Tenor (tm) VoIP MultiPath Switch/Gateway and Call Relay Products - P104-12-00 Release Notes

This document lists features and enhancements, as well as resolved and open inconsistencies, for the following products running software version P104-12-00:

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 CR 60

Note: Release P104-12-00 does not apply to Tenor CR-SP and Tenor CMS.

Interoperability

The Tenor DX, Tenor AX, Tenor AS, Tenor AF, and Tenor BX running P104-xx-xx interoperate with Tenor Configuration Manager, CM104-14-00.

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

See http://www.quintum.com/support for all required files and firmware update instructions for each product.

Open Inconsistencies in Release P104-12-00

2753 Tenor resets in certain scenarios using Routing Server and SIP

We have found the Tenor may reset in certain scenarios using the SIP protocol. This is in lab conditions under heavy load.

At this time, we advise you not to use SIP with the Tenor (Tenor DX, Tenor AS, Tenor AX and Tenor AF) running P104-12-00, and routing via the Routing Server. This issue is under investigation.

2777 Problems with telnet from recent versions of FreeBSD, Linux and MacOS

There have been rare problems reported with telnet from recent versions of telnet in FreeBSD, Linux (SUSE and Slackware) and some versions of MacOS. The symptom is that the Tenor will terminate the telnet session almost immediately (you will see a login prompt, but it will disconnect before you can login). This seems to be a problem primarily with consoles within X Windows and not with pure shell outside of a GUI.

This appears to be a problem with telnet echo negotiation, a bug in the affected telnet clients. A few customers have had good results upgrading to the latest version of KDE and Gnome terminal. This issue is under investigation.

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.

2741 IPDialPlan configuration for transferred calls is not followed

If the IPDialPlan is configured to delete the first 4 outgoing digits, and replace them with something else, the original dialed digits are kept without replacement. The transfer fails because number pattern is not matched by the transferred party.

2733 Cannot reconnect after accepting a Call

When a Tenor accepts a "call waiting" call while on hold from remote side, after finishing the Call Waiting call, it cannot connect back to the original call. As a work around, do not accept call waiting call while you are on hold or if you do accept it, you won't be able to connect back to original call.

2734 Hung calls for AttendedTransferKeystroke configured to non ‘HU’

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

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.)

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.

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 ' '

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.

2341 Remote NAT does not work on SIP calls

The RemoteNAT feature does not work on SIP calls.

2271 Too many digits in Pattern (UPDP) results in Tenor hanging

Using UPDP, when adding a pattern of 16 or larger, you will receive an error message, but when you do a submit, the Tenor will hang. This applies to CLI only (not Configuration Manager GUI).

New Features Introduced in P104-12-00

2323: Message Waiting Indication Feature

The Message Waiting Indication feature provides the ability to poll a Voicemail server for new voicemail messages. When new messages are available, a "stutter" will be applied to the dial tone on that port.

Subscription is an explicit registration by the Tenor with the SIP Voice mail Server in order to obtain message status during a specified period. Many Voicemail Servers feature an implied subscription, which means that you will be notified of messages waiting without needing to subscribe. Your system administrator will inform you if you need to subscribe to retrieve messages; it may not be necessary.

All fields are located in the global SIP signaling group (SIPSG), excepted where noted.

MWISubscribeDuration. The MWISubscribeDuration specifies how long the registration will last. The default is 60 minutes, during which time you will be sent a notify message if there is a change in voicemail status. After half of this specified period has passed, the system will attempt to resubscribe. Valid entry: 0-65535 minutes, default 60 minutes.

MWIFailureRetryDelay.This setting indicates how long the Tenor will wait after a failed attempt to subscribe to the MWIServer before trying to subscribe again. The default is 30 minutes. Valid entry: 0-65535 minutes, default 30 minutes.

MWIServer. The MWIServer is the IP address or host name of the SIP Voice mail Server. This will be provided by your system administrator; it may not be the same as your SIP Proxy Server. When this address is entered, a Message Waiting Indication, or "stutter tone," may be provided on the line to signal that there is a voice mail message waiting.

MWIServerPort. The port used for connection to the SIP Voice mail Server. Valid entry: port number, default 5060.

MWIUserName (available in the individual User Agent section). This is the phone number/extension or name that identifies the voicemail account. Valid entry: 0-15 alphanumeric characters.

MWIPassword (available in the individual User Agent section). This is the optional password that may be assigned by your system administrator to gain access to the voicemail account. It is not required in many cases. Valid entry: 0-15 alphanumeric characters.

1379 Inbound DNIS Translation

A new DNS Translation table enables you to attach a previously defined Inbound DNIS Translation Directory to a Trunk Circuit Routing Group, LCRG and/or IPRG.

On calls inbound on a LCRG/TCRG/IPRG, the incoming called party number can be translated using the inbound DNIS translation table before internal call routing is performed.

For specific configuration information about this table, see the Command Reference Guide.

2363: Allow Static DNS entries

A new DNSHosts prompt allows you to maintain a list of static DNS entries that will take precedence over other Tenor settings, such as a primary/secondary DNS Server. This table is similar to a hosts file on a Windows PC, or the /etc/hosts file on UNIX. You may make up to 32 entries with HostName/IPAddress/Priority definitions in the DNSHosts table.

For example:

DNSHosts
 HostName                        IPAddress     Priority
 -----------------------------   ------------  --------
 mysipserver.quintum.com         1.2.3.4
 tocktick                        1.2.3.5        1
 ticktock                        1.2.3.6        2

HostName

The HostName can be a URL like www.yahoo.com, or any other legal host name (see RFC 952 for a definition of legal host names) that you wish to associate with an IP address.

IPAddress

This is the ipV4 IP address that you want to associate with a HostName. More than one IP address may be entered for a HostName. For example, in the case of a URL like www.yahoo.com, multiple IP addresses can be defined so that there is load sharing among several servers.

