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Tenor(tm) VoIP MultiPath Switch/Gateway and Call Relay Products
P107-09-00 Release Notes

This document lists features and enhancements, as well as resolved and open inconsistencies, for VoIP MultiPath Switch/Gateway and Call Relay Products running software version P107-09-00. These Release Notes also incorporate all resolved issues and changes included in Maintenance Releases P106-12-01 through P106-12-14.

For Response Point only: All features/enhancements/inconsistencies noted for Response Point are applicable only to those units listed here: http://www.quintum.com/responsepoint/rp_products.html.

The following sections are included:

Products Affected

The following products run software version P107-09-00:

  • 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 DXi PCIe VoIP Multipath Switch/Gateway
  • Tenor CMS VoIP MultiPath Switch/Gateway
  • Tenor Call Relay 60
  • Tenor Call Relay SP
  • Tenor Gatekeeper+
  • Tenor Response Point

Interoperability

The Tenor DX, Tenor AX, Tenor AS, Tenor AF,and Tenor BX running software version P107-07-03 interoperate with Tenor Configuration Manager, CM107-03-00.

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

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

Major Features

This section includes major features introduced in P107-09-00, as well as any new major features introduced in Maintenance Releases P106-12-01 through P106-12-14.

You now have the option to acquire certain features via a license upgrade. Contact your sales representative and/or price list for more information.

3563 Local call hold and call transfer (for H.323 only)

A new call transfer feature enables a call to be transferred from one port to another port on the same Tenor. The Tenor takes the dialed transferred-to number, and if it is routed to the same Tenor, a call will be made out a local channel. Both unattended and attended transfers are supported.

To initiate an unattended transfer, #90 is the default key sequence. ‘90’ can be changed to a different sequence if desired by doing ‘utk xx’, where xx is another key sequence.

To initiate a call hold: at LCRG, #46 is the default key sequence for keeping a call on hold and then taking a call off hold. ‘46’ can be changed to a different sequence if desired by doing ‘hk xx’, where xx is another key sequence.

4365 One number Fax solution

Note: This feature is intended for Tenor DX only.

A new feature in the Tenor allows voice and fax on the same number. An incoming fax call will be automatically routed to a pre-configured server. This new feature requires the following four entries in the var_config.cfg file:

enableCNGdetection 1

oneNumberFAX <access_number_pattern> <fax_mailbox_number>

  • access_number_pattern is the partial or whole access number dialed to reach the Tenor (i.e., 17324609000 or 1732*)
  • fax_mailbox_number is the mailbox number on the Microsoft Exchange Server where the fax is stored

CNGToneInitTimer <initial_time>

  • initial_timer is the interval between receiving the call and starting to look for the cng tone. Valid values: 1 - 60 seconds, default 1 second.

CNGToneDetectTimer <cng_detection_duration>

  • cng_detection_duration is the period for which the Tenor listens for the fax cng tone. Valid values: 1 - 60 seconds, default is 8 seconds.

Other Enhancements

This section includes other enhancements introduced in P107-09-00, as well as any other enhancements introduced in Maintenance Releases P106-12-01 through P106-12-14.

2870 DB Folder is protected from deletion

The cfg/db/ folder is now more protected from accidental deletion. In order to delete the folder now, you must do the following: Rename the folder and then delete the renamed folder.

3633 Hook Flash Using SIP INFO

Flash Hook can now be transmitted between two Tenors using SIP. This new feature supports Flash hook being detected and transmitted via the SIP INFO message.

3701/4306 Firmware management recovery of Tenor

A new feature, Tenor Safe Mode, provides a safe way to load system firmware when the Tenor keeps resetting because of corrupted system code. If the boot code on the Tenor detects the system code is corrupted, the Tenor will go into Safe Mode. In this mode the Tenor goes through the DHCP process and there are two ways firmware can be uploaded: Automatic and Manual.

Automatic Download. If the DHCP server provides TFTP information, the Tenor will attempt to get the system firmware using its internal TFTP client.

Manual Download. If the DHCP response message does not contain TFTP information, the Tenor will launch its internal FTO server so a remote user can upload system firmware onto the system from an FTP client as follows:

ftp 192.168.1.100
prompt for user name/password
put tnrsys.bin

The Tenor will go into DHCP mode only if the last working configuration was set for DHCP. If the Tenor's last working configuration was static, the Tenor will use the static IP and manual FTP is required to download code.

When a system firmware download is finished, the new system code starts automatically.

3789 Inbound DNIS translation on IPRG

There is now a new option to associate an Inbound DNIS Directory to an IPRG. This option is available through the CLI, under IPRG.

3856 Fax Bypass Mode

A new feature enables fax calls to be bypassed if an incoming call matches a configured number. Two new var_config.cfg commands enable the fax bypass mode:

Cidbypass. Caller ID number. For example, to bypass caller id 17325551212, ensure cidbypass 17325551212 is added to the var_config.cfg file. Up to 200 numbers may be entered, each one on a separate line with cidbypass.

For example, to bypass 17325551212 and 18475551212, the following 2 lines must be added:
cidbypass 17325551212
cidbypass 18475551212

When an incoming call matches a Caller ID configured, the call never routes over the IP. The following CH event log will be printed to show that the call is bypassed: Caller ID bypass; skipping IP route.

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

3945 Support of SIP G.729 Codec with annexb=no

On outgoing SIP calls, when G.729 is used and VADEnable is off on IPRG, SDP will include annexb=no. In reply, the Tenor will accept G.729 with or without annexb=no. In replying to the inbound SIP G.729 calls with annexb=no, the Tenor will include annexb=no.

