smartware r6.t release notes - patton electronics · in the sip interface it is possible to...

Patton Electronics Company, Inc.

7622 Rickenbacker Drive

Gaithersburg, MD 20879 USA

Tel. +1 (301) 975-1000

Fax +1 (301) 869-9293

[email protected]

http://www.patton.com

2011 Patton Electronics Company.

All Rights Reserved. Copying of this document or parts of it is prohibited.

SmartWare R6.T Release Notes

Build Series 2018-01-12

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 2018-01-12

SmartNode 4110S Series R6.T Build 2018-01-12






SmartNode 4600 Series R6.T DSL Build 2018-01-12




SmartNode 4830 Series R6.T DSL Build 2018-01-12












SmartNode DTA Series R6.T Build 2018-01-12

Customer Deliverable Documentation

Revision 1.00, March 7, 2018

Rev. 1.00 07-03-18 2/31

About this Release

R6.T is a SmartWare Technology Release. Please see the White Paper about SmartWare software

releases https://www.patton.com/whitepapers/SmartWare%20Release%20Strategy%20Whitepaper.pdf

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 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 Large Series (HW Version: 1.x, 2.x, 3.x, 4.x)

SmartNode 4900 JS Series (HW Version: 1.x, 2.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 5480, 5490 Series (HW Version: 1.x)

SmartNode DTA Series (HW Version: 2.x, 3.x)

https://www.patton.com/whitepapers/SmartWare%20Release%20Strategy%20Whitepaper.pdf

Rev. 1.00 07-03-18 3/31

History of Solved CTS Cases

The following list refers to open cases in the Change Tracking System 'CTS'.

This Build Series 2018-01-12

12622 Deadlock at boot up when syslog tries to log to a remote host

When syslog is configured to log to a remote host using a host name, the DNS resolution could

generate a deadlock at boot up. The device would then end up in an endless reboot.

This issue has been resolved and now syslog remote host name DNS resolution cannot

generate deadlock anymore.

12624 Privacy not handled if P-Preferred-Identity or P-Asserted-Identity header is missing

The Privacy header was ignored if the P-Preferred-Identity or P-Asserted-Identity header was

missing. In this case no Privacy header was set in the outgoing SIP packet. Now the outgoing

packet has the Privacy header set if it is present on the incoming packet or if a P-Asserted-

Identity or P-Preferred-Identity header is present.


12619 Penalty box triggers action script on target reachability

The penalty box of a SIP interface can be configured to periodically check the reachability of the

remote by sending SIP OPTIONS. This feature can be used to avoid timeouts when sending the

SIP INVITE message.

Two new Action Script events are also generated with the penalty box feature:

1. OPTIONS_TARGET_REACHABLE

2. OPTIONS_TARGET_NOT_REACHABLE

The first event is triggered when the Patton Device receives an answer to the SIP OPTIONS

request sent to the remote for the first time. The second event is triggered when the first event

has already been triggered and no answer is received to the SIP OPTIONS request.

These two new Action Script events can be used to change the configuration dynamically

according to the reachability of SIP remotes. See the New Configuration Commands section for

more details.

12620 E1 MFR2 call dropped with Argentina profile

In Argentina metering pulses might be sent over MFR2. The Smartnode did not recognize these

metering pulses and treated them as ClearBackward signal, which was the cause of the

dropped call.

Rev. 1.00 07-03-18 4/31

Now the Smartnode, when configured to use Argentina R2 profile, recognizes these metering

pulses and ignores them.

12631 Upgraded DSP firmware for SN494x/5x/6x/7x/8x/9x, SN548x/9x and SN466x/7x devices

For SN494x/5x/6x/7x/8x/9x and SN548x/9x devices the VoIP-stream processing DSP has been

upgraded to version 700.36 and for SN4666x/7x devices to version 700.25.

12633 SIP call with relayed T.38 fax transmission failed

Under the following conditions a SIP-to-SIP T.38 fax call would fail when the SIP Re-INVITE to

T.38 is received:

codec negotiation is disabled in the voip profile

"fax transmission relay t38-udp" is configured in the voip profile.

No final response to the Re-INVITE is ever sent. The call is ultimately terminated by the peer

with a BYE request.

12637 SIP call IP-IP codec negotiation with unsupported codec in ReINVITE did not work

If codec negotiation is enabled, a SIP-to-SIP would fail when a SIP Re-INVITE with an

unsupported codec is received.

