SmartWare R6.T Release Notes

Transcription

1 Patton Electronics Company, Inc Rickenbacker Drive Customer Deliverable Documentation Revision 1.00, March 30, 2017 Gaithersburg, MD USA Tel. +1 (301) Fax +1 (301) SmartWare R6.T Release Notes Build Series SmartWare is the embedded application software of the SmartNode series of VoIP Gateways and Gateway Routers. SmartWare provides a full set of IP routing features, advanced Quality of Service and traffic management features plus industry leading Voice over IP functionality including SIP and H.323 Released Build Numbers SmartNode 4110 Series R6.T Build SmartNode 4110S Series R6.T Build SmartNode 4120 Series R6.T Build SmartNode 4300 Series R6.T Build SmartNode 4400 Series R6.T Build SmartNode 4520 Series R6.T Build SmartNode 4600 Series R6.T Build SmartNode 4600 Series R6.T DSL Build SmartNode 4660 Series R6.T Build SmartNode 4670 Series R6.T Build SmartNode 4830 Series R6.T Build SmartNode 4830 Series R6.T DSL Build SmartNode 4900 Series R6.T Build SmartNode 4940 Series R6.T Build SmartNode 4950 Series R6.T Build SmartNode 4960 Series R6.T Build SmartNode 4970 Series R6.T Build SmartNode 4980 Series R6.T Build SmartNode 4990 Series R6.T Build SmartNode 5200 Series R6.T Build SmartNode 5400 Series R6.T Build SmartNode 5480 Series R6.T Build SmartNode 5490 Series R6.T Build SmartNode DTA Series R6.T Build Patton Electronics Company. All Rights Reserved. Copying of this document or parts of it is prohibited.

2 About this Release R6.T is a SmartWare Technology Release. Please see the White Paper about SmartWare software releases for more information about this terminology. Supported Products SmartNode 4110 Series (HW Version: 1.x, 2.x, 4.x, 5.x) SmartNode 4110S Series (HW Version: 1.x, 2.x) SmartNode 4120 Series (HW Version: 1.x, 2.x, 3.x) SmartNode 4300 JS Series (HW Version: 2.x) SmartNode 4300 JO Series (HW Version: 1.x) SmartNode 4400 JS Series (HW Version: 2.x) SmartNode 4400 JO Series (HW Version: 1.x) SmartNode 4520 Series (HW Version: 1.x, 2.x, 4.x, 5.x) SmartNode 4600 Series (HW Version: 1.x) SmartNode 4600 Large Series (HW Version: 1.x, 2.x) SmartNode 4660, 4670 Series (HW Version: 2.x, 3.x, 4.x) SmartNode 4830 Series (HW Version: 1.x, 2.x, 4.x, 5.x) SmartNode 4830 Large Series (HW Version: 1.x, 2.x, 3.x, 4.x) SmartNode 4900 JS Series (HW Version: 1.x, 2.x) SmartNode 4900 JO Series (HW Version: 1.x) SmartNode 4940 Series (HW Version: 5.x) SmartNode 4950 Series (HW Version: 5.x) SmartNode 4960 Series (HW Version: 1.x, 2.x, 3.x, 4.x, 5.x) SmartNode 4970, 4980, 4990 Series (HW Version: 1.x) SmartNode 5200 Series (HW Version: 6.x) SmartNode 5221 Series (HW Version: 4.x) SmartNode 5400 Series (HW Version: 5.x) SmartNode 5480, 5490 Series (HW Version: 1.x) SmartNode DTA Series (HW Version: 2.x, 3.x) Rev /28

3 History of Solved CTS Cases The following list refers to open cases in the Change Tracking System 'CTS'. This Build Series Deny VLAN configuration when port dsl is in ATM mode A user was able to configure the dsl port with VLANs when ATM encapsulation was set. However, the results do not show up in the running-config. This is because VLANs are not supported in ATM mode. This change allows the configuration of VLANs only when the port dsl is in EFM mode Crash at bootup due to exit command in the configuration The device crashed if, while parsing a hand-crafted startup-config at bootup, an exit command was parsed which tried to exit from the configuration mode. This could happen if the configuration contains an invalid mode command, in which case the consecutive exit command is not executed inside that mode, but in the next higher level mode. Now a warning message is printed into the event log if this case arises Support for the SN-DTA/2BIS2V4HP/EUI model Support for the SmartNode model SN-DTA/2BIS2V4HP/EUI has been introduced. Only software equal to or newer than Build Series must be loaded for proper operation of this device Error on command low-bitrate-codec Due to recent changes in hardware and software the command low-bitrate-codec has become obsolete on certain products because the DSP supports both low bitrate codecs (g723 and g729) at the same time. For these products the command had been removed from the software. This caused CLI parsing errors for existing configurations containing the command. These errors did not have an impact on the proper functioning of the device. In order to avoid any false errors reports, the command low-bitrate-codec is again accepted on these products. It is simply ignored where not needed and it does not appear in the runningconfig either SmartNode devices occasionally crashed when processing mis-ordered rtp-map telephony-event Fixed an issue where SmartNode devices crashed in rare cases of SIP-to-SIP calls (back-toback user-agent scenario). The problem occurred when a peer sent SDP responses with the telephony-event rtp-map in a different order than the corresponding payload-type EFM interface drops all packets with certain non-fcs-capable DSLAMs Rev /28