3997 IVRType configuration added to var_config.cfg

Configuration options for IVRtype have been made available through the var_config.cfg file for the IVR pin length and card length.

ivrtype 2. Variable card length (255) and no pin (0).

ivrtype 11. Variable card length (255) and no pin (0).

ivrtype 3. Variable card length (255) and variable pin length (255). If the pin is not set to 1, or the pin length is set to 0, the pin length is not required. If the pin is set to 1 and the pin length is set to non-zero, the pin is required.

ivrtype 6. Variable card length (255) and variable pin length (255).

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

3999/4109 Transfer on busy or no-answer to support interoperability with Exchange Server (UM)

A new feature, available for circuit calls only (FXO-FXS and FXS-FXS), supports "transfer on busy or no answer", which supports call forwarding to the proper Exchange Server (UM) mailbox for voice and fax calls intended for a local FXS port. Enable this feature as follows:

On the FXS port's LCRG/HLDND, add the UM's phone number (i.e., 3001).

Configure the following var_config.cfg items:

  • SipTransportDefault. Specifies TCP transport. Example: SipTransportDefault 1
  • UMAddress. UM's IP address. Example: UMAddress 192.168.1.118:5065
  • UMPilotNum. UM's pilot voice access number. Example, UMPilotNum 999
  • UMCFNA. Enables the call forward on no answer. Example UMCFNA 10 (10 specifies the timeout in seconds; 0 disables UMCFNA)
  • UMCFB. Enables the call forward on busy and 0 disables it. Example UMCFB 1 (enables call forward; 0 disables call forward)

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

4000 MWI support SIP to Q.SIG Direction

MWI support for SIP to Q.SIG direction has been added to the Tenor.

4020 Increased number of User Agents

This feature increases the number of User Agents used by the Tenor in order to accommodate incoming call requests. The maximum number of UAs per SIPSG has been increased from 24 to 30.

4036 SIP Supplementary Services are now supported in Call Relay

Support for SIP-to-SIP call relay calls to pass through supplementary services has been added to the Call Relay.

4049 New IP Quality mechanism

The Tenor now uses a new mechanism, IP Quality Measurement Solution (IPQM), which comprises of one or more IP Quality Measurement Servers (IPQMS) deployed in the VoIP network. The new feature facilitates the measurement of the IP quality along VoIP ‘paths’ comprising two or more VoIP devices, over which the media is transported. This new mechanism, IPQM, does quality testing by sending properly formatted ‘probe’ packets to Quintum devices, specifically IP Quality Measurement Deamons (IPQMd), and receiving and responding to UDP packets arriving on a configurable port.

You must configure the Tenor for this new mechanism to work. Configure the following in the var_config.cfg file:

IPQMd Port #. Port Number. Valid entry: integer between 1 and 65535. Ensure this port number is unused in the Tenor.
IPQMd Password. Password. Valid entry: up to 19 characters in length.

4057 Caller-ID Based Routing

A new feature, Caller-ID based routing, is now included in the Tenor for FXO lines. For this feature to be enabled, the span must be configured to FXO (2) through the DNCM (DNChannelMap) table.

4081 R1 Modified; ST Tone

The Tenor now supports MF (700, 1700 Hz), which is added to the list of supported R1Modified UCST tones. The UCST configuration parameters now include: 0 (Tone frequency is 1500, 1700Hz), 1 (Tone frequency is 900, 1700Hz), and 2 (Tone Frequency is 700, 1700 Hz).

4118/538 G.726 Codec Support Added (SIP only)

For SIP only, G.726 (ADPCM) support has been added at rates of 16, 24, 32 and 40 Kbps. The new values are available through CodecVoiceCoding.

4134 Two new CDR formats, 5 and 105 (Tenor CMS and Call Relay SP only)

Two new CDR format 5 and 105 have been added to support Call Relay SP and CMS only. These fields match contain the same data stream as Selection 99 and 199, respectively, but do not include the Access Number field.

4145 SIP Enhancements

A few SIP enhancements are included in the Tenor to support the NTT switch:

  • When a call is put on hold, RTCP transmission will continue so the switch (if using RTCP packets as keep alive) does not drop the call.
  • In response to an Invite, if the Tenor receives multiple 18x messages, the Tenor will now use the SDP associated with the first 18x message. (Previously, the Tenor used the SDP associated with the latest 18x message.)
  • The Expires header is added to the Invite message.
  • The call processing in Tenor has been improved for cases where a session renewal fails due to the 491 (Request Pending) message, or due to some other error replies.

4152 For SIP calls: nextnonce support

If a server issued a nonce for one time use, it sent a nextnonce message. After receiving this message, the Quintum still used the "old" nonce in sending future registration messages. The Tenor is now able to send the received nextnonce in the next register message (within the same SIP dialog).

4175 Caller ID check improved on outbound calls

For SS7, a new feature enables Caller ID check to be improved for all outbound calls.

4188 New RElay ANI configuration

The RElayANI field (available through ISDNSG, CASSG, and IPRG) has been changed to include new configuration entries to support Caller ID length checking. The options available for RElayANI are as follows (ISDNSG, CASSG, and IPRG-specific fields are noted):

0 No Relay ANI.

1 Relay ANI. ANI is not relayed on outbound calls through this routing group.

2 Relay ANI with TON, NPI.  ANI is relayed on outbound calls through this routing group (default). Not available in CASSG.

4 No Relay ANI with Restrict. If an outgoing call (through TCRG, IPRG, or LCRG) matches the pattern and a caller ID is not present, or if the caller id length is less than the minimum specified, the call is not sent out the outgoing trunk.

