For configuration, several new options are
available in LCRG and TCRG CLI groups: Hold, HoldKeystroke, UnattendedTransfer,
UnattendedTransferKeystroke, AttendedTransfer, AttendedTransferKeystroke, and
CallHangupKeystroke.

1806 Database file changed

The feature changes the binary current_db database into a ASCII-based db.txt. The ASCII-based is
now user readable and can be easily debugged.

1815 Option in SIP INVITE message is now configurable

A new setting, ProxyAddrInFromHdr, (available under SIPSG) allows the user to determine if
the IP address of the Invite's From header's SIPS URI Host is the configured Proxy IP (default) or
the Tenor’s own IP.

1823 Put-thru feature added on receipt of SIP Refer msg

The following feature has been implemented: when a call is up and the SIP side receives a Refer
with a refer-to address which points back to our own gateway, instead of initiating a new SIP call
(which would loop back to the Tenor), the incoming phone number in the refer-to is sent to the
Tenor’s routing engine and a new call is made. This eliminates a SIP leg.

1833 Reverse Battery Improvements for glitches (Tenor AS, AX)

Improvements have been made to Reverse Battery to fix some glitches with the voltage.

1888 Outbound Proxy Name support in SIP

The Tenor now supports Outbound Proxy Name.

1904 DNS Server through DHCP Enabled

The ability to obtain DNS servers via DHCP has been enabled in the Tenor through a new
command, UseConfiguredDnsServers. This command is included in a new set of prompts under
DNSServer, which includes PrimaryServerIPAddr and SecondaryServerIPAddr (moved from
the EthernetInterface prompt).

1924/1889 Increased efficiency for SIP calls

Several changes have been made to the SIPSG parameters. MediaInProvisionalResponse is now
named SDP180Ringing, InviteFromUriHost is now named ProxyAddrInFromHdr. Other
SIPSG changes are listed with their respective features in this document.

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1928 Display IE (Description Information Element) is now disabled by default
"Display IE" was recently added to allow for CallerID with name. Some protocols would fail if
this was sent. Now, by default, the Display IE will not be sent.

1941 T310 Timer added

A new T310 timer (available in IPRG) is a new option added for when an ISDN system receives a
Call Proceeding message. If no Alerting, Progress, or Connect message is received within the
duration of the T310 timer, the call clears. This commend enables you to configure the T310
timeout for the IP Routing Group, which may be required for compatibility with some switches.

1943 Added Call Waiting feature

Added the ability to enable/disable Call Waiting through a new command, CallWaiting (available
through LCRG and TCRG). When this feature is configured, when one call is active on a CAS
channel, a second incoming call for the same channel will not be rejected. Instead, a call waiting
tone is generated. Call waiting is only supported for an incoming SIP call. If the 2nd call is a
circuit call, the call waiting feature will not be attempted. For call waiting to work, both first and
second calls have to be SIP on one side and CAS on the other side.

1964 DefaultANIPresentationInd
The DefaultANIPresentationInd (available in IPRG and ISDNSG) relates to the presentation
indicator (CLIR bit) in the Calling Party Information Element in the outbound H.323SETUP
message. There are three options available: 0 - (default) Relay from incoming leg of the call,
1 - If no ANI is available the CLIR bit it set to restricted. When ANI is available, use the value
from the incoming leg, and 2- If ANI is present, send it with CLIR bit set to restricted. If ANI is
not available set CLIR bit to restricted.

1965 Disable SDP in Progress message now supported

The ability to disable SDP in the PROGRESS message had been added to the Tenor through the
SDPin183Progress command, available in SIPSG.

1966 Added PIN Retries and DisablePrompt Parameters

For IVR calls, the ability has been added (through IVRRetries) to specify how many times a caller
can re-enter their account number/card number/PIN number before getting disconnected. In
addition, a parameter has been added (IVRDisablePrompt) for you to disable playing the credit
amount, credit time or both. Both of these parameters are available through the LCRG and
TCRG.

1970 Onhook voltage changed (Tenor AS and Tenor AX only)

The Tenor’s on-hook voltage was 45V, which has been changed to 48V.