Priority

The Priority setting is optional, and may not apply in all cases. It is useful when you have assigned multiple IP addresses to a HostName, and you want these IP addresses to be attempted in a specific sequence, like a hunt group.

1069: Session Timer Configurables

The Session Timer setting allows you to turn on a timer to check if a SIP session is still active, or should be terminated. This feature need not be supported on both ends of a call. Even if one endpoint requires it, the feature will work. It is especially useful if you have been having trouble with hanging calls. This command allows for a periodic refresh of SIP sessions to determine whether the SIP session is still active.

SIPSG Global section (applies to all User Agents in a SIP Signaling Group) includes the following new parameters:

SessionTimer

If this field is enabled, it exposes the commands SessionTimerMinSE (which conveys the minimum allowed value for the session timer) and SessionTimerExpires (which conveys the lifetime of the session).

Possible values are:

0 - Disabled (default) - The SessionTimer feature is not negotiated in setting up the call.

1 - Supported - The SessionTimer value is advertised, and is applied if it is supported on either end. The call will still connect if the SessionTimer feature is not supported at both ends.

2 - Required - The SessionTimer value is negotiated, and if it is not supported on both ends of the call, the call will be re-attempted as if it is configured as Supported. The caller does the session refresh.

SessionTimerMinSE

This setting defines the minimum allowed value for the Session Timer. The SessionTimer must be enabled for this setting to be available, and the negotiated SessionExpires value cannot be less than this value.

The SessionTimerMinSE setting establishes the lower bound for the session refresh interval. The purpose of this header field is to prevent hostile proxies from setting arbitrarily short refresh intervals so that their neighbors are overloaded. Each proxy processing the request can raise this lower bound (increase the period between refreshes), but is not allowed to lower it.

Valid entry: 90 to 3600 seconds, default 600 seconds.

SessionTimerExpires

The SessionTimer must be enabled for this setting to be available. The SessionTimerExpires setting defines the lifetime of the SIP session. The value is advertised in the INVITE request. The value is negotiated between Caller, Proxies on the path, and the called party.

The SessionTimerExpires setting establishes the upper bound for the session refresh interval. Any proxy servicing this request can lower this value, but it is not allowed to decrease it below the value specified in the SessionTimerMinSE setting.

Valid entry: 90 to 3600 seconds, default 3600 seconds.

2280: Allow UserAgent field to be configurable

The UserAgentHeader field enables you to define a string in the User-Agent header field, available in SIPSG.

Valid entry: 0-31 characters, default "Quintum/1.0.0"

In most cases, the default value of "Quintum/1.0.0" is appropriate for the string in the SIP User-Agent header field. This string is configurable in case the remote end of a connection requires a specific string in this field. This string may be up to 31 characters in length, no spaces.

2304: SIP Proxy Fail-Over behavior configurable

Desired behavior when receiving error responses is different for various vendors and ITSPs. Some proxies/ITSPs require fail-over to a secondary Proxy on receipt of an error response while application servers such BroadSoft do not. Other proxies and ITSPs may require one or the other behavior.

A new field, ProxyFailoverBehavior, controls whether or not the Tenor will attempt a call through a secondary proxy when it receives an error response. This is not a case where the primary proxy is unavailable. Rather, the primary proxy has returned an error message such as 503 Service Unavailable.

Valid entry:

0 - Behave normally, fail-over on error response

1 - Don't fail-over on error response (BroadSoft), default

The default setting of Disabled allows the call to fail if an error message is received. If you enable ProxyFailoverBehavior, the Tenor will attempt the call through the secondary proxy.

828: SIP call clearing after alerting/progress

A new configurable timer, SIPNoConnectTimeout addresses the situation where an incoming or outgoing SIP call has reached the stage where provisional messages such as "Alerting" and "Call Proceeding" have been exchanged, but the call has not connected. If a Connect message is not received within the configured time, the call is cleared. This is similar to the H.323 T310 timer.

Note: This configurable should handle calls in alerting state on both calling and called gateway. When the timer expires, the call is removed if no connect is received at any endpoint. (The lack of a connect message may be due to remote gateway reboot, or network failure).

Valid entry: 32 to 3600 seconds, default 180 seconds.

1081: SIP, Support PRACK methods

PRACKMethod

The PRACK message provides acknowledgment for provisional responses, such as Alerting, Progress, and Proceeding. This allows provisional responses to be exchanged reliably. It also insures that messages are exchanged in the proper sequence, because each message must be acknowledged before the next is sent.

The settings are as follows:

0 - Disabled (default) - Provisional response acknowledgment is not performed.

1 - Supported - PRACK is in passive mode. The Tenor will respond with PRACK if it is supported on the other end, but does not require it.

- When the Tenor initiates a call, it sends a Supported header. If the far end sends a reliable provisional acknowledgement, the Tenor responds properly.

- When the Tenor receives a call containing a Supported or Required header, it sends reliable provisional acknowledgements.

- When the Tenor receives an Invite containing no Supported or Required header, it will NOT send reliable provisional acknowledgements.

2 - Required - PRACK is in active mode. The other end must support PRACK, or the call will fail.

- When the Tenor initiates a call, the Invite will have a Required header. If the far end responds with an Unsupported header, the call will fail.

- When the Tenor receives an Invite with either the Supported or Required header, it will send reliable provisional acknowledgements.

- When the Tenor receives an Invite without either the Supported or Required header, then the call will fail with a "403 Forbidden" response.

2424: RADIUS; Generate Only One Stop-Accounting Per Leg

AccountingType

Determines the number of start, stop, and update accounting messages the Tenor delivers to the RADIUS Accounting Server per call.

Valid options are as follows:

0 - A value of "0" means that no messages will be sent (default). In other words, it means Start-Accounting, Stop-Accounting, and Update-Accounting messages are disabled. However, authentication messages will still be sent, depending on the IVRType configured in the Telephony or IP routing group.