5 Relay ANI with Restrict. Sends call but with CallerID checking as in option 4.

6 Relay ANI with TON, NPI. Sends call but with CallerID checking as in option 4. (Not available in CASSG.)

99 Get ANI from file (Not available in IPRG.)

Refer to these Release Notes for configuration options; documentation will be added to the Command Reference at the next release of software.

4222 Calling party category from SIP to IAM message of SS7 signaling

The Tenor now relays the Calling Party Category from the SIP side to E1, and vice versa, in the SS7 IAM message. The calling party category appears in a field in the INVITE message.

4201 SIP Unregister when T1/E1 is down (applies only to Tenor with digital interface, T1/E1)

If the PSTN connection (Digital Interface 2) on the Tenor is down or out of service, the Tenor now responds with a SIP unregister to proxy; no further registrations are sent to the proxy. To enable/configure this feature, a new var_config parameter is available: sipunregT1E1down a1 a2 a3 a4 a5 a6 a7 a8, where a* is DI # beginning from 1.

For example, with 1 card, 2 DIs (DI1 is connected to PBX; D12 is connected to the PSTN), sipunregT1E1down 2 indicates the following:

  • If DI2 goes down with an alarm, SIP sends unregister to proxy
  • If DI1 gets an alarm, SIP registration is unaffected
  • In SIP unregistration state, PBX user can make an outgoing SIP call via proxy as long as the proxy accepts the call or SIP static routing call

4212 Calling Party Number in ISDN Redirect

The Calling Party Number is now displayed in the ISDN Redirect Number.

4217 Call Forwarding

For incoming SIP calls, Ring No Answer Calls can now be forwarded to an off hour answering service. It enables calls to be redirected out the FXO line to a specific PSTN number. This feature can be configured through the var_config.cfg file as follows:

diversion <dn> <type> <new dn> <Redirecting num> <timeout>

DN. Dialed Number of endpoint located on the Tenor.
Type. Valid entry: u (unconditional), b (busy), nr (no answer/no response), bnr (busy/no-answer)
New DN. New destination number (normally the number of the answering service company or Voice Mail System). This can also be 'um' for Unified Messaging
Redirecting num. This is the number that get placed into the Diversion

For example: diversion 20157 nr 12010605555 20157 10

4224 Manufacturing test improved

The manufacturing test for Tenor products has been improved, which includes a more efficient process for allocating DSP channels.

4238 New command enables immediate resync with provisioning server

A new command try resync (available under Auto Provisioning) is now available to enable the Tenor to resync with the provisioning server immediately after the command is issued. The Tenor retrieves the configuration file and resync activity is printed out in the console screen.

4239 Scheduled Resynchronization at a specified time

The Tenor now has the ability to resynchronize the configuration files regularly at a particular time of day. The ability to schedule configuration resynchronization outside business hours minimizes the interruption caused by this process. A new field, ResyncTimeOfDay (available under Auto Provisioning) enables the Tenor to resync the configuration files according to the time of day. Valid entry: hh:mm format.

4240 New configuration capabilities via DHCP

The Tenor now supports the following new capabilities:

  • Configure NTP servers and offset from UTC via DHCP. In its DHCP broadcast message, the Tenor will request an NTP server IP address and UTC offset. Once these values are received, the Tenor will connect with the NTP server to set its time (without requiring a reboot).
  • Auto configure ProfileRule via DHCP. In its DHCP broadcast message, the Tenor will request a Profile Rule. When the Tenor receives a Profile Rule from the DHCP server, the Tenor performs the Auto Provisioning feature (without requiring a reboot).

4244 Support flashhook using RFC2833

The Tenor now supports flashhook generation and detection using RFC2833. Supplementary services must be turned off in the LCRG (config-TrunkCircuitRoutingGroup) and TCRG (config-TrunkCircuitRoutingGroup) for the Tenor to pass flashhook via RFC2833 between the CAS and SIP sides.

Also, DigitRelaySIP (config-IPRoutingGroup) must be set to a value of 3 (RFC2833) and the Flashhook Signal parameter must be enabled (via config-CASSignalingGroup-line and config-CASSignalingGroup-phone).

4261 Response Point Support

The Tenor now supports Response Point.

4264 Improved capability on second call failure

Previously, when a user made a single call and hits hook flash, if the call was unanswered or if it failed, the Tenor automatically returned the initial call. Now, if the second call fails, the user hears a busy (if 486 rcvd) or fast busy (all others). The user can choose to hang up or hit hook flash to return to the original call. Also, if the second call does not answer within three minutes, the user will hear a fast busy. The user can then choose to hang up or hit hook flash to return the original call.

4281 RFC2833 supported in H.323 signaling

RFC2833 based digit relay is now supported for H.323 signaling.

4284 Call disconnect on far end causes fast busy or three beeps

When a call is up, if the far end disconnects, the user will hear a fast busy and then is able to hit flashhook to return to the original call. If the far end that is on hold disconnects, the user will hear a one time '3 beep' sequence to indicate that the other call is gone. The call will remain connected to the current endpoint.

4293 IVR MultiSession was not working

The IVRMultiSession feature was not working properly (entering ## did not work). This has been resolved.

4304 Alcatel Diversion Support

The Tenor now supports Diversion Information from Alcatel PBX over Q.SIG interface for UM integration.

4308 G.729 with 10, 30, 50 or 70 msec has improved voice quality

When the Tenor interoperated with certain gateways (with G.729 codec with a payload size of 10, 30, 50 or 70 msec) the voice quality was not optimum. This resulted from a problem with the DSPs having a padding byte at the end of each RTP packet at payload sizes of 10, 30, 50, or 70 msec. As a way to remove these padding bytes (to improve voice quality), a new config item, NoRTPPadding, enables the Tenor to look for a Padding bit set in the RTP header of (non-fax) media packets (available through var_config.cfg). If a Padding bit is found to be set, it will be cleared and the length of the RTP packet will be reduced to remove the padding bytes. The following entry is needed in the file to activate this feature: NoRTPPadding 1.