2011 Added UDP Registration to Reverse Telnet

In order to eliminate a multitude of open TCP connections between the
TunnelServer and Tenors that have Reverse Telnet enabled, a UDP Registration Packet is now sent
periodically to the TunnelServer from the Tenors, instead of establishing TCP connections.

2026 Inbound IPRG based on Media GW or Proxy

Selection of inbound IPRG based on either Media GW or Proxy (Signaling GW) has been made
configurable using var_config through a new option IPRGByProxy. The default entry is 0.

To incorporate the IPRGByProxy feature, you must include the configuration information as part
of the var_config.cfg file. If you are not familiar with the var_config.cfg mechanism, please go to
Quintum’s support site at http://www.quintum.com/support and enter the search word
var_config.cfg in the search box.

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The varconfig format is IPRGByProxy 0. When set to any value greater than 0, inbound IPRG
will be selected based on Signaling GW (Proxy)IP address.

2057/1834 Relay Calling Name feature added

A new feature, Relay Calling Name, has been added. This is available in ISDNSG, CASSG, and
ISDNSG. This setting controls the relay to the outbound leg of the Calling Name received from the
incoming leg. The Calling Name has meaning only when it is combined with the Calling Number
(ANI). Two fields support this feature: RelayCallingNAMe and DefaultCallingNAMe.

2070 Calling Party ID support in DASS-2 protocol (Tenor DX only)

Caller ID now passes through DASS-2 trunks in both incoming and outgoing DASS-2 calls.

2188 New feature enables you to determine number of digits from caller ID to send out
A new field, OutboundANINumDigits, has been added to the LCRG and TCRG directories.
Using this field, you can determine the number of digits from the caller ID to send out.

2297 New feature to set IP at bootup

A new feature has been implemented to eliminate the need for the comport when logging into the
Tenor for the first time. The default configuration is DHCP (Dynamic Host Configuration
Protocol) enabled. If DHCP fails, the Tenor gets an IP address of 169.254.xx.yy, where xx and yy
are the last two bytes of the serial number (in decimal). The discovery function in the
Configuration Manager Wizard enables the Tenor to “discover” local Tenors and identify the
associated IP addresses.

If you have a Com port, when you log into a unit through the COM port at bootup, you now have
the option either to restore factory defaults or to configure the static IP address to the Tenor. When
you initially connect to the unit through the COM port, the following message is displayed,
“WITHIN 4 SECONDS, PRESS ‘r’ FOR FACTORY DEFAULTS or ‘i’ to SET STATIC IP”.

See below for information on both.

Set Factory Defaults - Select ‘r’

If you select ‘r’ within 4 seconds a message will ask you to confirm. Type y. The
configuration will be set to factory defaults and the IP address related information will
remain unchanged.

Set Static IP - Select ‘i’

If you select ‘i’ for Static IP, you are then given the option to configure the IP address,
Subnet Mask, and Default Gateway as follows:

• Select ‘i’ within 4 seconds. A message will ask you to follow the prompts to
configure the IP information.
• Login to the system. The default user name/password is admin.
• The current IP address, Subnet Mask, and Default Gateway are displayed. A
message asks you whether you would like to change these. Click y (yes) or n (no).
• If you click y, follow the prompts to change the ethernet information are displayed.
A message is displayed to tell you the Ethernet interface has been successfully
configured and the unit will be restarted.

1749 StaticChannelConnection added

A new option, StaticChannelConnection (available under DigitalInterface) enables you to map
channels to static channel connections. This only applies to SS7 configurations.

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3.0 Resolved Inconsistencies
The following section describes resolved inconsistencies which apply to all Tenor products; if a
resolved inconsistency is applicable to a specific product, it is noted.

The Resolved Inconsistencies included in Software Release Notes for Software Patch Release
Tenor AS/AX/BX/DX/CMS P102-11-00 (and all previous patch releases), have been included in
this release. See http://www.quintum.com/support to view this document.

699 Incoming RAS messages not validated

There was a problem with the way the Tenor was handling primary/secondary GK failover.