1 - Tenor sends 1 stop accounting message per call.

2 - Tenor sends 1 stop accounting message per leg of a call. A complete call consists of an incoming leg and an outgoing leg. The outgoing leg may not complete the call on the first attempt. One outgoing leg Stop-Accounting will be generated for each attempt, regardless of failure or success.

3 - Tenor sends 1 start and 1 stop accounting message.

4 - Tenor sends 1 start and 1 stop accounting message for the answer (incoming) leg, and 1 start and 1 stop accounting message for every outbound (originate leg) attempt.

5 - Tenor sends 1 stop accounting message for the answer (incoming) leg, and only 1 stop accounting message for the originate (outgoing) leg. When this option is used, Start-Accounting and Update-Accounting messages are disabled and will not be generated. This setting is recommended if using RADIUS.

The table below shows how the setting of AccountingType changes the configurability of RADIUS start, stop, and update accounting messages (as well as the update interval).

  AcctType    Start      Stop       Update        UpdateInterval
  0           disabled   disabled   disabled      disabled
  1           disabled   1          configurable  configurable
  2           disabled   2+         configurable  configurable
  3           1          1          configurable  configurable
  4           2+         2+         configurable  configurable
  5           disabled   2          disabled      disabled

Example

If 3 outbound routes are attempted, and the third is successful, there will be the following number of messages:

Type 0: No messages

Type 1: 1 stop accounting message only

Type 2: 1 stop accounting message for the answer leg, and 3 stop accounting messages for the originate (outbound) leg

Type 3: 1 start and 1 stop accounting message only

Type 4: 1 start and 1 stop accounting message for the answer leg, and 3 start and 3 stop accounting messages for the originate (outbound) leg

Type 5: 2 stop accounting messages only

2517/2139: Support for SS7 ISUP-R

Note: The SS7 Signaling Group prompt is only available for Tenor CMS, code version P103-00-00 or later.

Protocol

Enables you to set SS7 to be used by the Tenor. The following settings are available under SS7SignalingGroup - Protocol:

1- ITU ISUP 1992 (default)

4- ISUP-R, Russian ISUP

ISDN User Part (ISUP) supports basic telephone call connect/disconnect between end offices. ISUP was established by the International Telecommunication Union (ITU) in 1992. Used primarily in North America, ISUP supports ISDN and intelligent networking functions. ISUP also links the cellular and PCS networks to the PSTN. The Russian version, ISUP-R, was established in 2000.

2434/1580: Add config item to select sending Nortel or Cisco format for SIP INFO msg

SIPInfoFormat

Available under SIPSG, this command allows you to select whether the Tenor sends a Nortel or Cisco formatted INFO message on outgoing calls.

Valid entry:

0 - Nortel (default)

1 - Cisco

1751: Add filenames/version numbers to show –v command display

When you execute the show –v command, the following filenames are displayed with the corresponding version numbers: CLI Help, CLI Error, and CLI Object.

1946: Add Session Audit functionality

For SIP, the Tenor now supports receiving a Session Audit message and responding to it.

2093: Added SIP Module version

When you execute the show –v command, "SIP module" and its corresponding version has been added to the display.

2171 DNS Redundancy enhancement

Previously, when a host failed, it is marked as failed and next host of the DNS entry would be selected. This condition remained until the DNS entry became stale and a new lookup was required. As an enhancement, once the Tenor gets a host failure, it forces a new DNS query. This occurs every five minutes and will continue to use an alternate host until the refresh.

2346 Same GUID for all sessions

Currently, there are two GUID types: Incoming GUID (a unique ID for all session calls, which means in the case of a multi session call, the GUID will always be the same for all of the session calls) and GUID (a unique ID for each session call). A new capability has been added for using Billing Vendor, type 3 (Porta Billing); the same GUID will be used for any session.

802 SIP Offer/Answer feature

The SIP Offer/Answer feature enables media negotiation between the Tenor and an endpoint where media negotiation can be initiated by the Calling Party or the Called Party.

2433 SIP Module in Tenor has been updated

The SIP module has been modified from "2.0.0" to "2.1.0".

2338 Support receiving and resending Nortel TelUri on Redirect

The Tenor now supports receiving and resending Nortel TelUri on a Redirect.

2007 Maximum number of characters for proxy password increased

The maximum number of characters for the proxy password has been increased from 11 to 32.

2392 Japanese Ring back tone support

Support for Japanese Ring Back tone added (400Hz, +/-20Hz).

2316 Multi-part mime support added

Nortel interoperability required support for multi-part mime headers. This is used to send SDP as well as other proprietary payloads. This support has been added to the Tenor.

2302 Broadsoft redundancy enhancement

Broadsoft redundancy provides a means to determine when a server has recovered and is available for use. It sends an Options message from the primary server application.

1907 New configuration option for RTP ports

A new setting, RTPVerify, addresses a problem with some routers and D-Link ADSL modems that change the RTP ports in the FastStart element of incoming Q931 messages. This new field is available in H323sg. Valid entries: 0 (Disabled) or 1 (Enabled).

2456 New evlog command to display EXCP messages

A new evlog command ev d excp, enables the Tenor to display only EXCP messages in the ev buffer.

2423 Generation of Remote Party ID headers

In general, the Remote-Party-ID header allows information to be passed back and forth from the PSTN side to a trusted SIP Server. There are two new commands to support the Remote-Party-ID: SendRemotePartyID and RPIDDefaultPrivacy.

SendRemotePartyID

This command controls whether or not the Tenor uses the Remote-Party-ID header when sending a SIP INVITE request or processing an incoming SIP INVITE. When this setting is enabled, it exposes the RPIDDefaultPrivacy setting.

0- disabled

1- enabled

RPIDDefaultPrivacy

0 - Disabled. Calling name and number pass through.

1 - Off. No calling name or number sent in the forwarded INVITE message.