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

4323 Upon receiving 180 Ringing, Tenor plays local ring back

Upon receiving an 180 Ringing, the Tenor will now play local ring back if media has not been received over the RTP channel, even if the 180 Ringing contains SDP. A new var_config.cfg option, EarlyMediaRingTone, enables the Tenor to determine if it should play ring back.

For example:

EarlyMediaRingTone 0. Upon receiving the 180 Ringing, the Tenor responds with a no RTP received.

EarlyMediaRingTone 1. Upon receiving the 180 Ringing, the Tenor plays a local ringback.

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

4324 Response Point Device firmware upgrade support (Response Point only)

Support for device firmware upgrade has been made available through the Response Point Administrator.

4325 Response Point Changes to support static IP (Response Point only)

Changes have been made to the Tenor to support Static IP for the Response Point.

4327 Option to disable sending SDP in Invite added so that RTP can be deferred until Connect (SIP only)

A new configuration command SDPinInvite was added to SIPSG. For SIP calls, this command gives you the option to enable/disable the sending of Session Description Protocol (SDP) in initial Invite messages. When disabled, the Tenor does not send SDP in the initial invite. Valid entries: 0 (Disable sending of SDP in initial Invite) and 1 (Always send a SDP in initial Invite (default).

4328 New features to Bearer Capability added (BRI only)

For calls origination from an ISDN phone or Analog (POTS) phone, new enhancements to the Bearer Capability are now supported in the Tenor.

Voice calls originating from an ISDN phone: This type of call will contain Bearer Capability = speech, and Higher Level Compatibility = Telephony.

Voice calls originating from an analog (POTS) phone connected to a Terminal Adapter or to an ISDN PBX: This type of call will contain the Bearer Capability = 3.1 Khz audio and must contain the IE Progress Indicator (PI) = 3 (origination address is non ISDN).

To enable this feature, include the following line in the var_config.cfg file: PTBearerCap 1.

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

4335 H.323 slow start results in one way audio audio (Call Relay only)

When the originating side and the Call Relay are configured to use slow start (DisableFS=1 in IPRG and MediaAfterConnect=1 in H323SG), RTP was not established
in the direction from the Call Relay to the originating end. This caused one way audio. This has been resolved.

4337 Tenor unable to retrieve parked call (Response Point only)

When a call was made between two Response Point IP phones and the user tried to retrieve the call, the Response Point phone played a fast busy and displayed an error. This has been resolved.

4339 RelayANI 2 in external routing server

For external call routing, a new option 2 has been added for Relay ANI in the IPRG. This new option will relay the ANI and set the the TON/NPI in the IPRG.

4340 SIP user account and password protected

A new feature now enables the Tenor to encrypt the configuration file that contains the SIP user account and password information. Through a new attribute, ProvisionMask (available through the AutoProvisioning prompt), you can define the data in the configuration file is encrypted. Valid values are as follows:

Provision Mask 0. No Encryption Mode. Information in the SIP user name and account are transmitted in the configuration file.

ProvisionMask 1. Encryption mode set. Information in the SIP user name and account are encrypted.

4388 Response Point supports Tenor Digital

The Response Point Interface now supports adding/configuring Digital Gateways. As a result, the Tenor DX is supported as a digital gateway.

4392 Response Point supports Tenor AFM

The Response Point now supports hybrid devices (single devices which support multiple device types). As a result, the Tenor AFM is included, which supports both FXO and FXS ports.

4396 180 Ringing default after Alert Message with PI=8; new varconfig to support previous application

As a default, upon receipt of an alerting with PI=8, the Tenor now sends 180 Ringing. Previously, upon receipt on Alerting with PI=8, the Tenor sent a 183 Progress to the SIP side as opposed to 180 Ringing. To support this interoperability behavior, a new varconfig command 183ProgOnAlertWithPI8 is now available. When this configuration option is set to 1, 183 Progress is sent over SIP when Alerting with PI=8 is received from the ISDN side.

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

4419 Allow unconfiguration of all phones or ATAs (Response Point only)

Response Point now enables a user to unconfigure all phones for a Tenor AFG or all ATAs for a Tenor AFT.

4425 Support of Flash for calls on hold

The Tenor now supports taking a call off hold using Flash (rather than using the "#" and any numeric button or "#" and "*" to take call off hold).

An example of a three party call using flash is as follows:

  • A calls B. B answers the call. A and B talk.
  • B puts call on hold by pressing flash. Gets 2nd dialtone.
  • B dials C. Phone at C starts ringing.
  • B hangs up while C is ringing. Call between A and C proceeds.
  • C answers call. B and C talk. B hangs up. Call connects
    between A and C.
  • C answers call. B and C talk. C hangs up. B presses flash to go back to A.
  • C does not answer call. B presses flash to go back
    to A.
  • B presses press flash again to go back to A without dialing.

4434 For Auto Provisioning, ResyncRandomDelay is now a random value

For the AutoProvisioning feature, each time the Tenor resynced with the provisioning server, it delayed "x" number of fixed seconds before it sent a request to the server, where "x" is the value set in the ResyncRandomDelay field. Now, the delay is set to a random value between 0 and the value set in ResyncRandomDelay. For example, if ResyncRandomDelay is set to 30, the Tenor randomly delays between 0 and 30 seconds before it sends a request to the server.