834 Tenor did not recognize channels as offline.

Since the Tenor did not recognize channels as offline, calls were still being delivered to the
channels and the calls would ultimately fail.

1262 Clicking noise heard when making an outgoing call (Tenor AS/AX only)
A clicking noise could be heard right before ringback started when making an outgoing call.

1264 DTMF change required reboot to take effect

When you changed DTMFONTime or DTMFOFFTime, after the submit, you were required to
reboot the box for the change to take effect.

1321 Manufacturing test failed for certain line type (Tenor DX only)
MFG test failed when LineType was set to E1.

1378 EXCP message appeared after command (Tenor BX only)

AN EXCP message would occur after executing a show –v command when connected to a Tenor
BX.

1485 CANCEL message did not have same CSeq # as INVITE

CANCEL sent from Quintum was always sending CANCEL w/CSeq# 1 even though the INVITE
was intended to cancel had a different number.

1488 Extra and Invalid Stop accounting message sent to Routing Server.
In Externally routed call (using routing server), the Tenor should have only sent a one stop
accounting from the termination leg of the call. Tenor sent stop accounting from the origination
leg, so if the first route failed, it would send the stop accounting to the routing server and the
routing server treated it as the end of call, and then the second route request was thrown away.

1489 The # character could be only configured from CLI

For UPDP (Digit Manipulation Translation), the '#' character could only be configured from CLI
and not from Tenor Configuration Manager.

1501 Factory default made the channels vanish (Tenor AX only)

When the Tenor AX was put into factory default, the database options to enable all the channels
(in sl sl2) were not present.

1503 Needed to interpret Call Proceeding with Progress IE as alerting

It has been found that in some cases, particularly on BRI, that for some non-ISDN destinations,
alerting was not sent. Instead, the Call Proceeding had a Progress IE with "Call is not end-to-end
ISDN".

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1509 Ext IP Addr was not populating startH245/Facility message
In the Facility that was sent to start the H245 Channel (tunneling false), the h245 address was
private instead of public.

1520 Ethernet Interfaces 2, 3, and 4 should not have been configurable (Tenor CMS only)
For the Tenor CMS, Ethernet Interfaces 2,3, and 4 should not have been configurable. These ports
were intentionally not operational; any configuration made to these ports would have been invalid.

1524 Multi session IVR calls did work when a Tenor was configured to use the Routing
Server
When using the Tenor with the Routing Server, multi-session calls could not be made. The call
went through, but when user entered ##, the call failed and went back to the IVR prompt.

1527 SIP parameter caused fast busy

When the SIP response contained the maddr parameter in the Contact header, response messages
did not get back to the correct destination and the call did not complete, which resulted in a fast
busy.

1531 Connection failure for NTT switch

When the ISDN switch type was set to 4 (NTT) and the orientation was set to user, under certain
circumstances, the call would have been disconnected.

1548 Quintum crashing when Connect rcvd and Facility sent

Tenors crashed after receiving a connect message from certain gateways. Tunneling was OFF in
these cases, and after getting the connect, the Tenor was sending a Facility w/startH245.

1551 Unnecessary exception occurred in Tenor (Tenor BX only)

An exception was occurring in the Tenor BX for CAS Signaling group, which did not exist. The
exception was: Failed to get CAS SG indices from DB.”

1576 Filter IP Directory was not working with multiple directories

When using multiple Filter IP Directories, the last directory is the only one that worked.

1583 Remove 800ms delay on phone side when set for flashhook (Tenor AS, AX only)
If flashhook was sent, when a disconnect was received, CAS started a timer to make sure the
disconnect was longer than the max flash hook time. After that expired, it unnecessarily started a
delay for 800ms.

1584 Literal * in Static Route Caused Exceptions

Exceptions were seen when the literal * (not the wild card * at the end) was used in static routes.

1585 Problem with how Tenor sent voice packets

There may have been problems with how the Tenor sent voice packets. For example, if the
VoiceCodec-1 in the GUI was set to G723.1 and the Voice Payload Size to something other than
30mS, the Tenor would always send 30mS.

1590 DMT feature with SIP failed

The DMT feature with SIP failed.