2 - URI. The URI (Uniform Resource Identifier) part of the Remote-Party-ID header is set to the configured string in the forwarded INVITE message.

3 - Name. The calling name in the Remote-Party-ID header is set to the configured string in the forwarded INVITE message.

4 - Full. If an incoming call has both calling name and number blocked, the Remote-Party-ID header would include a "privacy=full" tag in the forwarded INVITE.

2120 Allow configuration of the fwd disconnect time

A new configuration field, ForwardDisconnectDelay, sets the time that qualifies a valid forward disconnect (battery removal) that the Tenor will accept. The SignalingType must be set to Loop Start, Forward Disconnect for this command to work.

ForwardDelayDisconnect

Valid entry: 200 - 2000 ms (500 ms, default user side; 800 ms, default network side).

2088 SIP Status command

A new command cmd sip status displays the status of the sip registrar(s) and proxy.

1988 IP Failures detected with RTCP packets

If a VoIP call was up between Tenor A and B, and IP hop 2 failed completely, autoswitch would not work (see below for diagram). The reason is that Tenor B would detect the IP failure (because its pings to Call Relay will not get any response), but it would not be able to inform Tenor A to switch the IP call to PSTN. Tenor A would not detect the failure because its pings will receive responses from the Call Relay.

Tenor A <--> IP Hop 1 <-->  Call Relay <-->  IP Hop 2 <-->  Tenor B

The Tenor now uses RTCP packets to detect an IP failure. The following entry of var_config is required to enable Autoswitch to use RTCP packets: RTCP-Autoswitch 1. If you are unfamiliar with the var_config file, see www.quintum.com and enter "var config" in the search box.

2590 Contact header added to provisional responses (180, 183)

For interoperability purposes, the Contact header has been added to provisional messages.

2551 Attended transfer disconnect timer now configurable

A new command AttendedXfrDiscExpiryTimer has been added to LCRG/TCRG to make the attended transfer disconnect timer configurable. Valid entry in seconds; default is 5. This field is available when you enable AttendedTRansfer.

2373 A new command has been implemented to specify the RTP port

The following entry in the var_config file is required to enable the RTP port: MinRTPPortNum (followed by a range). If you are unfamiliar with the var_config file, see www.quintum.com and enter "var config" in the search box.

2046 LoopBack status now displayed (Tenor DX/CMS/BX only)

A new line, Loopback Status, has been added to the display when you execute the status command under DigitalInterface-SLxDV1DIy. The Loopback status of the digital interface is displayed: None (down), Local (faces in), or Remote (faces out).

2591 Channelized passthrough fall back to regular passthrough

When passthrough enable is set to 3 (channelized passthrough), there is channel to channel mapping between incoming and outgoing trunks. When the requested channel is not availble, instead of not routing the call, the Tenor now will fallback to regular passthrough and look for any available channel in the trunk.

To use this new feature, add the new configuration option CPTFallback to the var_config file. Settings include:

0: No change (default) If the requested channel is not found, do not route the call

1: If the requested channel is not found, look for any available channel from the same trunk. If you are unfamiliar with the var_config file, see http://www.quintum.com and enter "var config" in the search box.

2560 Default SIP User Agent included

A default SIP User Agent with Listening Port 5060 is now created by the system under SIPSG-1.

914 Login/increase security

As a way to further increase security and track the number of all unsuccessful login attempts, the Tenor will now internally log exception/syslog/alarm/trap for any failed login on Telnet, FTP, or Confiuration Manager (GUI). The IP address and application will also be included in the exception.

2699 Send PI=8 on ISDN call originating side

A new feature enable PI=8 to be sent in the alerting message on the ISDN call originating side. Include this change through the var_config.cfg file. If you are unfamiliar with the var_config.cfg file, see http://www.quintum.com and enter "var config" in the search box.

SendPI 1

In the var_config.cfg file, the SendPI 1 command enables sending P1=8 in the alerting message on the ISDN call originating side (setting to default value of 0 will not change the behavior).

2298 Enhanced IVR feature in var_config.cfg

Through the var_config.cfg file, there is a new IVR option type 3 which enables you to do DNIS based authentication.

Note: When using this option, when you set the IVR access # to '*", multission will not work when the ivrtype is configured to 3.

RESOLVED INCONSISTENCES/CHANGES SINCE P103-08-00 GA RELEASE

1772: Certain IP failures did not generate an SNMP trap or alarm

IP failures such as OSCF, 303TO and 310TO, generated an exception, but did not generate a trap or alarm. These now will now generate the proper alarms.

2067: PrimarySIPServer set to 5060 caused continuous reboot

In SIPSignalingGroup, if you accidentally set the PrimarySIPServer field to a port number such as 5060, a continuous reboot may occur. This reboot only occurred if there was a UserAgent listening on the same port number. This has been resolved.

2086: Tenor was not able to come off hold after putting endpoint on hold

There was a problem with the Tenor's hold requests and responses; with certain endpoints, the Tenor was not able to come off hold after putting an endpoint on hold. This has been resolved.

2091: Allow Proxy Address was not set in Referred-By header

When a Refer was sent to Broadsoft, the referred-by header was set as the address of the Tenor, and not the address of the Proxy. To fix this problem, the SIPServerInFromHdr field (available under SIPSG, previously called InviteURIfromHost) enables you to select the address of the proxy (SIP Server IP) OR the address of the Tenor (Tenor IP).

2092 Re-invite did not work with some endpoints

Re-invite did not work with some endoints; when a Tenor received a re-Invite with no SDP, it sent only the currently negotiated codec. To fix this problem, if the Tenor now receives any Invite, with no media (SDP), it sends its entire list of media capabilities.

2176 Call Throttling EXCP's were not sent as traps or alarms

Call Throttling exceptions were not being sent as traps or registered as alarms. This has been resolved.