4449 Response Point DID Support (Response Point only)

For Response Point, when a Tenor is installed as a Voice Service, the Response Point Wizard gives the option to define any DID numbers that you may want to associate with extensions. As a result, the Tenor now passes through the "dialed" number to the Response Point as opposed to simply sending a '0'.

Resolved Inconsistencies

This section includes inconsistencies resolved in P107-09-00, as well as inconsistencies resolved in Maintenance Releases P106-12-01 through P106-12-14.

2947 SIP INVITE failover/retransmit count not expired

The request retransmit count configurable was not being read. This has been resolved.

3873 Tenor responded when an unmatched media content is presented

When an unsupported audio codec was sent to the Tenor, unexpected behavior occurred. The phone rang and the call disconnected, and there was no audio flow. This has been resolved.

3907 Tenor could not process INVITE with multiple escape sequence in Contact header

When the Tenor received an INVITE with a Contact header containing a multiple escape sequence, the Tenor failed to process the INVITE and dropped the call. This has been resolved.

4015 Redirect Issue for the gateway address

If the Tenor's default gateway was set to one IP, and the network LAN redirected the IP traffic to another gateway address, the Tenor built a dynamic host table for this new destination. Although the initial calls were successful, eventually when there were too many entries in this host table and the Tenor was not accessible via telnet or FTP, it required a reboot. The network buffers are now increased to support 1800 entries; once the number of entries reach 1800, the entries are deleted.

4088 No action taken on 487 as response to outgoing call

When the Tenor sent an Invite and the destination responded with a "487 Request Terminated", the Tenor did not take action and timed out. This has been resolved and the Tenor now responds.

4116 Update Request not Ok'd Correctly

For SIP, when the Tenor received an Update, it was not Ok'd or handled properly. This has been resolved.

4120 Extension number ignored in Invite message

When an extension number was specified in the Invite message (i.e., 9000; ext 1234), the Tenor ignored the extension number 1234 and did not dial it after the connect. This has been resolved.

4174 ZoneName field updated in help

The help for the ZoneName field (available through GatekeeperParam) has been updated to include 31 allowable characters, rather than 32.

4192 CDR broken in case of call fail

For an incoming IP call (SIP or H323), under certain call failures, the Tenor would not properly report CDRs. This has been fixed.

4197 Missing CSeq header in Invite caused Tenor to reset (SIP only)

Missing CSeq header in Invite caused the receiving Tenor to reset. This has been resolved.

4199 SIP in "from uri" should not have been in uppercase

When making outbound calls from Exchange through OCS, the From URI header contained a URI with "SIP" (uppercase), which caused a decode failure. The Tenor is now able to decode the SIP (uppercase) parameter.

4204 SIP encode and decode failures in Replaces part of refer-to header

With a SIP call, the Refer-to field had REPLACES (uppercase), which caused a decode error. The Tenor is now able to decode the REPLACES (uppercase) parameter.

4205 AT&T Flexreach: decode failure of P-Asserted ID header

Invite messages for both P-Asserted)-Identity and P-Preferred-Identity were not being decoded correctly. This has been corrected.

4206 Circuit switch video calls did not work in Aus and SS7 in Comstar (CMS only)

For circuit switched 'Unrestricted Digital' PRI calls, the video call was not working and for 'static channel connection' in SS7 mode, the links were not coming up. Both of these scenarios were happening when the lines were configured as E1. The problem occurred because the back plane law variable setting was inconsistent with how the Tenor code was using it internally.

A new var_config configuration option ForceBackplaneCompanding0E1T forces the back plane to 0 (E1) or 1 (T1). Set this option to 0 (E1) as a fix for the E1 problems (forces the backplane to be set to E1).

Note: A reboot of the Tenor is required for this change to take affect.

4207 Re-Invite due to session timer was treated as a forked invite (SIP)

After a call was established, if the SIPUA sent a session timer re invite, the Tenor treated it as a forked invite and dropped the message. This resulted in UA taking down the call. This has been resolved.

4208 Negative call durations displayed for CMD Calls command

Sometimes, the Tenor displayed negative call durations when a cmd calls command was executed. This has been resolved.

4214 Payload negotiation sometimes failed

G.729 payload was not being set by the Tenor as expected. For example, an incoming payload of 30ms was being ignored sometimes and instead, the configured value of 20ms was being used. Although the rtp streams showed that both endpoints seemed to adjust to 20ms, the voice was not optimum. This has been resolved.

4216 Tenor reset on receipt of Re-Invite

When a call was up in the Tenor and the far end attempted to initiate a transfer, a Re-Invite was sent with a very large SDP. The SDP was converted to the Tenor's internal format and this resulted in the system crashing. The Tenor has been updated to increase its capability to handle a larger SDP size.

4218 Q.SIG Voicemail Interface did not decode proper fields

Q.SIG voicemail interface did not decode proper fields.

4220 Pass Through type 2 did not work / alias name match

When the TCRG is configured for passthrough type 2, and there is no Alias Name configured in the DNCM, the call is routed incorrectly.

4223 Redirect did not pass through diversion reason field

When the Tenor sent an Invite and received a Redirect with a reason in the diversion header, the subsequent Invite had the diversion header, but did not include the reason field. The reason field is now being included in the Invite.

4224 Certain 20-port DSP card manufacturing defects not caught by Tenor AX Manufacturing Test

Some 20-port DSP cards (used in Tenor AX and DX units) had manufacturing defects, but they were not caught by the Tenor AX Manufacturing Test. Although the manufacturing test ran these cards "fine", problems occurred when one of the bad DSP channels was used. Either no dial tone was heard, or the DTMF was not recognized (depending on the nature of the DSP card defect). This problem has been fixed.

4227 On receipt of unattended transfer to a bad number, Tenor disconnected

The Tenor now keeps the original call up in the event of a failed call on receipt of an unattended transfer.