4 Some DSLAMs expect the CPE to remove Frame-Check Sequence (FCS) from all Ethernet packets. So far, the EFM port configured in ATM mode did not remove the FCS. Most DSLAMs configured in non-fcs mode were still fine with this, but some older DSLAMs did not accept the packet. As a consequence, the PPPoE over ATM connection could not be established even though the link was up. More precisely, the PADI frames that the Patton device sent in the PPPoE discovery stage were dropped by the DSLAM. In order to be compatible with the few restrictive DSLAMs a new configuration command has been added in the port dsl mode. The [no] fcs command configures whether the Patton device shall send Frame-Check Sequences (FCS) (default: on). See the New Configuration Commands section for more details Memory-leak when sending CANCEL message over SIP For every CANCEL request sent over SIP, SmartWare lost a small part of its memory. Due to the small size of the memory leak, there was no immediate impact. Over time the memory fragmentized and caused the memory manager to slow down. This could have a negative impact on the call performance and ultimately lead the device into an unresponsive state where no calls would be processed anymore. Build Series Endless call-waiting on second call (FXS) When a second incoming call was waiting on an FXS interface, this second call was dropped after 15 seconds and the call-waiting indication beep-tone to the party in call, receiving the second call, heard only 2 beeps. Now the call-waiting timeout is infinite. This means that the second call remains in ringing state until the caller gives up or the called party answers this call. Furthermore the call-waiting indication "beep-tone" to the called party is also played until the call is answered or the call is given up Quoted SIP diversion-header parameters were ignored Quoted parameters of received SIP diversion headers were ignored and the device's default parameters were used instead. The SIP stack now properly parses and processes SIP diversion-headers parameters, whether they are quoted or not Endless dial tone on second call (FXS) When a call was put on hold on an FXS interface, the user heard the dial tone for 7 seconds only and afterwards the call was released and the release tone played. As a consequence the address-completion timeout had no effect if it was set to a value greater than 7 seconds (12 seconds by default). Now the dial tone timeout is infinite and effective if the user configures no address-completion timeout in the configuration mode context cs. Otherwise the configured address-completion Rev /28

5 timeout takes precedence and terminates the played dial tone when expired (12 seconds by default) Ignored the recorded-units parameter in a SIP INFO AOC header Fixed an issue where the device ignored the 'recorded-units' parameter of a received SIP INFO message with an AOC header SmartNode sent wrong response after a SIP ReINVITE with unsupported codecs When the SmartNode received a SIP ReINVITE with unsupported codecs, it responded correctly with a '488 Not Acceptable' message. However, if the device received another SIP ReINVITE later with supported codecs, it failed and responded with a '100 Trying' message. Now, the device correctly responds with a '200 OK' message to a SIP ReINVITE with supported codecs after a ReINVITE with unsupported codecs SIP handover on RTP loss with Hunt-Group The overall use-case is to provide a fallback for SIP calls. The configuration of the different destinations is done by creating multiple SIP interfaces and routing calls to them from a huntgroup service. A new command for specifying how the fallback calls are referencing to the original call is added. See the New Configuration Commands section for more details. This feature requires the sip-handover license Unable to exit from the configuration mode port dsl Fixed an issue whereby the exit command in the configuration mode port dsl was missing on all products with the EFM interface Provisioning now supports HTTP chunked transfer-encoding Some web servers are always using chunked encoding to transfer data to the client, independently of the amount of bytes that have to be transmitted. SmartWare now supports HTTP based provisioning using Transfer-Encoding chunked. Build Series Support HTTP without Content-Type header When doing provisioning with HTTP locations or issuing a file download with the copy command from HTTP, then the file could be only downloaded successfully when the server added in its answer a Content-Type header. With this implementation SmartWare now assumes a default content type of "text/plain". This allows to provision and copy from HTTP servers which do not always add a Content-Type header RTP payload-type translations for DTMF RFC2833 events Rev /28

