3Com® VCX™ V7122 SIP VoIP Gateway Release Notes

3Com® VCX™ V7122 SIP VoIP
Gateway Release Notes
Version 4.4
http://www.3com.com
Part Number 900-0263-01
Published June 2005
1
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3Com VCX V7122 SIP VoIP Gateway Release Notes
CONTENTS
ABOUT THIS GUIDE 7
How to Use This Guide 7
Conventions 7
Related Documentation 8
Documentation Comments 8
CHAPTER 1: WHAT’S NEW IN RELEASE 4.4 9
General Gateway New Features 9
Release 4.4 9
Release 4.4 Beta 11
Routing and Manipulation New Features 12
Release 4.4 12
Release 4.4 Beta 13
SIP New Features 14
Release 4.4 14
Release 4.4 Beta 16
SNMP and Web Server New Features 16
Release 4.4 16
Release 4.4 Beta 17
Miscellaneous New Features 17
Release 4.4 17
Release 4.4 Beta 18
Resolved Constraints (Release 4.4 Beta) 18
New and Modified Parameters 19
Release 4.4 19
Release 4.4 Beta 26
3Com VCX V7122 SIP VoIP Gateway Release Notes
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CHAPTER 2: SIP AND PSTN COMPATIBILITY 37
PSTN to SIP Interworking 37
Supported Interworking Features 37
Unsupported Interworking Features 38
Supported SIP Features 38
Unsupported SIP Features 40
SIP Compliance Tables 41
SIP Functions 41
SIP Methods 41
SIP Headers 42
SDP Headers 44
SIP Responses 44
CHAPTER 3: KNOWN CONSTRAINTS 49
SIP Constraints 49
Gateway Constraints 49
Web Constraints 50
SNMP Constraints 50
CHAPTER 4: 3COM VCX V7122 SIP SUPPLIED SOFTWARE KIT 51
Supplied Software 51
CHAPTER 5: RECENT REVISION HISTORY 53
Revision 4.2 Rev 03 53
General New Features (Version 4.2 Rev 03) 53
SIP New Features (Version 4.2 Rev 03) 53
Resolved Constraints (Version 4.2 Rev 03) 54
New Parameters (Version 4.2 Rev 03) 57
Revision 4.2 60
SIP New Features (Version 4.2) 60
General New Features (Version 4.2) 61
Embedded Web Server New Features (Version 4.2) 64
SNMP New Features (Version 4.2) 65
Resolved Constraints (Version 4.2) 65
New Parameters (Version 4.2) 66
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3Com VCX V7122 SIP VoIP Gateway Release Notes
APPENDIX A: OBTAINING SUPPORT FOR YOUR 3COM PRODUCTS 75
Register Your Product to Gain Service Benefits 75
Solve Problems Online 75
Purchase Extended Warranty and Professional Services 75
Access Software Downloads 76
Contact Us 76
Telephone Technical Support and Repair 76
3Com VCX V7122 SIP VoIP Gateway Release Notes
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3Com VCX V7122 SIP VoIP Gateway Release Notes
ABOUT THIS GUIDE
This document describes the release of the 3Com® VCX V7122 software version 4.4
supporting SIP (Session Initialization Protocol).
Information contained in this document is believed to be accurate and reliable at the time of
printing. However, because of on-going product improvements and revisions, 3Com cannot
guarantee the accuracy of printed material after the Published date nor can it accept
responsibility for errors or omissions. Updates to this document and other documents can be
viewed by registered Technical Support customers. For registration information, see
Appendix A: Obtaining Support for Your 3Com Products.
How to Use This Guide
This book covers these topics:
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Chapter 1: What’s New in Release 4.4
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Chapter 2: SIP and PSTN Compatibility
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Chapter 3: Known Constraints
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Chapter 4: 3Com VCX V7122 SIP Supplied Software Kit
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Chapter 5: Recent Revision History
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Appendix A: Obtaining Support for Your 3Com Products
Conventions
Table 1 lists conventions that are used throughout this guide.
Table 1
Icon
Notice Icons
Notice Type
Description
Information note
Information that describes important features or instructions.
Caution
Information that alerts you to potential loss of data or potential damage
to an application, device, system, or network.
Warning
Information that alerts you to potential personal injury or death.
The symbol 0x indicates hexadecimal notation.
3Com VCX V7122 SIP VoIP Gateway Release Notes
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Related Documentation
The following documents are available on the 3Com Partner Access Web site for the 3Com
VCX V7122 SIP Analog Gateways:
ƒ
3Com VCX V7122 VoIP SIP Gateway User Manual
ƒ
3Com VCX V7122 SIP VoIP Gateway Installation Guide
Documentation Comments
Your suggestions are important to us because we want to make our documentation more
useful to you.
