Release Notes for P104 Patch Releases

Tenor Gateways and Call Relays - AS/AF/AX/DX/BX/CMS/CR60/CRSP

A software patch is a release of software that has been created by Quintum to resolve an inconsistency or 'bug' that has been identified within a previous release of Generally Available (GA) software or patch. They may also add features. Patch software does not go through any testing from Quintum, but has been verified by the customer that the identified the bug is resolved with this patch.

Patches (for Second Generation Tenors) can be identified by the 3rd set of digits in the release name as PX-Y-Z. Where the Z is the patch number. All patches will be built on top of the latest GA release of software. For example, P102-11-00 is the latest GA software, then the first patch for this software will be P102-11-01.

If additional patches are required for a specific GA release, each new patch will contain all previous patch fixes. For example, the patch P100-11-02 will include all patches from P100-11-01.

On a scheduled basis, Quintum will release a full GA version that will include all previous patches and will have been put through our full system tests.

We hope that this process will aid our customers by providing fixes in a more rapid manner.

CMS COMPATIBILITY NOTE:

P101 and LATER CODE ON CMS ONLY SUPPORTS SERIES 2 CONTROLLER AND PERIPHERAL CARDS!

See this document for more details:

http://www.quintum.com/support/products/2G/cms/sysdoc/CMS_series2_notice.pdf

The notes below are cumulative since the last GA Release, P104-12-00. The most recent changes are in the top section, the history is similarly in most recent first order.

P104-12-20

3428 Incorrect handling of multi-session IVR key input

3449 Improper handling of Radius 'username' field for multi-session IVR calls

3625 Missing ringback on certain SIP calls originated from interface using DASS2 signaling

P104-12-19

3762 Modified E1 TX Pulse Template in Tenor DX models

3763 Added FXS/FXO daughter board IC revision to manufacturing test output

P104-12-18

2880 Added support for expanded Tenor AF product line

3623 Added support for '-NF' FXS daughter boards

P104-12-17

1788 Enhanced Ethernet port manufacturing test

3504 Failure To detect FXO loop drop for Tenor Analog models

3546 Manufacturing test initialization delay

3575 Ethernet manufacturing test improvements

P104-12-16

2837 Use DN for Authorization in Register

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

This is enabled by setting SIPUseDNinRegister 1 in var_config.cfg

3359 Incoming Fax Call Causes DSP Crash If FaxRelay Is Disabled

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

3435 CH: SIP Mica/moca can cause memory leak

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

P104-12-15

3283 Media Stream Loopback functionality not supported in Call Relay

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

3397 Gateway could crash if FaxRelay redundancy enabled

Under heavy load and certain codec configs, and FaxRelay redundancy configured, it was possible to crash the gateway. This has been resolved with various checks added.

3404 System upgrade may cause file corruption

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

3408 Incorrect User-Name Value in Start Accounting

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

3410 CR60 crashes while trying send RTCP and packet saver packets

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

P104-12-14

2578 Implement Uniform Transfer Feature

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

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

Currently, the default for Unattended Transfer is 90.

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

Here is how it will work:

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

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

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

If call 2 fails, we disconnect the original call.

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

2648 Toll Free Feature is Not Working

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

3028 Channels not being freed

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

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

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

3141 Support for SIP alias names

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

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

There are two new config items:

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

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

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

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

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

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

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

3361 Australian Progress Tone not working

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

3380 PTE= 2 on Tcrg also presents LDN routes

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

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

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

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

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

With H323 slow start it is possible to directly negotiate T.38 fax without ever going to the voice mode. In that case, when fax complete event is received, tenor should not try to switch to voice mode as no voice codec is negotiated. This is now the case.

P104-12-13

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

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

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

If a terminating endpoint does not support faststart (h323 calls) or does not send 183Progress with sdp (sip calls) the tenor does allocate dsp till it receives the codec information usually in the connect msg. During this priod it can not collect multisesion digits. A DSP has now been allocated to the call if multisession is enabled, and as a result this has been resolved.

3333 Hardware Registration takes too long

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

3347 Some calls through a DX have large amount of static

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

3362 BX occasionally boots as DX

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

3370 Add support for 1 second delays when using Pulse dial

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

P104-12-12

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

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

3099 RoHS support for CMS

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

3239 MaxTalkTime is not working if radius sends credit-time

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

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

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

3330 CMS: DSP Status Light always red

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

3345 Allow code to ignore license file when allocating DSP's for CallerID

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

P104-12-11

2707 Busyout for T1 & E1

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

It is implemented in var_config.cfg:

gkbusyout "counter"

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

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

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

2850 CH crashed when passed FACILITY message to ISDN

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

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

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

var_config.cfg entry:

CASNoDNISRerouteto xxxyyyzzzz

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

2891 Caller Name from Facility IE in SIP invite

The ability to receive callerID with name in a FACILITY from ISDN, and put it into a SIP invite (name in the From: field) has been added.

2933 config calling party TON and NPI for SS7

The ability to configure the TON and NPI of a call outbound on an SS7 trunk was added.