4228 Status DS1 reported wrong number of active channels

When a status DS1 command is used, the Tenor Digital units reported the wrong number of active channels. This has been resolved.

4229 DXi upgrade from 1 port to 2 ports, Slot 2 "Name" was not updated

With a DXi card upgrade from 1 port to 2 ports, the name for Slot 2 was not updated to reflect the change. This has been resolved.

4231 E1 links would not come up when upgrading software

With a DS1 card with E1 and T1 on the same card, when upgrading the software to P106-12-00, the E1's physical layer went down. This resulted from a problem in the DS1 card, and has been fixed.

4242 DNS Resolution timed out before 1st attempt completed

In AutoProvisioning, DNS resolution occasionally timed out before the 1st attempt results were returned; the Tenor waited for a retry timeout to expire before attempting the download again. This has been resolved.

4246 Multiple contacts in redirect caused call failures

For SIP, if the Tenor received a 3XX response to INVITE with multiple contact headers, the call failed. This resulted in the Tenor sending a subsequent Re-INVITE filled with incorrect information. This has been resolved.

4247 For SIP, a negative port number was sent out in the INVITE

If the port number was configured 50600 (greater than 32767) than a negative port number was sent out in the INVITE. This has been resolved.

4252 For Refer messages, Tenor used incorrect source for From Header

For a Refer message, the Tenor used the configured proxy value in the From header, instead of taking the information that was used in the original Invite transaction. This has been resolved.

4259 Call 2 for attended transfer did not have ringback

For attended transfer, there was no ring back when the Tenor received the Progress message for Call 2. This has been resolved.

4261 Response Point Changes (Response Point only)

Many changes have been included to support Tenor's interoperability with the Response Point product, including license updates.

4262 P-Asserted-Identity Decode failure/Register failure

When an Invite was sent with a P-asserted Identity header from SIP to the Tenor, the decoding of the P-asserted identity header failed because of the presence of the <tel:550011> line. Also, Tenor failed to decode a SIP 401 message at the authentication header field containing the stale="true". These issues have been resolved.

4263 Auto Provisioning parameters did not display correctly

When an XML file was loaded into the Tenor with the Auto Provisioning parameters configured, some of the parameters were still showing as default when a show, show –xc, or show –l command was used. This issue has been fixed.

4268 Call park did not work when '#' was sent

When a call was up and connected to the park server, if the user pressed '#', the Tenor processed this incorrectly and put the call on hold (in preparation of receiving command digits, such as 48 to hang up the current call). When the park server received the Hold, it disconnected the call and the park failed. This has been resolved.

4271 Certain fields not applied to Q931 message in case of external routed call

In IP outbound call, ANIScreenInd and ANIPresentationInd field in Q.931 are set according to user configuration in IPRG. But when the call was routed by external routing server, the IPRG configuration was not applied to the Q.931 message. This has been resolved.

4274 Unprovisioned Ports are registering (Response Point only)

Unprovisioned ports on the Tenor were sending registrations to the Response point when they should not have been. This has been resolved.

4278 Reset during conference establishment

When a user had two calls up and hit hook flash to conference all three calls together, a reset occurred. This has been resolved.

4282 Calls dropped after Tenor sent connect too soon

If a call was answered automatically (without a ring first), the Tenor sent a connect too soon before the PSTN was ready and the call was dropped. This has been resolved.

4285 No Dialplan set TON/NPI to invalid value

In a call from a FXS port without dial plan configuration, the DNIS with EOD was translated to a wrong TON/NPI value (such as 165). This has been resolved.

4286 In Australian Dialplan, international number was translated into a wrong number

In Australian Dialplan, international number was translated into a wrong number. (For example, the Australian Dialplan international number 001182324395299 was incorrectly translated to 6282324395299). This has been resolved.

4288 DHCP: system kept resetting after getting new IP address

If the first DHCP offer came from a server too late (more than 30 seconds), the Tenor kept rebooting. Improvements in DHCP timing have been made to resolve this issue.

4293 IVRMultiSession was not working

The IVRMultiSession feature was not working properly (entering ## did not work). This has been resolved.

4296 Max-Forwards not included in PRACK message

The Mediation Server expected Max-Forwards in the Tenor's PRACK message, but they were not included. As a result, calls failed. This has been resolved.

4298 Response Point

On a Response Point Tenor, if a user made changes to parameters set automatically by the system, these changes were overwritten upon reset/reboot of the Tenor. These changes should have stayed, despite the reboot. This has been resolved.

4301 ISDN Span reset (PSTN side did not receive acknowledgment)

When T1 link was removed and then returned, the Tenor did not resync until it initiated a span reset. A new var_config.cfg command, ResetSpanAfterLinkUp, will reset the T1 link after layer 2 comes up. The value (in seconds) of this parameter controls the interval between two resets. Values 1 to 30 will set the interval to 30 seconds; values greater than 30 will set the interval to the specified value. For example, ResetSpanAfterLinkUp 30 resets the T1 link after 30 seconds.

4303 Response Point Base failed to discover the Tenor (Response Point only)

Due to a timing issue, the Tenor was not being discovered by the Response Point. This has been resolved.

4312 CMD Test Command Fails on Response Point units (Response Point only)

CMD test was failing on the Response Point Unit. This has been resolved.

4313 Response Point decode error in INVITE (Response Point only)

In an incoming INVITE message, the Response Point decoded the message incorrectly. This has been resolved.

4317 Decoding issue with Tenor's Escape Processing Functionality

There was a decoding problem with the Tenor's escape processing functionality. A Refer message failed to decode the Replaces part of the Refer To header. This has been resolved.