Please send e-mail comments about this guide or any of the VCX V7122 documentation to:
VOICE_TECHCOMM_COMMENTS@3com.com
Please include the following information with your comments:
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Document title
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Document part number (usually found on the front page)
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Page number
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Your name and organization (optional)
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Example:
3Com VCX V7122 SIP VoIP Gateway Release Notes
Part Number 900-0263-01
Page 25
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3Com VCX V7122 SIP VoIP Gateway Release Notes
CHAPTER 1: WHAT’S NEW IN RELEASE 4.4
This chapter covers these topics:
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General Gateway New Features
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Routing and Manipulation New Features
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SIP New Features
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SNMP and Web Server New Features
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Miscellaneous New Features
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New and Modified Parameters
General Gateway New Features
Release 4.4
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Max call duration – Users can now limit the maximum duration of a call. When this time
expires, the call is released (from both sides - IP and Tel).
Relevant parameter: MaxCallDuration.
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Additional fields were added to CDR reports: Call Setup Time, Call Connect Time, Call
Release Time, RTP Delay and Jitter, RTP SRC of local and remote sides, Redirect
number, Redirect TON/NPI and Redirect reason.
Note: The Call Time parameters are included in the CDR only if NTP is used or if the
gateway’s local time and date were configured.
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The following RADIUS enhancements were added:
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An Accounting Start report.
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A unique Session-ID was added to the start and stop accounting messages to
correlate between messages of the same call.
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Additional fields were added to the accounting report: Call Setup Time, Call Connect
Time and Call Release Time.
Relevant parameter: RadiusAccountingType.
3Com VCX V7122 SIP VoIP Gateway Release Notes
9
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MFC R2 Brazil “Clear Back” feature is now supported – When the PBX sends a Suspend
signal to the gateway, the 3Com VCX V7122 starts a Regret Timer and sends a Hold ReInvite message to the IP. If the gateway receives an Unhold message from the PBX, it
sends a Retrieve Re-Invite message to the IP. If the timer expires, a Release message is
sent to the IP. If a Release message is received from the PBX, the gateway releases the
IP call. If Release message is received from the IP, the gateway releases the PBX call.
Relevant parameters: RegretTime, EnableHold. (If EnableHold = 0, the Re-Invite
message isn’t sent.)
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MFC R2 Brazil Hold Timeout is now supported – When the gateway receives a Hold
message from the IP, it starts a timer. If this timer expires before Unhold Re-Invite is
received, the gateway releases the IP call.
Relevant parameter: HeldTimeout.
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NI-2 Calling Name – Interworking of PRI to SIP, and SIP to PRI Calling Name.
The Calling Name can be received via one of these methods:
ƒ
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A Facility IE in the Setup message that includes the Calling Name.
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A Facility IE in the Setup message signals that additional information is following.
After the Setup message, a Facility message is received that includes a Facility IE
with the Calling Name information (applicable only to NTÆTE direction).
If the gateway receives an ISDN Disconnect message with Progress Indicator = 1 or 8
before a Connect message is received, it now sends a 183 message to IP. If PI is not
received in the Disconnect message, the call is released. Thus, a voice channel is
opened to play announcements. The ‘PIForDisconnectMsg’ parameter can be used to
override the PI value that is received in the ISDN Disconnect message.
Relevant parameter: PIForDisconnectMsg.
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Playing Ringback Tone (RBT) to Tel for ISDN calls – The gateway is now able to decide
whether the RBT is played to the subscriber by the gateway itself or by the PBX. This
feature can be used when the PBX is not able to play the RBT by itself.
Relevant parameter: LocalISDNRBToneSource.
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NI-2 second redirect number – Users can now select and use (in Invite messages) the
NI-2 second redirect number, if two redirect numbers were received in Q.931 Setup for
incoming TelÆIP calls.
Relevant parameter: ISDNInCallsBehavior_x = 262144.
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Support for partial retrieval of the Redirect Number (number only) from a Facility IE in the
Setup message was added. Applicable to Redirect number according to ECMA-173 Call
Diversion Supplementary Services.
Relevant parameters: SupportRedirectInFacility, ISDNDuplicateQ931BuffMode.
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CAS NFA transfer – The gateway now supports the CAS NFA DMS-100 protocol,
including blind transfer (using Refer) to remote PBX extension.
3Com VCX V7122 SIP VoIP Gateway Release Notes
Relevant parameter: TrunkTransferMode_X.
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Advice of Charge (AOC) – The gateway now supports reception of ISDN (Euro ISDN)
AOC messages. These messages can be received during a call (Facility messages) or at
the end of a call (Disconnect or Release messages). The gateway converts the AOC
messages into SIP Info (during a call) and Bye (end of a call) messages using a
proprietary AOC SIP header. The gateway supports both Currency and Pulse AOC
messages.
Relevant parameter: EnableAOC.
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Supporting TDM Tunneling - The 3Com VCX V7122 TDM Tunneling feature allows you
to tunnel groups of digital trunk spans or timeslots (B-channels) over the IP network.
TDM Tunneling utilizes the internal routing capabilities of the 3Com VCX V7122 (working
without Gatekeeper control) to receive voice and data streams from TDM (1–16
E1/T1/J1) spans or individual timeslots, convert them into packets and transmit them
automatically over the IP network (using point-to-point or point-to-multipoint gateway
distributions). A 3Com VCX V7122 opposite it (or several 3Com VCX V7122 gateways,
when point-to-multipoint distributions is used) converts the IP packets back into TDM
traffic. Each timeslot can be targeted to any other timeslot within a trunk in the opposite
3Com VCX V7122.