6 When no DSP was involved, for example in a Session Border Controller, the RTP payload type for DTMF RFC2833 events was tunneled even if the configured payload-type on one side was different from the other side. Now the RTP payload-type for DTMF RFC2833 events always matches the configuration of the corresponding VoIP profiles Fallback to G.711 when T.38 fax is rejected in 200 OK Improved interoperability in a specific T.38 scenario: 1. The SmartNode is configured to send T.38 fax and to fall back to G.711 in the case the peer does not support T The peer does not support T.38 but instead of rejecting the offer with a 4xx, it rejects it by setting the port to zero in the SDP of the 200 OK. If these conditions are met, the SmartNode will now re-invite with a G.711 offer SIP UPDATE mid-dialog fails In a SIP UPDATE message that is received mid-dialog, the SDP content was not interpreted correctly and resulted in an error in most cases. The call was subsequently terminated with a BYE request immediately after reception of the UPDATE message. Build Series SDP support for SIP UPDATE Support for codec renegotiation after receiving a SIP UPDATE with SDP content has been implemented. What to expect - For an UPDATE coming during an Early-Dialog, the SmartNode will answer: If the call is SIP-TDM (i.e. transcoding, ending calls locally) -> 200 OK, and will update codecs and privacy headers. If the call is SIP-SIP (ip-ip codec negotiation) -> 491 Request Pending, and will not update anything. If the call is SIP-SIP and the UPDATE message does NOT include SDP -> 200 OK, and will update only privacy headers. For an UPDATE coming Mid-Dialog, the SmartNode will answer: If the call is SIP-TDM (i.e. transcoding, ending calls locally) -> 200 OK, and will update codecs but not privacy headers (not needed). If the call is SIP-SIP (ip-ip codec negotiation) -> It will trigger RE-INVITE codec renegotiation with the peer and will forward back its answer (i.e. 200 OK, 488 Not Acceptable Here). Rev /28

7 If the call is SIP-SIP and the UPDATE message does NOT include SDP -> 200 OK, and will update only privacy headers Improve feedback for invalid provisioning locations The provisioning profile allowed specifying archive files as provisioning locations. These led then to failures in the provisioning with misleading error descriptions. Now the configuration of these locations itself is denied to make it clear that these are not supported at all Start SIP re-registration after half of the expiry time SmartWare started SIP re-registrations only short before the actual registration expiry time. In some cases the time needed for DNS resolution lead to an expiration of the registration before the renewal was completed. Therefore the behavior is now changed to start the re-registration after half of the expiry time, to have enough time for renewal before the registration actually expires PPP LCP magic number not random The SmartNode s magic number negotiated with PPP LCP was not determined randomly. The magic number is used to detect link loops, so it should be changed for new connections. Now the magic number changes to a new random value for every new connection attempt PPP LCP MRU cannot be forced above the underlying link MTU In some rare situations the LCP MRU must be configured to 1500 even though the actual link MTU is smaller (e.g bytes). One reason for this is when a PPPoE connection is terminated by two L2TP-chained BRAS (Cisco and Juniper, see one of the BRAS will drop PPP frames if the MRU is not negotiated to the tunnel MTU, which typically is at 1500 bytes. SmartWare already provides a configuration command in the PPP profile to force the MRU to 1500 bytes: mru min 68 max 1500 default 1500 ignore-link However, the ignore-link parameter, which should force the MRU to the specified value despite any underlying link restriction, only worked for the max but not for the default parameter. Now SmartWare correctly negotiates an MRU of 1500 with the command above PPP LCP Echo-Request payload size cannot be configured In some rare situations the payload of LCP Echo-Requests must be greater than zero bytes: One such case is if the SmartNode creates a PPPoE connection over the EFM-DB in ATM mode to a BRAS that does not pad the Ethernet frames. In this case very small ATM cells are dropped, including Echo Requests without payload. We changed the default payload size from 0 to 8 bytes and made the payload-size configurable with the lcp-echo-request command in the PPP profile (see New Configuration Commands section below). Build Series Speed test causes a crash Rev /28

8 When executing the speedtest command the SmartNode would crash if the used file was bigger than a certain size. This has been corrected and the files used for speed test can have an arbitrary size. Build Series Incorrect codec order in SDP negotiation in special use-case When IP-IP codec negotiation is enabled, SmartNodes respect the codec preference order of the peer side (the one sending the 200 OK with the supported codecs by both sides) and forward it back to the caller. However, a problem occured when the locally configured codecs in the VoIP profile were the same -and in the same order- as the ones received from the peer in the 200 OK. In this case, the codec order was not respected and it lead to incompatibility and no voice in the call. This issue has been fixed Broadsoft Flash-Hook signaling in SIP was ignored Broadsoft servers use a specific format for transmitting a Flash-Hook signal over SIP. This consists of a SIP INFO message with the content type of "application/broadsoft" and a message body with the text "event flashhook". This specific format of signaling a Flash-Hook was ignored SmartWare does not handle/answer SIP UPDATE requests with SDP content With SmartWare R6.T the SIP Allow header in INVITE requests and 200 OK responses to INVITEs was introduced (CTS 12505). This header also lists the UPDATE method. Although SmartWare supports the UPDATE method for privacy header updates it does not support SDP content in UPDATE requests. Some peer User Agents however sent SDP content in UPDATE requests when SmartWare announced the method in the initial INVITE, which caused the SIP call to fail on our side. Therefore the UPDATE method has been removed from the Allow header of INVITE requests and 200 OK responses to INVITEs. Build Series Introduce ATM encapsulation for G.SHDSL.bis-EFM interface The SmartNode models with a G.SHDSL.bis-EFM interface (model codes ending with /2G and /4G) can now be configured for ATM mode. The ATM mode supports one PVC only. In addition to new commands the structure of existing commands has changed. See more details in the New Configuration Commands section SIP Allow header field missing in INVITE and 200 OK According to RFC3311 a SIP user agent should add an Allow header field to the INVITE request and to the 200 OK response. So far SmartWare did not populate those messages with Rev /28