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

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

For example:

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

2996 Change Default Value of AniInfo for Outbound FGD Calls

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

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

3005 Terminating pound (#) in ivr with no dialplan makes dialed number private

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

3017 CNAME support for NI2 switch

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

3018 ss7 supports original called number, redirecting number and redirection number

CMS running SS7 was not passing through original called number, redirecting number and redirection number from incoming pstn line to outgoing pstn line. It will now.

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

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

3041 AS -> CRSP Slow Start calls Fail

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

3051 community string not being sent in SNMP traps

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

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

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

3060 LCRG forced routing and Ignorednis changes

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

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

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

3069 call failure when using external gk

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

3106 Delay login prompt until hardware registration is finished

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

3107 ss7 can not use routing server

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

3142 Garbage in ISDN setup CNAM for DMS switch

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

3151 Outbound Access level does not properly work with SIP

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

3172 PPPoE Termination Procedure Change

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

3193 Tenor watchdogs when radius receives corrupted access accept packet

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

3196 DHCP task crashed in case of one DNS server configuration

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

3215 AllowOnlyProxyCalls set to 0 after deleting DB

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

3253 BX MFG test broken - second card

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

3265 Small Tenors Only; Print MAC address in bootup scroll

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

3271 Unwanted "," at end of the column help

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

3277 With Pulse dialing line side mute never gets undone

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

3282 Busyout call setup fail by seizure timeout

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

3301 CRSP crashing on receipt of various H323/Q931 msgs containing H245ControlMessages

It was possible to crash the Call Relay SP if an H.323 Alerting, Connect or Facility message arriced with tunneled H245 Control Messages. This has been resolved.

P104-12-10

2450 Analog MFG Noise Test - NOT RUN AXM1600

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

P104-12-09

3006 new framer driver for Infineon version 2.2

Support for new Infineon framer chip.

3071 New serial number prefixes for RoHS boards

Serial number prefixes were incremented for RoHS compliant systems.

P104-12-08

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

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

P104-12-07

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

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

3045 Dropped calls due to Session Timer not cleaned up properly

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

2887 Help file updated for Brazil and India Progress tone support.

Minor change to the 3rd column help file to show the addition of Brazil and India progress tone support.

P104-12-06

2837 Change reversed

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

P104-12-05

3039 BX only - SIP FAX issue -Incorrect T38MaxBitRate encoded

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

P104-12-04

2804 Provide a temporary solution for Telnet Linemode Problem

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

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

3010 exception "can not send bye"

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

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

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

P104-12-03

2942 Japanese Caller ID Could Cause Disconnect

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

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

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

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

2745 TCRG Passthrough 2 not presenting IP route

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

2749 IPRG parameters not taken in count for Supp Services

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

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

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

2837 Use Extension for Registration and Username for Authorization

Note: This was removed in P104-12-06.

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

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

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

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

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

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

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

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

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

This does NOT apply to CMS or Call Relay SP.

2887 Implement Brazil and India Progress Tones

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

• 11 Brazil
• 12 India

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

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

2914 All analog - change fwd disconnect trigger

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

2934 Ring trip broken with some slic revisions

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

2936 SIP, BYE authentication fails with Vonage proxy

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

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

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

2965 Taiwan Static Dialplan

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

2999 Call Waiting causes crash after first call disconnects

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

P104-12-02

2753 DX resets when working with Routing Server

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

2857 Memory leaking on SIP Redirect call

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

2877 Add the version number info in reset.log file

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

2881 CMS crashes when it runs out of memory buffers

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

2885 DTMF payload values are reversed - we tx what we should rx

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

P104-12-01

2398 Remote call id is not parsed properly

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

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

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

2734 Hung calls for AttendedTransferKeystroke configured to non 'HU'

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

2741 Refer-To header does not follow IPDialPlan configuration

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

2749 IPRG parameters not taken in count for Supp Services

Parameters set in the IPRG were not being observed on Supplementary Service calls (such as transferred calls). This has been resolved.

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

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

2770 Transferred call is being aborted before Connected

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

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

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

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

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

2779 The IrDA receiver on the CMS should be turned off

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

2781 When MICA exceeded, incoming SIP calls not allowed

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

2784 Authentication with Asterisk broken

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

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

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

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

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

2795 Voice not heard when BYE received early in transfer

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

2801 Session Timer with Transfers causes one way digit transmission.

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

2803 StaticChannelConnection disappeared after CMS reboot

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

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

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

2815 CR60/CRSP only: 2833 packets are not dropped when passed to h323

When DTMF packets are converted from 2833 to H245, original RFC 2833 packets are not dropped. In most of the cases, since the other end is H323 GW, these packets will be dropped. But in cases where the H323 GW interprets both h245 and 2833 packets, it will generate two tones. This has been resolved, RFC2833 DTMF packets are extracted from the RTP stream when it has been converted to H.245.

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

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

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

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

2835 StopAcctID Prameter is Ignored When Start-Accounting is Used

The StopAcctID parameter was not taken into consideration when StartAccounting was enabled. This has been resolved.

2853 ivrtype 4 call gets connected in invalid authentication

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

2858 Change factory default for AllowOnlyProxyCalls

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