Instead of rejecting the unsupported codec with a SIP 488 error message, no final response to

the Re-INVITE is ever sent.


12585 Increased Number of Retries for HTTP Provisioning

For provisioning locations using the HTTP protocol, the files and resources are downloaded by

issuing an HTTP request to the server. When this initial request is not answered by the server

within 3 seconds, the request is repeated. This could happen up to 3 times with a maximum

time of 12 seconds until the HTTP connection is terminated. Because this time is too short for

certain applications providing dynamic resources, the number of retries has been increased to 9

for a total time of up to 30 seconds until the connection is terminated.

12595 Crash on DNS Lookup of SIP Trusted Host Names

In the SIP interface it is possible to configure a list of trusted hosts. Incoming calls which are not

originating from these trusted hosts are not accepted on the corresponding interface. Because a

trusted host can be specified as host name it needs to be resolved into an IP-Address before

the trust of an incoming SIP call can be verified. Up until now the resolution of a trusted host

name happened after the reception of the INVITE request. Because the resolution with the DNS

system itself can take some time, this could lead to delays of calls or crashing the device.

Rev. 1.00 07-03-18 5/31

Therefore trusted host names configured in SIP interfaces are now resolved with the DNS

system upon configuration and continuously kept up-to date. As a consequence incoming SIP

calls do not need to wait before the trust can be verified.

12600 Presentation-Indicator and Screening-Indicator from diversion header have random

values and cannot be changed

PI and SI from diversion header had random values and they could not be changed using a

mapping table. They could not be used in a routing table either.

Now PI and SI are correctly initialized and properly set in the diversion header. We can also

create mapping tables to change their values.

12603 Upgraded DSP firmware for SN494x/5x/6x/7x/8x/9x, SN548x/9x and SN466x/7x devices

The DSP firmware has been upgraded to version 700.23.

12605 Web GUI - Spelling errors in the SIP Interface 'call-transfer' section

Corrected spelling errors in „Telephony > Call-Router > Interface > SIP > (choose any SIP

interface) > Configuration > Call-Transfer‟:

1. "Accept" with description „Accepts REFER messagss from the connected user agent‟,

should be "messages" instead of "messagss".

2. “Pull-In” with description „Detects external call loops and connects intern through‟, should

be “internally” instead of “intern”.

12615 Wrong Codec negotiated for SIP to SIP Call

When receiving an INVITE with twice the same codec type inside the SDP, the intersection

logic let slip a wrong codec from the local configured voip profile into the resulting list, breaking

codec negotiation for SIP to SIP calls.


12580 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.

12590 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.

Rev. 1.00 07-03-18 6/31

Now a warning message is printed into the event log if this case arises.

12596 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 2017-03-14 must be loaded for proper operation of

this device.

12597 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 running-

config either.

12598 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-to-

back 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.

12599 EFM interface drops all packets with certain non-FCS-capable DSLAMs

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.

12601 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.

Rev. 1.00 07-03-18 7/31


11215 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.

12435 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.

12573 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

timeout” takes precedence and terminates the played dial tone when expired (12 seconds by

default).

12574 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.

12576 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.

12578 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 hunt-

Rev. 1.00 07-03-18 8/31

group 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.

12579 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.

12586 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.


12540 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.

12551 RTP payload-type translations for DTMF RFC2833 events

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.

12560 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.38.

2. 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.

12567 SIP UPDATE mid-dialog fails

http://birs.inalp.intranet/trac/smartware/wiki/SmartWare

Rev. 1.00 07-03-18 9/31

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.


12526 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).

If the call is SIP-SIP and the UPDATE message does NOT include SDP -> 200 OK, and

will update only privacy headers.

12549 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.

12550 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.

12553 PPP LCP magic number not random

Rev. 1.00 07-03-18 10/31

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.

12554 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. 1492 bytes). One reason for this is when a PPPoE connection is

terminated by two L2TP-chained BRAS (Cisco and Juniper, see http://www.gossamer-

threads.com/lists/cisco/nsp/71142); 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.

12555 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).