9 the Allow header field. Although this header is optional it caused some interoperability problems in early media scenarios if the peer SIP endpoint rejected to use UPDATE messages without SmartWare pre-announcing this method. SmartWare now adds the Allow header field to the INVITE and the 200 OK message and announces support for the NOTIFY and all other methods supported. Note that SmartWare does not support UPDATE messages with SDP content yet Provisioning profile added to factory config for several products The provisioning profile was missing in some factory configurations. This caused some problems for customers using our EMS or using the redirect server with devices installed in the field. Doing a factory reset of these devices was a problem without having the call home functionality included in the factory default configuration. The provisioning profile has been added to the factory config for the products SN4830/LL, SN4830/LLA, SN4940, SN4960, and SN Low-bitrate-codec configuration missing on SN-DTA and SN4120 On certain SmartWare devices only one low bitrate codec can be active at the same time. For configuring this, the comand low-bitrate-codec (g723 g729) in the mode system/ic voice 0 is used. On the models SN-DTA and SN4120 this command was missing and is now made available No music on hold Hold music was not transparently passed through the Patton device in some SIP-to-SIP call scenarios. The problem only appeared if after the on-hold Re-INVITE message the A-side sent another Re-INVITE message to change media-properties for the hold music. With this fix all in-band data is transparently passed through the Patton device if the holding phone signals to have in-band data available. Build Series SNMP Traps for FXO Port Line Up Reload graceful fixed and enhanced: introducing graceful timeout Preferred transport protocol for penalty-box SIP options is now configurable Build Series Support for SIP Reason header SIP overlap reception lost dialed digits FXO call dropped on early flash-hook signal H.323 calling-name information not passed to outgoing call Rev /28

10 12484 Show hardware information of G.SHDSL submodule SmartNode crash with debug sip registration Wrong source port in SIP REGISTER SIP re-register after contact expired Build Series Support for SN4112S Build Series Commands on EFM interface did not provide any help text Fixed back-to-back codec negotiation when codec preference order was altered by peer Losing memory when using PRACK with SIP SIP stack upgrade to V Build Series Fax detection during non-transcoded SIP-to-SIP calls Better handling of failing EFM daughterboard upgrade Control SDP announcement in provisional responses Digest Authorization scheme is now case-insensitive Build Series Add port option to spoofed commands in SIP gateway Send INVITEs to registrar SNR and attenuation values for devices with ADSL interface Build Series Support for CAMA emergency E911 protocol Tunneling of ISDN UUI1 information over SIP Support for R2 to R2 call scenario Addition of MWI capability on FXS with battery-reversal Build Series Rev /28

11 12110 ISDN calls are now visible through SNMP Configuration of G.SHDSL card is correctly applied during first boot Firmware upgrade through HTTP fixed Build Series Extended REFER timer for call-diversion Accept 406 status code as fax failover trigger PPPoE and VLAN configuration fixed for EFM SmartNodes Early-media fix SIP endpoints leak Build Series HTTP/HTML provisioning fixed Media detection timeout caused loss of voice Correct upgrade status displayed when upgrading the EFM card via CLI New FXS profile supporting current of 30 ma SNTP servers updated in factory-config EFM daughter card upgrade/downgrade support for new card firmware EFM port status correctly displayed with new card firmware Corrected EFM port configuration after reboot Enhanced link configuration command for EFM daughter card EFM configuration option on the web GUI for SN5490 models Build Series Corrected reception and transmission of SIP AOC headers Rejected INVITE with ambiguous SDP content header Failed to send SIP BYE message Build Series SIP Trusted Host enhancement Configurable time zone for CDR messages (accounting data) Rev /28