4326 For Response Point, click to call failed (Response Point only)

A problem appeared in the Invite that the Tenor initiated in response to the Refer. This has been resolved.

4329 Incoming call to an unprovisioned line was routed back out to the PSTN (Response Point only)

If a call came in to the Tenor on an unprovisioned line, the call was routed back on another line with the dialed number of '0'. The Tenor should not have been answering incoming calls on unprovisioned lines. This has been resolved.

4332 Codec in H323 Bearer Cap was always Mu-Law

When you have configured a Tenor to use G.711 alaw only, the bearer capability in H.323 SETUP message was still configured as G.711 ulaw. This has been resolved.

4335 H.323 slow start resulted in one way audio (Call Relay only)

When the originating side was configured to use slow start (DisableFS=1 in IPRG and MediaAfterConnect=1 in H323SG), RTP was not established in the direction from the Call Relay to the originating end. This caused one way audio. This has been resolved.

4336 Re-provision of Tenor caused error

When a phone associated with the Tenor (provisioned in Response Point) was unprovisioned and the Tenor was reboot, an error message appeared if you attempted to provision the Tenor in the Response Point GUI. This has been resolved.

4337 Tenor unable to retrieve parked call (Response Point only)

When a call was made between two Response Point IP phones and the user attempted to retrieve the call, the Response Point phone played a fast busy and displayed an error. This has been resolved.

4338 Auto Provisioning: admin password error

When the admin password was included in the Auto Provisioning configuration file, an error would be displayed. This has been resolved.

4342 Reset due to external routing request

When a Tenor sent an external routing request message to the server, the system reset (in case of the radius event log on).When the Radius log was off, the system did not reset. This has been resolved.

4343 Help corrected for CLI command

Help has been corrected for the command help cliht (available under the sipsg prompt).

4348 Tenor reset under specific condition

The Tenor reset under the following condition: ivrtype was set to 9 in LCRG or TCRG and ToneBasedSupervision was set to 1 or 3 in cassg. This has been resolved.

4364 User Agent setting caused Tenor to crash

The Tenor would crash when all the columns of the SIP User Agent table were changed simultaneously. This has been resolved.

4371 Attended transfer to a busy user was not working correctly

With attended transfer, a user was able to transfer a call to a busy user. This has been resolved.

4372 Nortel H323 Invalid TON

In the Tenor, the UPDP for Nortel set the H323 TON to an invalid number (i.e., TON = 07). This has been resolved.

4385 CallerID Generation altered ring cadence (Analog only)

Caller ID generation type 1 increased the duration of the first ring off cycle sent to the Phone/PBX by approximately two seconds, which caused certain PBXs to drop the call.

4394 SIP INFO encode/decode problem

SIP INFO was not being encoded/decoded correctly. When you would press *, #, or A-D, the digits would not be translated correctly. For example, when a call was made to an Automated attendant and the user pressed the # key, the call would not go through. This has been resolved.

4398 Decoding problem with payload

For SIP messages, the Tenor did not decode the payload correctly. This has been resolved.

4400 Forward Disconnect did not work during Answer Supervision (Analog only)

When an outgoing call was in answer supervision state, the forward disconnect functionality was ignored. This has been resolved.

4401 False alarm of Card Type Configuration Mismatch

When the Tenor reboot, it issues an alarm (false) of Card Type Configuration Mismatch, which can be viewed by "alarm a" command. This has been resolved.

4422 Tenor sent out wrong number for TON

While sending H.323 calls, the Tenor sent out International as the TON, rather than the configured National. This has been resolved.

4402 Audio heard after receiving a hold

When a call was up between the Tenor and Microsoft Office Communicator, and the the call was put on hold, audio could still be heard. This has been resolved.

4433 Radius; Accounting and Authorization Data Mismatch

For Radius, there was a mismatch between the authorization request and the accounting request. IVR calls were not getting billed properly when certain upgrades took place. This has been resolved.

4444 Adding a user to a phone caused the Tenor to crash when the User Agent was changed (Response Point only)

For the Response Point Manager, when a phone was added as a voice service, and then a user was added to that phone, the Tenor crashed. This only happened when the User Agent was changed (i.e., deleted or disabled). This has been resolved.

4462 Large number of User Agents may have caused reset

If a large number of User Agents were configured and a database submit was done, the Tenor reset. For example, a crash was seen when about 70 UAs were configured on an Tenor AF and/or about 80 UAs were configured on a Tenor DX. This has been resolved.

4471 Enabling EarlyMediaTone may have caused PRI call failure

When EarlyMediaRingTone was set to 1 in varconfig, the PRI calls may have failed. This has been resolved.

4482 Local Loop type not written in database (Response Point only)

When a telephone line was connected to an FXO port, the LocalLoopType setting (available via config-CASSignalingGroup-line) was automatically detected but not written into the database. This has been resolved.

4489 Call Transfer problems resolved

The following problems were resolved with Call Transfer:

Attended Transfer. A called B, B put A on hold & dialed C, if C didn't answer within 1 minute, B & A should have been connected back. But C kept ringing even after 1 minute & B did not connect back to A. This has been resolved and the call is returned back to A successfully.

Unattended Transfer. A called B, B put A on hold & dialed C. If C didn't answer within three minues, the call was still kept busy. This has been fixed to drop the entire call if C doesn't answer within three minutes.

Other Changes

This sections includes other general changes (such as interoperability changes and configuration option changes) in P107-09-00, as well as changes that occurred in Maintenance Releases P106-12-01 through P106-12-14.

4137 Default behavior for PRACK changed

Support the PRACK. The default behavior for PRACK has been changed to enable the PRACK message only if one side (caller or called party) requires it and another side (caller or called party) supports it.