2209 Single stage dialing was not working correctly

The single stage dialing feature was not working corrently. Authentication was failing by setting the username field (account number) in pre-authentication to the DNIS. This has been resolved.

2240 Some parameters were always used from the first SIPSG

If a gateway had multiple SIPSignalingGroups configured and a SIP call came in through a Signaling Group other than the first one, some parameters were still used from the first Signaling. This has been resolved.

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

On receipt of a SIP message with an empty Supported header, the Tenor did not see subsequent SDP information. As a result, the call connected but with only 1-way voice. This has been resolved.

2282 Area Code was being added to LDN for CallerID

When PrefixCCAC was set to 0, and the CallerID type was set to 3 (which is default), the area code was being added to the front of the first LDN for caller ID. When the PrefixCCAC was set to 0, the first LDN should have been sent unmodified for caller ID. This has been resolved.

2325 Support for Tel URL formats

To meet Nortel interoperability standards, the Tenor now supports Tel URL parameters in the name section of the SIP url.

2335 IVR Disable prompt did not work when using Configuration Manager

When using the Configuration Manager, in TCRG only (not applied to LCRG), any changes to the IVR Disable prompt (available via Trunk Circuit Routing Group - IVR tab) did not work. This has been resolved.

2317 Tenor did not send correct DN in REFER message for Attended/Unattended Transfer

When the Tenor was the initial call termination UA and it initiated an Attended or Unattended Transfer, the REFER message that it sent contained a Referred-By header with the DN of the origination Tenor for the initial call, instead of its own DN. This has been resolved.

2364 Incorrect routing when "maddr" is in request URI

Incorrect routing occurred when the Tenor received an "maddr" parameter in the Request URI. This has been resolved.

2368 Contact header missing from REFER message caused call to drop

When using SIP with certain devices, when the Tenor sent a REFER message in a call transfer, it didn't have the mandatory contact header. As a result, the message "400 Bad Request" was returned from the device and the call was dropped. This has been resolved.

2395 When in Supplementary Services, INFO messages were not suspended

When the configuration for DigitRelaySIP (available through IPRG) was set to 4 or 5, and the Tenor was entering supplementary services, INFO messages were still being sent when they should have been suspended. This has been resolved.

2397 Idle BX Tenor generated noise to BRI S/T bus (Tenor BX only)

Noise was heard on a telephone connected to a BRI bus. This has been resolved.

2278 Problem Switching from Primary SIP Server to Secondary SIP Server IP

There was a problem in the Tenor switching from the Primary SIP Server IP Address to the Secondary SIP Server IP Address. This has been resolved.

2531 In rare cases, buffer overflow caused problems

Under rare circumstances and in certain configurations, it was found that a database file (db.txt) file may cause the buffer to overflow and cause problems. This has been resolved.

2513 Tenor not detecting the # character in Info message

The Tenor was not detecting the '#' character as a valid digit in the Nortel INFO message. This has been resolved.

2508 SIP Info messages from Nortel were being rejected

The Tenor was sending "501 Not Implemented" messages in response to Info messages received. This has been resolved.

2453 Content Length field not in ACK message

Using SIP, the Content Length header field (indicates the size of the message body sent to the recipient) was not in ACK message when there was no SDP present. As a result, this may have caused audio issues. This has been resolved.

2431 Tenor did not properly negotiate incoming offer, which caused one way voice

The Tenor did not properly negotiate an incoming Offer on a reInvite. The result was one way voice. This has been resolved.

2426 Default for RequestReTransmitCount field changed

In SIPSG, the default for RequestReTransmitCount has been changed from 6 to 11. This has been resolved.

2409 ACK to 407 Response going to Tenor instead of Proxy

When the Tenor sent an Invite to Proxy, it is challenged with a 407 Response, and the Tenor sent the ACK to itself rather than the Proxy. This has been resolved.

2367 Authentication w/Nortel SIP Registrar failed because MD5 was lower case

The Tenor failed authentication because it was setting the "MD5" string received on the challenge to lower case in its reply, when it shoudl have been in uppercase. This has been resolved.

2320 Broadsoft Authentication challenge problems

Registering with the Broadsoft server failed when authentication was enabled in the Broadsoft server. This has been resolved.

2303 Tenor only transmitting ACK for first 200 OK

If the caller was a Quintum Gateway, the Tenor was not transmitting ACK message for every retransmission of 200 OK message received. The Tenor was only transmitting ACK for the first 200 OK. This has been resolved.

2144 SS7 and DASS2 failed to detect red alarm (Tenor Digital only)

SS7 and DASS2 failed to detect red alarm message. This has been resolved.

2343 Occasionally, the Tenor could not authenticate registration

On intermittent network failures, the Tenor failed to Re-REGISTER with proper credentials. This problem has been resolved.

2479 Dass2 did not filter X from the Calling Party number

In Dass2, 'X' separates user and network part of the caller id. The Tenor did not filter X out of the Caller ID string. This has been resolved.

2361 SIP Listening Port unavailable

The SIP Listening Port was unavailable. In this release, you are able to set the SIP Listen Port to 5061.

2313 Changing any setting under MasterChassis caused TSI clock to reset (Tenor CMS only)

When a change was made to any setting under MasterChassis, it caused the TSI clock to reset. Some of the calls would fail. This has been resolved.

2279 Tenor did not allow multiple of the same transactions in a CallSession

If the Tenor received multiple INFO messages, the first was read and processed, but subsequent messages were not. This may have cause problems with some features. This has been resolved.

2090 Digits sent in format not supported by opposite end

The Tenor was sending digits in RFC2833 format, even if the terminating gateway did not support it. As a result, the terminating gateway could not hear the digits. This has been resolved.

2589 Inter-Tel ACK message failure

The Tenor sent a malformed ACK message to an Invite from the Inter-Tel. This has been resolved.