12514 Speed test causes a crash


12517 Incorrect codec order in SDP negotiation in special use-case

12520 Broadsoft Flash-Hook signaling in SIP was ignored

12528 SmartWare does not handle/answer SIP UPDATE requests with SDP content


12501 Introduce ATM encapsulation for G.SHDSL.bis-EFM interface

12505 SIP Allow header field missing in INVITE and 200 OK

12506 Provisioning profile added to factory config for several products

12516 Low-bitrate-codec configuration missing on SN-DTA and SN4120

Rev. 1.00 07-03-18 11/31

12527 No music on hold


12482 SNMP Traps for FXO Port Line Up

12486 Reload graceful fixed and enhanced: introducing graceful timeout

12498 Preferred transport protocol for penalty-box SIP options is now configurable


12438 Support for SIP Reason header

12467 SIP overlap reception lost dialed digits

12474 FXO call dropped on early flash-hook signal

12481 H.323 calling-name information not passed to outgoing call

12484 Show hardware information of G.SHDSL submodule

12487 SmartNode crash with “debug sip registration”

12490 Wrong source port in SIP REGISTER

12491 SIP re-register after contact expired


12464 Support for SN4112S


12454 Commands on EFM interface did not provide any help text

12462 Fixed back-to-back codec negotiation when codec preference order was altered by peer

12463 Losing memory when using PRACK with SIP

12469 SIP stack upgrade to V4.2.14


12425 Fax detection during non-transcoded SIP-to-SIP calls

12426 Better handling of failing EFM daughterboard upgrade

12444 Control SDP announcement in provisional responses

12448 Digest Authorization scheme is now case-insensitive

Rev. 1.00 07-03-18 12/31


12410 Add port option to spoofed commands in SIP gateway

12417 Send INVITEs to registrar

12431 SNR and attenuation values for devices with ADSL interface


11932 Support for CAMA emergency E911 protocol

12384 Tunneling of ISDN UUI1 information over SIP

12411 Support for R2 to R2 call scenario

12414 Addition of MWI capability on FXS with battery-reversal


12110 ISDN calls are now visible through SNMP

12403 Configuration of G.SHDSL card is correctly applied during first boot

12405 Firmware upgrade through HTTP fixed


12366 Extended REFER timer for call-diversion

12390 Accept 406 status code as fax failover trigger

12398 PPPoE and VLAN configuration fixed for EFM SmartNodes

12401 Early-media fix

12402 SIP endpoints leak


12354 HTTP/HTML provisioning fixed

12368 Media detection timeout caused loss of voice

12369 Correct upgrade status displayed when upgrading the EFM card via CLI

12379 New FXS profile supporting current of 30 mA

12387 SNTP servers updated in factory-config

12391 EFM daughter card upgrade/downgrade support for new card firmware

Rev. 1.00 07-03-18 13/31

12392 EFM port status correctly displayed with new card firmware

12393 Corrected EFM port configuration after reboot

12395 Enhanced link configuration command for EFM daughter card

12397 EFM configuration option on the web GUI for SN5490 models


12351 Corrected reception and transmission of SIP AOC headers

12271 Rejected INVITE with ambiguous SDP content header

12272 Failed to send SIP BYE message


12188 SIP Trusted Host enhancement

12268 Configurable time zone for CDR messages (accounting data)

12276 Removed no form of address-translation incoming-call commands

12297 Calling-Name support for DMS-100

12301 SIP AOC-E (advice of charge) not sent

12309 DTMF tones propagation on the SIP receive side

12326 Changed verbiage in GUI for reloads

12333 More tolerance for SDP parsing of received messages

12344 calling-redir-e164 cannot be configured in web GUI