1591 Media stream failed to adopt the new IP

When a gateway's IP changed, media stream failed to adopt the new IP.

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1598 Tenor may have shown framing loss after reboot (Tenor CMS and DX only)
When configured for CAS (E1-R2), the Tenor may have shown framing loss after reboot and the
line never synched up.

1599 Radius user length should have been greater than 2

According to RADIUS spec, the User-Name string should have had something in it. If there is no
data, a null character should have been used.

1609 No dialtone with specific phone type (Tenor BX only)

The user would receive no dialtone when Siemens Profiset 70ISDN phone was connected to a
Tenor BX.

1617 MoIP did not work

MoIP did not work because of internal problems.

1632 Signaling Only calls did not work when Routing Server was used to Route calls
Codec negotiation was broken for Signaling only/Routing Server combination.

1651 Pulling out RJ11 line did not trigger Fwd Disconnect (Tenor AS/AX only)
Pulling out RJ11 line from PSTN port did not reliably trigger a forward disconnect (if configured).

1653 CAS crashed when a call was disconnected after dial tone detect
CAS crashed if a call was disconnected immediately after the dial-tone was detected.

1661 SIP could not accept ack with SDP on call terminating (outbound) leg
SIP could not process SDP received in setup ack, if setupAck was received by the call terminating
leg.

1664 DX did not recognize E&M Wink Idle state when connected device resets
When certain connected devices reset, they sent messages which the Tenor saw as an incoming
seizure. When a message was returned that the Tenor did not recognize, this resulted in Tenor
falsely identifying all channels as busy.

1666 Tenor using Routing server

When Tenor was using Routing Server it may have crashed when an autoswitch call was
attempted.

1674 Filtering out * and # in H323 Call

Digits E or F were being filtered out in H323 call by invalid digits. These should have been be
processed and converted into * or #.

1675 Tenor DX to Tenor DX MoIP calls failed (Tenor DX only)

On certain occasions, Tenor DX to Tenor DX MoIP calls worked very infrequently and baud rate
was low (20-some Kbits/sec). After a few minutes, the call dropped.

1682 SIP GUID should be printed as unsigned integer

SIP GUID should have been printed as unsigned int rather than int. Since currently it was printed
as an int, it showed up as a negative number in SIP messages. Customers billing system failed to
match GUID because of this issue.

1684 RFC2833 digit relay info was not in SDP for re-invites
When the Tenor set up the SDP for a response to a re-invite, it was not checking to see if it was
configured to do 2833, then the digit relay (telephone events) was put into it’s re-invite response.

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1686 Quintum did not handle Mid-Call ReInvite when ReInvite had no SDP
On receipt of a ReInvite w/o SDP, the Tenor was not providing a 200 OK.

1698 SRC call signaling IP was not in H323 setup message

When the Routing Server was used for routing, the Q931 setup message was sent out without SRC
call signaling IP. Certain softswitches needed this IP address for authentication purposes.

1700 Gatekeeper reset under certain condition

The gatekeeper reset when the RRQ rcvd was missing the RAS and CallSig Addresses.

1707 Ethernet Disconnected alarm would not clear

When a Tenor was booted up from power without Ethernet Connection, the “Ethernet
Disconnected” alarm would never clear.

1738 RTCP packets triggered NAT/PPPOE port change by mistake

When watching for address/port changes from remote gateway due to NAT or dynamic IP
assignment (Korean), incoming RTCP packets were mistaken as RTP with changed port number.
Subsequently, outgoing RTP packets were directed to the wrong port (RTCP port).

1739 SIP Cancel Processing, problems fixed

Improvements to SIP cancel processing have been made.

1740 Tenor was not accepting SIP calls without “from” tag
Tenors were not accepting SIP calls without the “from” tag.

1741 For SIP, the interval was too brief in REGISTER

For SIP, the REGISTER should have been be retried based on the value provided in Min-Expires
header present in the 423 response. This has been fixed.

1762 Alarm message duplicated (Tenor DX only)

The following alarm message was being duplicated on the Tenor DX: “Card Type Configuration
Mismatch”.