2284 Supplementary/CANCEL not sent after ringing timeout

A CANCEL message was not sent to the transferee phone after a ringing timeout. As a result, the transferee phone kept on ringing until the phone trying to make the transfer was hung up. This has been resolved.

2074 SIP: When Transferring a call, transferror was not following dial plan rules

For SIP, when using unattended transfer, the transferror was not following the rules of the dial plan (including 10 digit dialing). This has been resolved.

2038 Tenor had problems keeping call on hold during keep alive re-invites

The Tenor had problems keeping keeping a call on hold. If the Tenor initiated a hold, and then either sent a Reinvite (for session timers) or received a Reinvite (for Session Audit), the Tenor did not properly keep the call on hold. This has been resolved.

2568 Incorrect message played out when IVRRetries field is set to more than 1

For IVR/RADIUS, when the IVRRetries field was set to more than 1, the message requesting you to enter a new card number was not played out. This has been resolved.

2561 Answer detection (for calls from NetMeeting) not working correctly (Tenor AS/AX only)

For a Tenor AS/AX interoperating with NetMeeting, the Answer Supervision feature was not working. This has been resolved.

2570 Not routing calls to the Tenor BX when layer 1 was down (Tenor BX only)

The Tenor did not route calls to the Tenor BX, if the port's layer 1 was not up. This has been resolved.

2555 CallerIDType did not work when DNChannelMap was used

CallerIDType did not work when DNChannelMap was used. This has been resolved.

2608 Fax decode problem

When testing with Inter-tel, a problem was encountered where the Tenor failed to decode the "T38" format string in the SDP. The Tenor expected "T38". This has been resolved.

2595 Changes to Public/Private status of DN's in DNChannelMap were not reflected in the GK Table

There was a problem with the DN channel map number registration with the GK. The problem occurred when the type of number was changed from public to private, or vice versa, after the number was entered in the database. This caused calls to fail. This has been resolved.

2491 Codec G723 not properly coded

When the Tenor was configured for G723 5.3, the output was 6.3. Some gateways may have problems with this. This has been resolved.

2660 PPPoE link had trouble coming up

PPPoE had trouble connecting under certain circumstances. This has been resolved.

2659 Busyout feature changed for channelized pass through

The Busyout feature is activated by how many channel are left in a trunk group. The channelized pass through feature maps an FXS port into a FXO port without splitting the trunk group. To use this feature, configure busyout 2, and it will work like a single channel trunk group busyout.

2658 OutboundCallDetection feature did not work (Tenor Digital only)

The OutboundCallDetection feature did not work. This has been resolved.

2650 Call Waiting reset occurred

A reset happened under certain Call Waiting conditions. This has been resolved.

2536 Tenor could not find itself as GK after getting new IP via DHCP

After getting a new IP from the DHCP server, the Tenor was not able to register with a Gatekeeper. This has been fixed.

2680 Flashhook causes one-way voice on outgoing calls to PSTN side

When using port mapping, flash hook caused one-way voice on outgoing calls to the PSTN side. This has been fixed.

2465 Tenor had memory corruption issue

The Tenor crashed because of a memory corruption issue. This has been resolved.

2598 When IgnoreDNIS was set, it should not have been applicable to inbound IP calls

IgnoreDNIS feature was intended for inbound circuit calls and should not have been applicable to IP inbound calls. This has been resolved.

2673 Prefixes were not being stripped in private number patterns in UPDP

When defining a UPDP pattern as private (type 7) a prefix (defined in prefix length) was not being stripped (this did not affect the same entry as an international number). This would cause problems in certain call scenarios. This has been resolved.

2660 PPPoE link did not come up

When a Tenor with PPPoE enabled was booting up, if the ethernet link came up after a certain amount of time, PPPoE connection never started. This has been resolved.

2706 Interoperability issue

There was an issue with the SDP payload, where the telephone-events payload type was not a valid number.

2685 DefaultANI under "R2", did not override any existing ANI

When configured for R2, the DefaultANI did not overwrite any ANI. DefaultANI was sent only if the other leg did not send any ANI. This has been resolved.

2429 Channel group info missing upon upgrade

If a channel group name was defined in Release P102.xx.xx and the Tenor has been upgraded to P103-08-00, the information and details for the group would have been missing. This has been fixed.

2723 Allow asymmetric values in DTMF payload

The Tenor now reacts to a Re-Invite with a new DTMF payload midcall. Previously, we ignored a new payload.

2738 Continous PPPoE retry

If the PPPoE server or network was down, or there was a password error, the Tenor tries to connect to the PPPoE link. After a few tries, the Tenor would not retry again. This has been resolved.

CHANGES FROM P103-08-00 INTRODUCED IN P103 PATCH RELEASES

P103-08-10

2376 Codec Profile attachment incorrect after reboot

There were several cases where codec information in the Codec Profile was lost after a reboot. These situations have been addressed and the problem resolved.

2400 Voice Codec changed after reset

There were several cases where codec information in the Voice Codec definition was lost after a reboot. These situations have been addressed and the problem resolved.

2602 Slot 2 down in PPPoE mode

There were several cases where slot 2 (line cards) would go down when PPPoE was enabled. This has been resolved.

2626 ISDNSignalingGroup-1 and cassg-phone/line gets reattached to ChannelGroup after Reset.

There were several cases where if a Signaling Group was detached from a channel group, it would be re-attached on reset. This has been resolved.

P103-08-09

2574 Minimum Dial Digits reset to 7

If mindigits in the dialplan prompt was set to greater than 7, it would be reset to 7 on a reboot. This has been resolved.

2546 Need to use interdigit timer, as opposed to critical timer, on R2

In certain situations, an R2 switch may not properly send a termination character (usually #) at the end of the DNIS, and the system may wait 20 seconds to process the call. This now uses the configurable interdigit timer (default 4 seconds, but configurable down to 1 second) instead of the hardcoded 20 second critical timer. This only was a problem with R2, other protocols were fine. R2 now behaves like the other protocols in this regard.