12 12276 Removed no form of address-translation incoming-call commands Calling-Name support for DMS SIP AOC-E (advice of charge) not sent DTMF tones propagation on the SIP receive side Changed verbiage in GUI for reloads More tolerance for SDP parsing of received messages calling-redir-e164 cannot be configured in web GUI Web GUI display issue: AOC SIP-header Support for upgrade to Trinity with multiple shipping-configs E1T1 channel-group encapsulation voiceband not available Build Series EFM card firmware version in show command Show command formatting for EFM card Crash in Advice of Charge scenarios when using SIP headers Random crash during calls Identical SIP-Header X-Org-ConnID for two simultaneous calls X-Org-ConnID with call diversion Build Series SIP B2BUA dynamic registration support Crash on terminating SIP call with DNS Parameters not applied to EFM daughter card during startup Local tax pulse generation on FXS Inbound call to FXO, ringing not always dispatched to the call-control SIP NOTIFY check-sync Screening and presentation indicators treated incorrectly for diverted calls Wrong help text for the annex-type parameter of EFM daughter card Build Series Rev /28

13 12220 SIP stack upgrade to V Fixed crash when receiving 302 moved temporarily SIP message SIP registration expiry time is wrong Radius authentication profile for SSH is not saved in the configuration Traffic-class not set on stack-generated SIP messages FXS port stops ringing the phone SIP AoC not sent when overlap dialing is enabled SN4660/SN HW QUEUE FULL error while receiving DTMF tone Auto-provisioning fails when DHCP option 66 is missing EFM, communication broken with the card Build Series Support for call deflection on ISDN Incorrectly encoded Calling-Name for NI-2 and DMS SIP AOC Header support New SIP Header X-USE302: YES Unique SIP connection ID for calls Invalid BGP identifier Configurable calling party or facility IE on ISDN Broken policy-routing for SIP calls over UDP Determine reachability with SIP OPTION requests SIP penalty box behavior improved Missing identity header for empty calling party number Identity headers not parsed for SIP overlap dialing calls SIP-Gateway wrong address lookup G.SHDSL link UP with cell delineation error or with training error Missing shipping-config after upgrade to Trinity SDP ptime attribute Rev /28

14 12245 Rejected INVITE when Call-ID contains a < character SIP request URI length limitation Local RAS port is configurable for H H.323 Gatekeeper fallback not working License installation fails on Trinity after upgrade from SmartWare Wrong MOS value for G Compatibility with EFM DB V3.3.1 Build Series PSTN configuration on a R2 interface was cleared after a few calls ISDN interface is capable of triggering actions Alcatel signaling method for flash-hook SIP info message SIP request not being sent Downloadable CDR records SNMP allowed network Display error of ISDN binding Adaptions to maximal possible SIP sessions on ESBR Availability of H.323 in ESBR products Support for EFM daughter card (Rev A) Duplicate T.38 attributes in SDP Logging error when a WAN card is detected but unknown Crash in SIP transfer scenarios Build Series Network and user provided secondary calling party number Encryption key provisioning PRACK not working for forked INVITE Incoming calls refused with 481 after PPP cycle Unknown SAPI message on E1 port Rev /28

15 12136 Crash during startup with large configurations (also fixes 12113, 12085, 12132) SIP overlap dialing causing unexpected reboot Reset log shows HW watchdog as Power off/man reset Changing SSRC causes one-way voice connection SIP register not working in combination with loopback interface ASN1 AOC not working Crash when receiving a SIP answer without Via header AAA framework problem Build Series Ethernet speed capability for manual settings Action script trigger for SIP registration G.SHDSL interface software upgrade failed RTP payload type conflict G.SHDSL mode auto detection fixed Support for short delay re-invite in SBC scenario SN4991 Models with ADSL interface supported RTP through VPN broken Wrong drop cause reported by SIP endpoints, resulting in failed call hunting Spelling error corrected on BGP configuration web page SN4660/70 cooling fan speed adjusted Build Series SIP supports TCP flows according to RFC SNMP OID for DSL card firmware version Support for SN4832/LLA and SN4834/LLA models New NTP server in factory-config SIP AOC XML support Sending tax-pulses on FXS for AOC Rev /28

16 12039 Incorrect answer to SIP INFO message G.SHDSL software upgrade progress indication updated Increased timeout for redirection service Session Progress not being forwarded in SBC scenarios Minimal SIP registration time Added support for SFP interface (Fiber interface) SIP calls over TCP failed DSL supervisor log notifies wrong DSL line state Missing user part from SIP contact header Build Series Invalid REGISTER request when spoofed contact is set SIP multipart message support Support for 4300/JO and 4400/JO products Media detection timeout Enhanced AAA debug logs New Patton corporate style applied to web interface Wrong help text for blink command Limit packets to prevent SIP overload condition SIP register back-to-back command removed Added option DHCP.66 error message when not available Concurrent Dynamic IP Configuration (DIC) removes default gateway Flash hook on FXO interface broken Build Series Configuration option for caller-id checksum verification on FXO interface G.SHDSL interface: service-mode auto-detection ISDN status errors on Web UI MWI on FXS not working Rev /28