1770 Tenors did not include tag in the “To” field of SIP failure message
Tenor was not sending the tag in the “To” field when Quintum sent failure messages.

1784 Overlap dial needed to ignored when certain protocols were used
When overlapdial was enabled with the Japanese ISDN protocol, it would cause calls to failure.
The following protocols will now ignore overlap dial: PRI NTT (INSNET1500), BRI NTT
(INS64), 5ESS, 4ESS, DMS.

1789 Data pass-thru calls were not passing through the Tenor
ISDN data calls were not passing through the Tenor.

1790 Modem bypass feature did not work under certain situation
Modem bypass feature did not work when used with external routing.

1816 Quintum returning incorrect SIP Port in 200-OK Contact hdr

For 200-OK and Accepted messages, The Tenor sent IP and Incorrect SIP listening port.

1819 SIP call was filtering out certain character

As "+" is a valid digit but it was being filtered out for SIP calls.

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1824 Problems with Autoswitch CDR problem have been corrected
The following problems with Autoswitch CDR have been fixed: autoswitch flag was not set
properly, incoming/outgoing field for Autoswitch call did not have passthru circuit, and fields were
not documented properly.

1835 Setup without channel id

In the BRI NTT switch, the setup message from user side did not have channel id. The Tenor
rejected the call because of this.

1848 No voice for certain NTT BRI environment

In NTT BRI environment, a problem caused the inband information indicator to be lost, which
resulted in no voice.

1849 STATUS message dropped the call

If the STATUS message was the first message to respond to the SETUP message, the call would be
dropped.

1850 When 1700 was dialed, it was routed to IVR

When a customer dialed a 1700 number, it was routed to internal IVR. Internally, the Tenor used
"1700*" as a DNIS for special IVR and the internal routing table always had to route this number.
But when 1700 was an actual dialed number, it should not have matched this internal route.

1854 Rejected incoming Invites that were not part of any dialog
If the Tenor received an INVITE that has a To tag and was not part of any existing dialog, it was
treated as a new call.

1874 Start Accounting packets did not contain correct call type
H323-call-type attribute in start accounting packet was not reported correctly.

1878 Applied short term inter-digit-timer (4-sec) in R2 DTMF

For a DX with E1 interface to a PABX: when signaling was R2 and the incoming
tone was DTMF, if the origination side did not send an end of digit, it took 20 seconds to go to the
next state. In case of DTMF, if overlap was set to 1 or 0, it did not take effect at all. CAS generated
a # if the DNIS counter was set, but in the DTMF case, it did not.

1887 R2 call failed when Q931 progress received while receiving user category
In case of R2, for incoming calls, while the Tenor was collecting user category (AnswerDigit = 3),
if it received Q931 progress message on the other leg of the call, especially with ProgressIndicator
= 8 (inband info available), the call would have failed.

1890 SIP Register response not processed correctly.

The SIP Register response failed, which resulted in failed registration.

1897 For SIP, # was not allowed in username

The # was not a valid user portion of the userinfo field. It is now a valid character.

1908 Setup message sometimes did not include called party number
In NTT network, the setup message coming from the switch sometimes did not have the called
party number, but has the Facility IE, which we needed to pass to the PBX side.

1917 Missing digits when RFC2833 received from Lucent TNT

It has been reported that DTMF digits were missing when calls were terminating from Lucent TNT
using SIP/RFC2833.

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1933 For PPPoe, from some ISP servers, PADT was ignored until a timer expired
For PPPoE, Tenor now sends PADT & PADI continuously until the server answers.

1937 IP Translation was not occurring

One way voice was being experienced when the Tenor was not translating the Private IP to the
configured Public.

1944 Tenor sent Alerting after Connect

Calls would randomly fail because an alerting message was sent by the Tenor after already sending
a Connect.

1960 # falsely detected

Occasionally, the # is long enough that it gets falsely detected as the # being the first digit of the
dialed number. This applied to ## multisession tone detection.

1962 RADIUS CDR incorrectly report voice quality

Bad IP quality event was not being reported in the CDR to populate the h323-voice-quality field in
the Radius stop accounting packet. The count of bad quality of events is now sent.