2571 Local ring back ALWAYS provided independent of ProgressTone setting

On a phone-to-PSTN circuit-switched call via an analog Tenor, no matter how the ProgressTone parameter was set in the TCRG and LCRG, CH always attempted to provide local ring back tone. This tone was disabled very quickly (in about 100 msec). However, this caused a clicking noise that is annoying to the caller.

This has been resolved.

2580 Crash UPDP table in memory

UPDP table in memory could get corrupted when any dialplan DB item was changed. When the Tenor boots up, the table was good, but after db change, it may have gotten corrupted, forcing a reboot.

This has been resolved.

2582 Busyout feature did not work in case of new passthru type "Quintum"

When one FXS is in conversation with another FXS or IP call and there was another incoming call from the corresponding FXO port (via passthrough and "busy out = 2"), the calling party of the incoming call to the FXO would be a busy tone, but there was continuous ringing.

This has been resolved.

2479 DASS2 did not filter "X" from the Calling Party number

In DASS2 there can be an "X" character in the Calling Party field from the telco. This was not getting filtered and processed, causing invalid CLID to be relayed. This has now been resolved; the Calling Party is now properly filtered in DASS2.

2555 CallerIDType did not work when DNChannelMap was used

Calleridtype was not being used when DNChannelMap was being employed. This caused improper delivery of CallerID using certain calleridtypes. This has now been resolved.

2566 Var config for stopping SIP RE-INVITEs sent by Call Relay

Some SIP devices do not like having periodic RE-INVITEs being sent to them, and would disconnect the call if they were received.

As a temporary solution, a var_config variable has been added to allow this to be configurable:

EnableCRReinvite 1 enables periodic reinvites, and is the default
EnableCRReinvite 0 disables periodic reinvites and must be explicitly set for these cases.

A permanent solution has been included in P104-11-00. See P104GAReleasenotes#1069:_Session_Timer_Configurables

P103-08-08

2505 Manufacturing Test enhancements for small Tenors (MFG and DMT)

Several enhancements for pre-shipment manufacturing tests.

2549 Support for new motherboard (ID 0110) (Tenor AX only)

Adds support for a new AX motherboard.

2538 FXS to FXS call routing support for DN channel map

DN channel map typically uses a single LCRG. CH did not allow calls to be routed on the same LCRG, therefore FXS to FXS calls did not work with DN channel map. This bug fix addresses this issue.

2343 Tenor occasionally can't maintain SIP registration with certain proxies

There were situations, especially noted with Vonage and Asterisk where the Tenor loses registration state. It could get into this state when the Proxy failed several authentications in a row. This seems to occur only in specific circumstances, usually with heavily loaded (or underpowered) proxies.

The process has been made more robust and tolerant of these failures, so this is resolved.

2527 Tenor continuously re-attempted SIP Registration (every ~100ms) upon receipt of Authentication Failure (401 - Unauthorized)

Tenor could get into a loop after receiving a 401 unauthorized to a REGISTER attempt. It would cause continuous, and frequent, re-attempts. Code has been fixed to eliminate this situation.

2535 Tenors Running P103-08-05 would drop calls after a few seconds with certain proxies

This was an unintended artifact of bug 2347 (introduced in P103-08-02). These calls were being mistaken for parallel forked calls and canceled. This has been fixed.

P103-08-07

2530 Call Waiting with DNChannel Map broken with new SIPSG scheme

Call Waiting did not always work properly when DNCHannelMap was in use. It would sometimes deliver the second call to the wrong channel. It now works properly.

2531 Error for some DB configuration

In some circumstances a manually created text db file could cause the system to constantly reboot. This has been resolved.

P103-08-06

2246 Gateway description is not updated to RMSS server until the Tenor is rebooted

When RMSS server protocol was UDP, the Gateway->Description was not updated to RNMS server until the box was rebooted. The RMSS agent now reads this variable for every database update, so a reboot is no longer required to update this value.

2424 NEW problem while fixing this bug: Cause code was converted from 16 to 41

As noted in earlier updates on this bug, there were still some cases where cause codes were getting converted to 41 in error on RADIUS stop accounting records. These have now been resolved.

Note: this addresses the known issues in AccountingType 5. AccountngType 5 should now be usable.

2427 Negative call duration in ivr call

Due to rounding, it was possible that the reported time in stop accounting could be slightly higher than the allowed credit-time. This would cause RADIUS servers to get a negative balance, causing problems on some servers.

The rounding algorithm was adjusted to avoid this. Also note, session time in Stop Accounting messages is now rounded to the nearest second, not always rounded down as it was in the past.

2507 Support for FTP via RMSS

The RMSS now supports ftp over a tunneled management session. This allows for firmware updates, as well as log collection. The method for doing this is documented in the RMSS documentation.

P103-08-05

2398 Remote call id was not parsed properly

As a result, same remote call ID is reported for all the calls. This has been resolved.

2424 Cause Code converted from 16 to 41 in error

Bug 2424 (in P103-08-01) has issues affecting all accounting types in RADIUS. Cause code was converted from 16 to 41 when a call is disconnected by the termination side first. This leads to billing problems. This has been corrected and should no longer occur.

2501 CH crashes when submit is done if calls are in connecting stage

System may crash when a submit is executed and calls are in connecting stage. This should no longer occur.

2502 Tenors have been seen to hang during bootup--tNetTask dies

Occasionally a Tenor will hang on boot up. Seems to happen most often immediately after a software update, but it is not restricted to that case. This has been resolved.

2503 RTP Packets from Asterisk not being processed. Bug 2428 fix introduced this problem

Bug 2428 fixed problems with certain fragmented UDP packets not being recognized by the Tenor. This fix broke RTP with a Linux-based Asterisk IP PBX, causing one-way voice. This has been resolved.