12345 Web GUI display issue: AOC SIP-header

12353 Support for upgrade to Trinity with multiple shipping-configs

12358 E1T1 channel-group encapsulation voiceband not available


12285 EFM card firmware version in show command

12290 Show command formatting for EFM card

12312 Crash in Advice of Charge scenarios when using SIP headers

12320 Random crash during calls

12330 Identical SIP-Header X-Org-ConnID for two simultaneous calls

Rev. 1.00 07-03-18 14/31

12341 X-Org-ConnID with call diversion


12243 SIP B2BUA dynamic registration support

12287 Crash on terminating SIP call with DNS

12302 Parameters not applied to EFM daughter card during startup

12307 Local tax pulse generation on FXS

12313 Inbound call to FXO, ringing not always dispatched to the call-control

12318 SIP NOTIFY check-sync

12319 Screening and presentation indicators treated incorrectly for diverted calls

12322 Wrong help text for the annex-type parameter of EFM daughter card


12220 SIP stack upgrade to V4.2.8

12239 Fixed crash when receiving 302 moved temporarily SIP message

12240 SIP registration expiry time is wrong

12255 Radius authentication profile for SSH is not saved in the configuration

12261 Traffic-class not set on stack-generated SIP messages

12270 FXS port stops ringing the phone

12272 SIP AoC not sent when overlap dialing is enabled

12277 SN4660/SN4670 - HW QUEUE FULL error while receiving DTMF tone

12282 Auto-provisioning fails when DHCP option 66 is missing

12298 EFM, communication broken with the card


10977 Support for call deflection on ISDN

11990 Incorrectly encoded Calling-Name for NI-2 and DMS-100

12028 SIP AOC Header support

12205 New SIP Header “X-USE302: YES”

12206 Unique SIP connection ID for calls

Rev. 1.00 07-03-18 15/31

12209 Invalid BGP identifier

12214 Configurable calling party or facility IE on ISDN

12215 Broken policy-routing for SIP calls over UDP

12216 Determine reachability with SIP OPTION requests

12221 SIP penalty box behavior improved

12222 Missing identity header for empty calling party number

12224 Identity headers not parsed for SIP overlap dialing calls

12230 SIP-Gateway wrong address lookup

12234 G.SHDSL link UP with cell delineation error or with training error

12241 Missing shipping-config after upgrade to Trinity

12244 SDP ptime attribute

12245 Rejected INVITE when Call-ID contains a „<‟ character

12247 SIP request URI length limitation

12252 Local RAS port is configurable for H.323

12253 H.323 Gatekeeper fallback not working

12256 License installation fails on Trinity after upgrade from SmartWare

12263 Wrong MOS value for G.723

12284 Compatibility with EFM DB V3.3.1


12016 PSTN configuration on a R2 interface was cleared after a few calls

12145 ISDN interface is capable of triggering actions

12167 Alcatel signaling method for flash-hook SIP info message

12169 SIP request not being sent

12183 Downloadable CDR records

12186 SNMP allowed network

12190 Display error of ISDN binding

12195 Adaptions to maximal possible SIP sessions on ESBR

Rev. 1.00 07-03-18 16/31

12196 Availability of H.323 in ESBR products

12199 Support for EFM daughter card (Rev A)

12204 Duplicate T.38 attributes in SDP

12211 Logging error when a WAN card is detected but unknown

12212 Crash in SIP transfer scenarios


11683 Network and user provided secondary calling party number

11925 Encryption key provisioning

11961 PRACK not working for forked INVITE

11973 Incoming calls refused with 481 after PPP cycle

12114 Unknown SAPI message on E1 port

12136 Crash during startup with large configurations

(also fixes 12113, 12085, 12132)