17 11860 SIP re-register not working ADSL annex A/B/M Improved dial on-caller-id on FXO Layer 2 COS for PPP and PPPoE control packets Administrator login to administrator exec mode MWI Subscription failing H.323 Call Resuming Dial tone played a second time G.S line rate negotiation fails at high distance FXO dial-tone detection Invalid SDP offer in SIP provisional response Build Series DTMF caller ID transmission on FXS Crash when a # character is present in SIP contact header Support for p-called-party-id header Incoming SIP calls refused with 481 after an IP address change over PPP Cooling fan always running at full speed on SN No final answer when receiving BYE SN4660/SN4670 Rev C and Rev D support Ethernet switch problem on SN4660/SN Enhancement of software upgrade procedure SIP Hold/Unhold behavior SIP q-value of SN-DTA clock synchronization SIP 503 error handling Broken T.38 transmission Redirection service for provisioning supported in factory configuration Rev /28

18 11950 Modified memory layout for SDTA, SN4552, SN4554 and SN FXS hanging calls Basic PRACK scenario broken Spurious error messages from G.S interface Missing command payload-rate on SN4660/SN Verbose software upgrade of G.S interface card Removed support for hardware version 4.x for SN4552, SN4562, SN4554, S-DTA Build Series Web interface generates a new identity PPPoA on G.S interface SN-Web page refresh causing reboot Enhance DSL status display DTMF Caller-ID reception on FXO Call transfer issue fixed CED-Tone Net Side Detection enhancement Trusted SIP hosts to improve security HTTP download failure blocks the SmartNode T.38 Fax transmission killed by CNG tone Auto-provisioning: redirection target reordering Missing strict-tei-procedure command Provisioning: prevent downloading incompatible configuration New provisioning placeholders Auto-provisioning factory-config (Redirection service support) Wrong G.S port state displayed SN5200 hardware-version 6.X support CED-Tone Net Side Detection not working Crash when downloading G.S firmware with web interface Rev /28

19 11850 Abnormal call termination due to misinterpreted SDP data Auto-provisioning: Target configuration without leading http or tftp Ethernet lockout on SN4660/70 Build Series Echo Cancellation with RBS ETSI Caller-ID not detected on FXO Q-value support for SIP REGISTER Music on Hold not played to SIP side Removed DSL options b-anfp and a-b-anfp Support for SN4660/SN4670 hardware Revision B Broken modem transmission using H Removed SIP Contact header verification in 200 OK messages First received IPCP frame dropped in during PPP connection establishment Added SDP attributes X-fax and X-modem support Forced Fax/Modem bypass Remote Early-H.245 initialization Wrong mapping table in R SN-DTA and SN4120 allow usage of g729 codec Build Series Fixed T.38 packets traffic-class BRI Daughter-Board HTTP User Agent enhancement G.SHDSL power and reset spikes Syslog-client no remote command crash Fixed display of mtu and mru max values in running-config Improved clocking precision for SN-DTA and SN Timestamp enhancement for milliseconds Rev /28

20 11685 Enhanced spoofed contact to accept hostname Fax T.38 not working with H Wrong facility callrerouting packet in case of CFU Missing facility from running-config New DSL supervisor mode observe Build Series Locking DNS records for SIP requests Improved configuration and display of bit rate for 4-wire G.S interface HTTP 302 Redirection now supported for provisioning Additional parameters in G.SHDSL status: SNR, Loop Attenuation, Port States Clock synchronization improvements Fixed police traffic class configuration option RTP payload type configuration Received maddr parameter is reflected in contact header Proper differentiation between SN4660 and SN4670 product types Spurious errors reported by SIP and SDP protocols BRI CRC Failures Potential memory leak in SIP state machine Support for SN-DTA and SN4120 series Global power-feed for BRI Performance improvements Wrong factory-config for SN products with DSL Rev /28

21 Caveat - Known Limitations The following list refers to open cases in the Change Tracking System 'CTS' CTS2236 Only G.723 high rate (6.3kbps) supported by H.323 (receive and transmit). CTS2702 TFTP may not work with certain TFTP servers, namely the ones that change the port number in the reply. When using the SolarWinds TFTP server on the CD-ROM this problem will not occur. CTS2980 With 10bT Ethernet ports, only the half duplex mode works. (With 10/100bT Ethernet ports, all combinations work.) CTS3233 The SolarWinds TFTP server version (1999) does not correctly handle file sizes of n * 512 Bytes. Use version (2000) or higher. CTS3760 The SIP penalty-box feature does not work with TCP. When the penalty box feature is enabled, the TCP transport protocol must be disabled using the no transport tcp command in the SIP gateway configuration mode. CTS3924 Changing a call-progress tone has no effect. Adding a new call-progress tone and using it from the tone set profile works however. CTS4031 The Caller-ID message length on FXS hardware with Chip Revision numbers below V1.5 is restricted to 32 bytes. If the message is longer the message will be truncated. The FXS Chip Revision can be displayed using the show port fxs detail 5 CLI command. CTS4038 When doing 'shutdown' and then 'no shutdown' on an ethernet port that is bound to an IP interface that receives its IP address over DHCP, the IP interface does not renew the lease. CTS4077 Using the command terminal monitor-filter with regular expressions on systems under heavy load can cause a reboot. CTS4335 The duration of an on-hook pulse declared as flash-hook has been raised from 20ms to 1000ms, to cover the most country specific flash hook durations. Existing installations should not be affected, as all on-hook pulses lower than 1000ms are declared as flash-hook, including the previous default of 20ms. Rev /28