2006 Radius Start-Accounting reporting wrong Called-Station-Id

Radius-Start Accounting message contained an access number in both the incoming and outgoing
legs, called-station-ID. The outgoing leg should have used the dialed number (destination number)
instead.

2009 Duplicate evlog messages in syslog server

After turning on evlog for one or more modules (excp was always on by default) the user had to use
“cmd evssyslog 1|0” to turn off evlog to the syslog server configured on syslogserver-1.

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4.0 Open Inconsistencies
The following section describes open inconsistencies which apply to all Tenor products; if an
inconsistency is applicable to a specific product, it is noted.

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.

866 Cannot register to Proxy with wildcard DNs in the Registration User field
Enabling registration to SIP proxy and using wildcard DNs, will cause the proxy to register with a
garbled address. As a work around, do not use wildcards in the DN.

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.)

1242 Private number dialing not working in SIP

When two Tenors are configured with point-to-point SIP, intercom is enabled on both sides, and
the Public and Private LDN are set on both sides, the Public number works but the Private number
gets a fast busy.

1528 Registrar IP is kept even when SIPSG is deleted

When the unit has two SIP signaling groups configured (i.e., SIPSG-1 and SIPSG-2), when you
delete SIPSG-2, the Tenor continues to send registration messages to the Register IP that was in
the deleted SIPSG-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.

2067 PrimarySIPServer set to 5060 causes continuous reboot

In SIPSignalingGroup, setting the PrimarySIPServer attribute to 5060 causes a continuous reboot.
This reboot only occurs if there is a UserAgent listening on port 5060.

2074 For SIP calls, unexpected Dial Plan issues may occur
For SIP, when initiating transfers, you may run into unexpected dial plan issues. As a work
around, you should add a “1” when configured for 10 digit local dialing.

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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.

2240 Some parameters are always used from the first SIPSG
If a gateway has multiple SIPSignalingGroups configured and a SIP call comes in through a
Signaling Group other than the first one, some parameters are still used from the first Signaling
Group.

2245 For SIP, empty Supported Header causes 1-way voice

On receipt of SIP message w/empty Supported header we don't see subsequent SDP information.
As a result, the call connects but with only 1-way voice.

2247 UserAgent parameters does 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 ' '

2277 If codec negotiated for first and second call is different, the result is no ring back tone
In Attended transfer, if the codec negotiated for the second call is different then the first call, it
causes no ring back tone for second call.

2315 Ethernet interface is disabled if DHCP discover fails

If the Tenor attempts to renew it's lease, and the DHCP server does not respond, it gives up after
about 2 minutes. After it gives up, the ethernet interface is disabled, and although the ei show may
display a valid IP, and siprd show may show a valid default gateway, when you try
to ping an IP on what appears to be the valid subnet, a system error appears.

2341Remote NAT does not work on SIP calls

The RemoteNAT feature does not work on SIP calls.

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5.0 Supplemental Information/Changes
1655 FaxModemCoding parameter
FaxModemCoding is now available under IPRG. When the Tenor is configured for modem over
IP calls, FaxModemCoding allows you to select which codec will be used.

1696 ID field now references trunk/routing groups

The Tenor now uses the "id" field in referencing trunk/routing groups for such things as the calls
command, CDRs, etc. Previously we would use the "name" field if it was populated. That was
determined to cause too much confusion.

1792 Radius Authorization for IVR Type 4 and 5 implemented

IVR Type 4 and 5 have been implemented in the Tenor.

1847 Cause code is changed to 41 for origination call fail

In case of a call fail on origination side (such as routing fail or bad format number) the cause code
has been changed from 34 to 41 for all the cases.

1919 When using DHCP, the IP address change causes reset in certain situation
When a unit is using the DHCP feature, the Tenor will reset in case of an IPaddress change for a
new lease time.

2054 Re-structured Site and DialPlan options

Removed fields from Site and moved to DialPlan. Re-structured DialPlan to add new fields.

AutoSwitchNumberType documentation incorrect for CLI help

For the AutoSwitchNumberType documentation, the default entry should be 0 (DID received).

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