12153 SIP overlap dialing causing unexpected reboot

12163 Reset log shows „HW watchdog‟ as „Power off/Man reset‟

12170 Changing SSRC causes one-way voice connection

12178 SIP register not working in combination with loopback interface

12182 ASN1 AOC not working

12184 Crash when receiving a SIP answer without Via header

12191 AAA framework problem


11214 Ethernet speed capability for manual settings

11944 Action script trigger for SIP registration

11984 G.SHDSL interface software upgrade failed

12079 RTP payload type conflict

12115 G.SHDSL mode auto detection fixed

12120 Support for short delay re-INVITE in SBC scenario

12125 SN4991 Models with ADSL interface supported

Rev. 1.00 07-03-18 17/31

12128 RTP through VPN broken

12130 Wrong drop cause reported by SIP endpoints, resulting in failed call hunting

12131 Spelling error corrected on BGP configuration web page

12135 SN4660/70 cooling fan speed adjusted


11863 SIP supports TCP flows according to RFC5626

11906 SNMP OID for DSL card firmware version

11916 Support for SN4832/LLA and SN4834/LLA models

11943 New NTP server in factory-config

11958 SIP AOC XML support

11959 Sending tax-pulses on FXS for AOC

12039 Incorrect answer to SIP INFO message

12062 G.SHDSL software upgrade progress indication updated

12074 Increased timeout for redirection service

12082 183 Session Progress not being forwarded in SBC scenarios

12083 Minimal SIP registration time

12086 Added support for SFP interface (Fiber interface)

12093 SIP calls over TCP failed

12097 DSL supervisor log notifies wrong DSL line state

12111 Missing user part from SIP contact header


11717 Invalid REGISTER request when spoofed contact is set

11846 SIP multipart message support

11854 Support for 4300/JO and 4400/JO products

11859 Media detection timeout

11912 Enhanced AAA debug logs

11999 New Patton corporate style applied to web interface

Rev. 1.00 07-03-18 18/31

12014 Wrong help text for blink command

12036 Limit packets to prevent SIP overload condition

12040 SIP register back-to-back command removed

12043 Added option DHCP.66 error message when not available

12045 Concurrent Dynamic IP Configuration (DIC) removes default gateway

12055 Flash hook on FXO interface broken


11497 Configuration option for caller-ID checksum verification on FXO interface

11728 G.SHDSL interface: service-mode auto-detection

11811 ISDN status errors on Web UI

11835 MWI on FXS not working

11860 SIP re-register not working

11879 ADSL annex A/B/M

11888 Improved dial „on-caller-id‟ on FXO

11908 Layer 2 COS for PPP and PPPoE control packets

11935 Administrator login to administrator exec mode

11937 MWI Subscription failing

11940 H.323 Call Resuming

11955 Dial tone played a second time

11981 G.S line rate negotiation fails at high distance

11989 FXO dial-tone detection

12019 Invalid SDP offer in SIP provisional response


10983 DTMF caller ID transmission on FXS

11716 Crash when a „#‟ character is present in SIP contact header

11785 Support for p-called-party-id header

11872 Incoming SIP calls refused with 481 after an IP address change over PPP

Rev. 1.00 07-03-18 19/31

11882 Cooling fan always running at full speed on SN4670

11889 No final answer when receiving BYE

11891 SN4660/SN4670 Rev C and Rev D support

11895 Ethernet switch problem on SN4660/SN4670

11898 Enhancement of software upgrade procedure

11899 SIP Hold/Unhold behavior

11902 SIP q-value of 1.0

11929 SN-DTA clock synchronization

11932 SIP 503 error handling

11936 Broken T.38 transmission

11942 Redirection service for provisioning supported in factory configuration

11950 Modified memory layout for SDTA, SN4552, SN4554 and SN5200

11954 FXS hanging calls

11957 Basic PRACK scenario broken

11965 Spurious error messages from G.S interface

11968 Missing command „payload-rate‟ on SN4660/SN4670

11983 Verbose software upgrade of G.S interface card

11987 Removed support for hardware version 4.x for SN4552, SN4562, SN4554, S-DTA


11560 Web interface generates a new identity

11638 PPPoA on G.S interface

11722 SN-Web page refresh causing reboot

11766 Enhance DSL status display

11773 DTMF Caller-ID reception on FXO

11778 Call transfer issue fixed

11780 CED-Tone Net Side Detection enhancement

11802 Trusted SIP hosts to improve security

Rev. 1.00 07-03-18 20/31

11803 HTTP download failure blocks the SmartNode

11804 T.38 Fax transmission killed by CNG tone

11810 Auto-provisioning: redirection target reordering

11814 Missing strict-tei-procedure command

11817 Provisioning: prevent downloading incompatible configuration

11819 New provisioning placeholders

11818 Auto-provisioning factory-config (Redirection service support)

11832 Wrong G.S port state displayed

11842 SN5200 hardware-version 6.X support

11844 CED-Tone Net Side Detection not working

11848 Crash when downloading G.S firmware with web interface

11850 Abnormal call termination due to misinterpreted SDP data

11852 Auto-provisioning: Target configuration without leading http or tftp

11855 Ethernet lockout on SN4660/70


11256 Echo Cancellation with RBS

11409 ETSI Caller-ID not detected on FXO

11534 Q-value support for SIP REGISTER

11636 Music on Hold not played to SIP side

11650 Removed DSL options „b-anfp‟ and „a-b-anfp‟

11664 Support for SN4660/SN4670 hardware Revision B

11696 Broken modem transmission using H.323

11708 Removed SIP Contact header verification in 200 OK messages

11709 First received IPCP frame dropped in during PPP connection establishment