22 However, care should be taken in analog line extension applications, to make sure that the flash-hook event is allowed to be relayed over SIP or H.323. This can be achieved by disabling all local call features in the fxs interface on context cs: no call-waiting, no additional-call-offering, no call-hold. CTS10392 The internal timer configuration command is only able to execute commands that produce an immediate result. Some commands that execute asynchronously cannot be executed by the timer. The following commands (among others) cannot be executed by the timer: ping traceroute dns-lookup copy any kind of files from or to a TFTP server reload without the forced option CTS10553 The command no debug all does not fully disable the ISDN debug logs. As soon as any other ISDN debug monitor is enabled, all the ISDN monitors that were disabled by no debug all are re-enabled. CTS10586 The codecs G.723 and G.729 cannot be used at the same time on all platforms, except on the SmartNode CTS10610 SmartNode 4960 Gigabit Ethernet does not properly work with Dell 2708 Gigabit Ethernet Switch. A work-around is to configure 100Mbit. CTS10730 Due to memory limitations it is not possible to download a software image to the SN4552 when two SIP gateways are active. CTS11114 On SN46xx units it can happen that there are more open phone calls requiring a DSP channel than DSP channels are available. This leads to the situation where a phone connected on a bri port rings and has no voice after the user picks it up. To limit the number of calls using DSP channels it is suggested to create a limiter service where each call from and to a bri port has to pass. When the total number of calls on the bri ports is limited to the number of DSP channels each call is going to have audio on picking up. CTS11786 On older SmartNodes the two debug monitors debug media-gateway rtp and debug call-control print out incorrect RTCP jitter values. CTS11816 Rev /28

23 The command call-control call drop <call> is not behaving as expected. It drops all calls but does not completely teardown all internal structures. Consequently the call numbers of the dropped call cannot be used anymore for further calls after executing this command. The same is true for the Drop all button on the web interface on the Active Calls tab of the Call-Router section. CTS12027 The following configuration may create duplicate packets: If one physical ethernet port is bound to two IP interfaces with different IP addresses and on both IP interfaces a SIP gateway is bound and some static routes are configured, then the SIP gateways may receive duplicate UDP packets. Rev /28

24 New Configuration Commands The commands documented in the Release Notes only cover new additions which are not yet included in the Software Configuration Guide for R6.9, available from Current Revision: Part Number: 07MSWR69_SCG, Rev. A EFM interface drops all packets with certain non-fcs-capable DSLAMs First appeared in build series: The EFM port of Patton devices (if configured in ATM mode) always sent Frame-Check Sequences (FCS) with Ethernet frames (for PPPoE). While most DSLAMs in bridged-non-fcs mode still accept frames with FCS, some DSLAMs strictly drop FCS frames. For those cases, the following CLI command can be used to remove FCS from sent Ethernet frames. Mode: port dsl <slot> <port> Step 1 Command node(prt-dsl)[<slot>/<port>]# [no] fcs Purpose Configures whether a Frame-Check Sequence (FCS) is added to the sent Ethernet frames if the port operates in ATM mode (default: enabled). (Some DSLAMs configured in bridged-non-fcs mode discard PPPoE frames with an FCS attached.) SIP handover on RTP loss with Hunt-Group First appeared in build series: The overall use-case is to provide a fallback for SIP calls. The configuration of the different destinations is done by creating multiple SIP interfaces and routing calls to them from a hunt-group service. The huntgroup does its normal hunt operation until the call can be established. When the handover is configured, the hunt-group remains in the call and provides a fallback mechanism if the RTP connection is lost. The hunt-group then establishes a new call to the next destination with the objective to replace the call with the broken RTP connection. If that new call is routed to SIP, it is going to have a Replaces header in the INVITE message to signal the handover to the involved SIP server. No audio data is streamed to the originating user until the media of the new call is fully connected. Rev /28