4138 Session Timers changed

Two configuration options, SessionTimerMinSE and SessionTimerExpires (available through SIPSG) have been changed to accept a value of 300 - 65535, default 600. Previously the range was 90 - 65535. This change improves the functionality of the Session Timer feature.

4165 References to AutoSwitch removed

All references to AutoSwitch functionality have been removed from the Configuration Manager and Command Line Interface.

4226 FaxNominalDelay default changed

The FaxNominalDelay (available through Fax Profile) default was 600, which caused many fax failures. This has been changes to 200.

4235 Modify AutoUpdate

AutoUpdate has been modified so that UpgradeRelease in the downloaded configuration file is not different than the UpgradeRelease in the running configuration (it will only be different than the firmware actually running on the device). Tenor will download and apply the new firmware from the provisioning server when the following conditions are met:

  • The retrieved UpgradeRule is configured
  • The retrieved UpgradeVersion is different from its current software
    version
  • The retrieved UpgradeEnable is configured

4236 Help made more consistent for FaxProfile parameters

The Help for FaxRelay and FaxModemCoding (both available under FaxProfile) have been modified to be more consistent.

4237 Configuration Resynchronization via SIP Notify

The process of configuration resynchronization (i.e. download and installation of a new configuration file) is initiated with a SIP Notify command.

4245 ISUP-R CalledPartyCategory Value Change

On incoming SS7 calls, the CalledPartyCategory is now set to 1 in the ACM message sent back from the Tenor (previously this field was hard-coded to 0). This value will have a default of 1 when ISUP-R is used.

In addition, a new varconfig value, SS7CalledPartyCategory is used to configure the CalledParty Category, if desired.

4255 RTP payload names are now configured in uppercase

Previously, the Tenor sent lower case payload names (g729) in the SDP. Some switches only recognize uppercase payload names (G729). As a result, the RTP payload names are now configured in Uppercase.

4258 Help for SessionTimeExpires updated

The Help for SessionTimerExpires (in Configuration Manager and Command Line Interface) has been updated to included a minimum value of 300.

4289 Time to detect DialTone has been increased

When executing the command debug test o or debug test t, the Tenor waited five seconds before reporting "Dial tone not detected" when no PSTN line was connected to the FXO port. Previously the wait time was two seconds.

4311 Response Point FXO (AFT) configuration changes

Support and configuration changes have been made for the Response Point FXO.

4344 Hunt feature clarification

Although there has been confusion about the interworking of the DNCM and the Hunt feature of the Tenor, the hunt feature of the LCRG and the TCRG has no
involvement with the DNCM. The system will not "hunt" though the entries of the DNCM if it finds a match. When a user creates more than one entry in the DNCM with the same DN (Dialed number), the system will only use the first DN. Even if the first DN is "busy" (engaged in a call), the system will not choose the second DN.

4383 Tenor supports up to 35 character in PDL

To support Microsoft Response Point, the Tenor now accepts up to 35 characters in the PDL. Previously, the Tenor accepted 31.

4389 FXS ports now uses the Response Point base unit as it's MWI server

If FXS ports are present (i.e., Tenor AFG and AFM), the address of the Response Point base unit is used as the Message Waiting Indicator (MWI) server and the outbound proxy server.

Open Inconsistencies

This section includes open inconsistencies in Release P107-09-00. Specific details are included for each open issue. If a work around is available, it is listed.

245 Windows XP file explorer does not interoperate well with the Tenor for FTP

When using Window 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.

1050 Specific database changes need a reset to take effect

When you change the CDR password or IP address, the Tenor requires a reset in order for the changes to take effect.

1134 Disconnect Supervision works only for option '2' (Tenor AX/AS only)

The Disconnect Supervision Options (# of on/off intervals per cadence cycle) works only when option "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 option 2. (Note: The default value changed to 2.)

1862 When receiving a malformed SIP message, Tenor does not return 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. As a work around, disable progress tones in the LCRG.

1987 ToneBasedSupervision not working on transferred call

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

2214 MaxForward may produce unexpected results

When using the MaxForward feature, unexpected results may happen. For example, when a Tenor is being called, it may not use its own Max Forward configuration for returning messages. Instead, it uses the Max Forward configured in the calling Tenor.

2247 UserAgent parameters do not accept blank value

To un-configure any of the UserAgent parameters, the change command with a blank value does not work. As a work around, put an empty string character in single quotes following the command, using the following format: change 1 PrimaryPassWord ''.

2341 Remote NAT does not work on SIP calls

The RemoteNAT feature does not work on SIP calls.

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.

3334 Tenor Migration between platforms not working as expected

After loading a Tenor S with a version of Tenor P (or vice versa), there was a problem with port usage. As a workaround, when migrating from a Tenor S to a Tenor P (or vice versa), do a setfactory command on the Tenor and then configure the unit manually.

3790 In IPRG "RejCallNoANI" does not work

The following scenarios are working opposite of what is expected:

  • When "RejCallNoANI" is enabled, and when when there is no entry in the DN channel map from the source Tenor, the call should not go through. The opposite happens in the Tenor, and the call does go through.
  • When there is an entry in the DN channel map and RelayANI is disabled, the Tenor will not send out any ANI to the destination. As a result, the call should fail because "RejCallNoANI" is enabled in the destination Tenor. The opposite happens and the call goes through.

4386 PTID not accepting correct parameter

The PTID (PassThrough ID) configuration (available through TCRG and LCRG) should be accepting values of -2147483648 to 2147483648. Currently, the configuration only accepts a range of -2147483648 to 2147483648. The configuration and documentation will be updated in the next release.