11755 Added SDP attributes „X-fax‟ and „X-modem‟ support

11776 Forced Fax/Modem bypass

11787 Remote Early-H.245 initialization

Rev. 1.00 07-03-18 21/31

11791 Wrong mapping table in R2

11806 SN-DTA and SN4120 allow usage of g729 codec


11434 Fixed T.38 packets traffic-class

11632 BRI Daughter-Board

11637 HTTP User Agent enhancement

11641 G.SHDSL power and reset spikes

11655 Syslog-client “no remote …” command crash

11674 Fixed display of mtu and mru max values in running-config

11675 Improved clocking precision for SN-DTA and SN4120

11684 Timestamp enhancement for milliseconds

11685 Enhanced spoofed contact to accept hostname

11697 Fax T.38 not working with H.323

11698 Wrong facility callrerouting packet in case of CFU

11726 Missing facility from running-config

11727 New DSL supervisor mode “observe”


11133 Locking DNS records for SIP requests

11487 Improved configuration and display of bit rate for 4-wire G.S interface

11592 HTTP 302 Redirection now supported for provisioning

11594 Additional parameters in G.SHDSL status: SNR, Loop Attenuation, Port States

11604 Clock synchronization improvements

11615 Fixed “police” traffic class configuration option

11622 RTP payload type configuration

11639 Received maddr parameter is reflected in contact header

11649 Proper differentiation between SN4660 and SN4670 product types

11653 Spurious errors reported by SIP and SDP protocols

Rev. 1.00 07-03-18 22/31

11654 BRI CRC Failures

11656 Potential memory leak in SIP state machine

11676 Support for SN-DTA and SN4120 series

11676 Global power-feed for BRI

11687 Performance improvements

11700 Wrong factory-config for SN products with DSL

Rev. 1.00 07-03-18 23/31

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 2.2.0 (1999) does not correctly handle file sizes of n * 512 Bytes.

Use version 3.0.9 (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. 1.00 07-03-18 24/31

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 4960.

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. 1.00 07-03-18 25/31

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. 1.00 07-03-18 26/31

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 www.patton.com.

https://www.patton.com/manuals/SmartWare_SCG_r69.pdf

Current Revision:

Part Number: 07MSWR69_SCG, Rev. A

Penalty box triggers action script on target reachability

First appeared in build series: 2017-09-11

Action scripts can be configured to trigger an event when the SIP OPTIONS target is not reachable

anymore.

1. First configure the penalty-box

Mode: context cs <cs-name>

Command Purpose

Step 1 node(ctx-cs)[<cs-name>]# interface sip <ifName>

Creates/Enters a SIP interface.

Step 2 node(if-sip)[<cs-name>.<if-name>]# penalty-box sip-option-trigger interval <interval> timeout <timeout>

Enables the penalty-box with SIP OPTIONS.

2. Then configure the Action Script

Mode: actions

Command Purpose

Step 1 node(act)# rule <name> Creates/Enters an actions script rule.

Step 2 node(act)[<name>]# condition cs sip:<if-name> { OPTIONS_TARGET_NOT_REACHABLE | OPTIONS_TARGET_REACHABLE }

Enter the condition in which the following action(s) have to be executed.

Step 3 node(act)[<name>]# action <command> Enter the action to execute when the

condition above is met.

EFM interface drops all packets with certain non-FCS-capable DSLAMs


http://www.patton.com/


Rev. 1.00 07-03-18 27/31

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>

Command Purpose

Step 1 node(prt-dsl)[<slot>/<port>]# [no] fcs 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


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 hunt-

group 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.

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>

Command Purpose

Step 1 node(svc-hunt)[<name>]# call-handover on-media-loss

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>

Rev. 1.00 07-03-18 28/31

Command Purpose

Step 1 node(pf-voip)[<name>]# media loss-detection <ms>

Configures the time in milliseconds since the last received RTP packet until the media connection is considered broken. Default: 1000 ms. Range: 1000-10000 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 3891. 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 X-Patton-Call-Failure: true

o 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>

Command Purpose

Step 1 node(if-sip)[<name>]# [no] call-handover signaling (replaces-header | patton-header)

Configures how the new call provides information about the old call. Default: replaces-header.

Rev. 1.00 07-03-18 29/31

Documentation

Please refer to the following online resources:

Software Configuration Guide SmartWare Release R6.9:


SmartWare Configuration Library:

http://www.patton.com/voip/confignotes.asp

Web Wizard Platform for Configuration generation:

http://www.patton.com/wizard/

SmartNode Utilities:

https://www.patton.com/support/upgrades/index.asp?um=SmartNode%20Utilities

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 192.168.1.1 255.255.255.0

mtu 1500


http://www.patton.com/voip/confignotes.asp

http://www.patton.com/wizard/

https://www.patton.com/support/upgrades/index.asp?um=SmartNode%20Utilities

Rev. 1.00 07-03-18 30/31

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. 1.00 07-03-18 31/31

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 2018-01-12

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; www.ethereal.com)

Your Coordinates:

For further enquiries please add your email address and phone number.

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