P103-08-04

2452 SIP/PSTN Interworking change is affecting passthru calls

A certain combination of events could prevent ring back from being heard on a pass through call. This has been resolved.

2463 Duplicate pattern entries in several tables not allowed

In P103-08-00 through P103-08-02 a restriction was put which prevented duplicate entries in multiple Hopoff number directories, DNIS translation directories, Bypass number directories, and caller id translation directories.

This restriction would prevent duplicates to be entered, and would delete them upon migration from earlier versions.

This restriction has been removed.

P103-08-03

2319 SIP Crash during Attended Transfer

It was possible for the Tenor to crash during an attended transfer, when call waiting was enabled and a call came in. Similar to 2454 below, same fix applied. This is now resolved.

2356 CANCEL was ignored by SIP server

Note: This fix was in P103-08-01.

2428 Quintum unable to process some fragmented UDP packets

Certain large UDP packets, that were fragmented, were not being recognized and processed. Not all fragmented packets were affected, but packets fragmented by certain IP devices were having problems.

This has been resolved.

2432 HDLC driver stopped sending after power up (CMS ONLY)

Under certain conditions a CMS board may not have sent any packets. This would have been obvious by doing "status" at the appropriate DI prompt. You would see layer 1 up, and layer 2/3 waiting for establishment. RX counters would be increasing, TX counters would not. This could often be resolved by reboot.

This has been resolved.

2435 Need to support some punctuation in password fields

There was a restriction not allowing certain punctuation in PPPOEPassword field. This caused some problems where these characters are used. " % & < and > are now allowed in the PPPOEPassword field.

2449 SIP: Call through Secondary proxy fails

If Tenor failed to route a call through Primary Proxy and attempted to route the call through Secondary Proxy, the call always failed. Even upon receiving of 100, 183 and 180 through Secondary Proxy, retransmit timer expired and we retransmitted INVITE.

This has been resolved.

2454 SIP CallWaiting: Incoming call while off hook after release causes Crash

A crash could occur if a user is off-hook, but the call has been terminated (far end disconnected) and a call-waiting signal is sent.

This only occurred in this fairly rare situation, and only if call waiting was enabled.

This has been resolved.

2457 No audio when put-thru call is made

If a put-through call was made, the call had no audio. It has been fixed, these calls should now work.

See P103-08-00 GA Release Notes for more information on the put-through feature.

2466 Crashes and hangs with CDRServer enabled

Crashes and system hangs were reported with Tenors using the CDRs (not RADIUS). This has been resolved.

2467 Facility msg w/startH245 being sent w/o H245Address

In certain situations, H.323 Facility messages with starth245 were sent without an h245 address. Calls to some devices, such as Nortel CSE1000 would fail. This problem has been corrected.

P103-08-02

2425 ARJ from the configured GK resulted in incorrect attributes in Stop-Accounting

For Accounting Type 2 and 4, where a Stop Accounting was sent for every route attempt, the Stop Accounting when an ARJ was received from the configured Gatekeeper had incorrect values.

Here are the affected attributes and their corrected values:

Tenor-NAS-Port(VSA 2) = IP address of the GK from which ARJ is received.
h323-call-type(VSA 27) = VOIP
Quintum-Trunkid-Out(VSA 231) = IP address of the GK from which ARJ is received.
h323-remote-address ( VSA 23) = IP address of the GK from which ARJ is received.

If the rejecting GK is the Tenor embedded GK, the address will show as "0.0.0.0"

2347 On outgoing Invite, accept response from 1st destination only, if multiple are received

Tenor does not currently support SIP parallel forking. If we received multiple routes from a proxy, we attempted to route, and failed, often resulting in a crash.

Tenor now only acts upon the first route received, ignoring other routes.

2436 DN Channel Map and One to One passthrough features do not work for DASS2

The DNChannelMap and related one-to-one pass through mechanisms did not support DASS2 trunks. It now does.

P103-08-01

2162 Route mfg test output to event log

MFG and DMT tests now puts any errors from those tests in the event log as an exception (excp).

2343 Tenor occasionally could not authenticate registration

There were problems with Tenor occasionally losing its registered status with the proxy. A reboot solved the problem.

This has now been resolved.

2356 CANCEL was ignored by SIP server

Some SIP proxies ignored our CANCEL message or would send a 481 response to a CANCEL request, causing hung calls for some period (they would eventually timeout). The CANCEL message was rejected/dropped because of the presence of "tag" in the To: header. To resolve this the CANCEL now is sent without any Tag parameter in To: header.

2388 Some Databases when manually edited can crash the Tenors

There were conditions where if a user manually edited the configuration db text file, the Tenor might crash. This has been resolved.

2396 Stopping local ringback when cut-through is desired

There were times where the Tenor generated local ring back, when it should have been passing through the voice path from PSTN. This has been resolved.

2399 System loses progresstonecountry config after reboot sometimes

There were situations where the setting for progresstonecountry would be lost on a reboot. This has been resolved.

2402 No dialplan (dialplancountry=255) does not work.

If you tried to disable the Dialplan by setting DialPlanCountry 255, the dial plan was still applied. This has been resolved.

2404 AX showed ??? as the line type in 'status ds1'

The command "st ds1" on an AX that has FXS ports and no FXO ports showed "???" for the FXS line type. This has been corrected.

2419 Tenor resetting on receipt of TCS w/no codecs

There were isolated cases where a tenor would receive a TCS with no Codec listed. This could crash the tenor. It no longer will.

2424 RADIUS; Generate Only One Stop-Accounting Per Leg

In P102 code, we changed RADIUS behavior to generate a Stop Accounting for every outbound route attempt for a call with accountingtype set to 2 or 4. This was well received by most, but caused some RADIUS servers to behave badly.

To resolve this, we have now added accountingtype 5. This will send a Stop Accounting for each leg of the call as before, but only 1 for the outbound leg regardless of the number of routes attempted on the outbound leg.