25 The old established call with the broken media connection is terminated once the new call has been established for a few seconds. Note: This feature requires the sip-handover license. The detection of broken media connection only works for SIP calls. The Replaces information is only present if the fallback destination is a SIP server. Mode: context cs <name>/service hunt-group <name> Step 1 Command node(svc-hunt)[<name>]# call-handover onmedia-loss Purpose Configures the hunt-group to stay in the call and issuing a replacement call when a loss of RTP stream is detected. Requires license: sip-handover. Default: disabled. In addition the time for RTP loss detection can be configured: Mode: profile voip <name> Step 1 Command node(pf-voip)[<name>]# media lossdetection <ms> Purpose Configures the time in milliseconds since the last received RTP packet until the media connection is considered broken. Default: 1000 ms. Range: ms. If the new call is routed to SIP, the reference to the old call can be signaled in the INVITE message to the SIP server in different ways: With a Replaces header: The new call has a Replaces header according to RFC This is the default signaling method. o Replaces: cab569ce83a9026ab;from-tag=1854a30e67;to-tag=6b357f4fa3 With Patton specific headers: The new call has two proprietary headers, one signaling the handover with "true" and one to provide the value of the Call-ID header from the old call. o o X-Patton-Call-Failure: true X-Patton-Call-ID: cab569ce83a9026ab No signaling: No specific information about the old call is added to the new call s INVITE message. Mode: context cs <name> / interface sip <name> Step 1 Command node(if-sip)[<name>]# [no] call-handover signaling (replaces-header patton-header) Purpose Configures how the new call provides information about the old call. Default: replaces-header. Rev /28

26 Documentation Please refer to the following online resources: Software Configuration Guide SmartWare Release R6.9: SmartWare Configuration Library: Web Wizard Platform for Configuration generation: SmartNode Utilities: General Notes Factory Configuration and Default Startup Configuration The SmartNodes as delivered from the factory contain both a factory configuration and a default startup configuration. While the factory configuration contains only basic IP connectivity settings, the default startup configuration includes settings for most SmartWare functions. Note that if you press and hold the system button (Reset) for 5 seconds the factory configuration is copied onto the startup configuration (overwrite). The default startup config is then lost. IP Addresses in the Factory Configuration The factory configuration contains the following IP interfaces and address configurations bound by the Ethernet ports 0/0 and 0/1. interface eth0 ipaddress dhcp mtu 1500 interface eth1 ipaddress mtu 1500 Rev /28

27 How to Upgrade 1. You have the choice to upgrade to R6.T with or without the GUI functionality. To upgrade to R6.T without the GUI functionality, enter the following command (telnet, console): copy tftp://<tftp_server_address>/<server path>/b flash: To upgrade to R6.T with the GUI functionality, enter the following command (telnet, console): copy tftp://<tftp_server_address>/<server path>/bw flash: 2. Load Patton-specific settings (preferences), if available: Extract the files b_patton_prefs and Patton.prefs into the same directory on the TFTP-server. copy tftp://<tftp_server_address>/<server path>/ b_patton_prefs flash: 3. Reboot the SmartNode afterwards: reload Notes about Upgrading from Earlier Releases Note that SmartWare Release R6.T introduces some changes in the configuration compared to Release R5.x, especially in the domain of FXO and ISDN. Please refer to the SmartWare Migration Notes R5 to R6 available at upgrades.patton.com. Rev /28

28 How to submit Trouble Reports Patton makes every effort to ensure that the products achieve a supreme level of quality. However due to the wealth of functionality and complexity of the products there remains a certain number of problems, either pertaining to the Patton product or the interoperability with other vendor's products. The following set of guidelines will help us in pinpointing the problem and accordingly find a solution to cure it. Problem Description: Add a description of the problem. If possible and applicable, include a diagram of the network setup (with Microsoft tools). Product Description: When reporting a problem, always submit the product name, release and build number. Example: SmartNode 4960 V1 R6.T Build This will help us in identifying the correct software version. In the unlikely case of a suspected hardware problem also submit the serial number of the SmartNode (s) and/or interface cards. Running Configuration: With the Command Line Interface command 'show running-configuration' you can display the currently active configuration of the system (in a telnet and/or console session). When added to the submitted trouble report, this will help us analyze the configuration and preclude possible configuration problems. Logs and Protocol Monitors: Protocol traces contain a wealth of additional information, which may be very helpful in finding or at least pinpointing the problem. Various protocol monitors with different levels of detail are an integral part of SmartWare and can be started (in a telnet and/or console session) individually ('debug' command). N.B.: In order to correlate the protocol monitors at the different levels in SmartWare (e.g. ISDN layer3 and Session-Router monitors) run the monitors concurrently. Network Traffic Traces: In certain cases it may be helpful to have a trace of the traffic on the IP network in order to inspect packet contents. Please use one of the following tools (supporting trace file formats which our tools can read): Ethereal (freeware; Your Coordinates: For further enquiries please add your address and phone number. Rev /28