Relevant parameters: EnableTDMoverIP, ProtocolType = 4 or 5 (Transparent),
CASTransportType = 1 (CAS signaling relay using RFC 2833).
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Support for SS7 tunneling was added. The SS7 tunneling feature facilitates peer-to-peer
transport of SS7 links between gateways that support the unique 3Com MTP2 Tunneling
application (M2TN) for transferring SS7 MTP2 link data over IP. In this scenario, both
sides of the link are pure TDM switches and are unaware of the IP tandem that is utilized
between them. Using M2TN, the network operator can support SS7 connections over IP,
carrying MTP level 3, as well as higher level SS7 layers (e.g., user parts and application
protocols, such as TUP, ISUP, SCCP, TCAP, etc.).
For the relevant parameters, refer to the 3Com VCX V7122 VoIP SIP User Manual.
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Support for calling card application was added. The 3Com VCX V7122 calling card
application capability (included in its IVR - Interactive Voice Response - feature) enables
Internet Telephony Service Providers (ITSPs) to provide a VoIP telephone service to
subscribers who have purchased calling cards in advance.
For the relevant parameters, refer to the 3Com VCX V7122 VoIP SIP User Manual.
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If calling party name is not defined (CallerDisplayInfoX = <name> is not specified per
gateway’s B-channel port), the calling number can be used instead. Applicable to TelÆIP
calls.
Relevant parameter: UseSourceNumberAsDisplayName.
Release 4.4 Beta
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Extensive Profiles support was added. Different Profiles can now be assigned on a per
call basis, using the Tel to IP and IP to Tel routing tables, or by assigning different
Profiles to the gateway’s endpoint(s). Each Profile contains parameters such as Coders,
3Com VCX V7122 SIP VoIP Gateway Release Notes
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T.38 relay, Voice and DTMF gains, Silence suppression, Echo Canceler, RTP DiffServ,
current disconnect, reverse polarity and more.
The Profiles feature allows the user to tune these parameters or turn them on or off, per
source or destination routing and/or the specific gateway or its B-channel. For example,
B-channels can be designated for Fax-only by having a profile which always uses G.711.
For more detailed information on the Profiles feature, refer to the 3Com VCX V7122 VoIP
SIP User’s Manual.
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Users can now monitor and record SIP real-time activity such as call details and call
statistics, including the number of call attempts, failed calls, fax calls, etc. The
accumulated data can be viewed in the Embedded Web Server (Status and Diagnostics
menu) and via SNMP.
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An option to configure a separate destination IP address for CDR Syslog reports was
added in order to work smoothly with 3rd party billing servers.
Relevant parameter: CDRSyslogServerIP.
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Users can now configure the gateway to receive T.38 fax relay packets into the same
port used by the RTP packets, instead of the RTP port + 2. This solves compatibility
issues with certain NATs and Firewalls.
Relevant parameter: T38UseRTPPort.
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T.38 Redundancy Enhancement - The redundancy of the low speed data is now
determined according to the enhanced redundancy parameter.
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Optimization of channel parameters when detecting fax or modem signals (applicable
only if the channel was opened with the G.711 coder). When detecting a fax or modem
signal on the terminating or originating sides, the gateway modifies the channel’s settings
to work with voice band data signals such as disable NLP, disable or enable Echo
Canceler (EC is enabled for fax calls and disabled for modem calls), disable silence
suppression and setting optimized Jitter Buffer mode.
Relevant parameter: FaxTransportType = 3 (Transparent with events).
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Alert Timeout (ISDN T2 timer) for outgoing call to PSTN can now be configured.
Relevant parameter: PSTNAlertTimeout.
Note: The PSTN stack T2 timer can be overridden by a lower value, but it can’t be
increased.
Routing and Manipulation New Features
Release 4.4
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IP addresses can now include wildcards – IP addresses in the ‘Source IP Address’
column of the ‘IP to Trunk Group Routing’ table and the ‘Source IP’ column in the
‘Destination Phone Number Manipulation Table for IP to Tel Calls’ can include the “x”
wildcard that represents single digits. For example: 10.8.8.x (10.8.8.0–10.8.8.9),
10.8.8.xx (10.8.8.10–10.8.8.99), 10.8.xx.xxx (10.8.10.100–10.8.99.255).
3Com VCX V7122 SIP VoIP Gateway Release Notes
Relevant parameters: PSTNPrefix, NumberMapIP2Tel.
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A ‘Source IP’ column was added to the Destination Phone Number Manipulation Table
for IP to Tel Calls. This field enables you to manipulate the destination number also
according to the source IP address of the call.
Relevant parameter: NumberMapIP2Tel.
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Supports digit delivery to the IP side. Using the manipulation tables the gateway can now
be configured to play pre-configured DTMF digits (per call), after the call is answered.
Relevant parameter: EnableDigitDelivery2IP.
Release 4.4 Beta
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Internal DNS table was added - Similar to a DNS resolution, translates hostnames into IP
addresses. This table is used when hostname translation is required (e.g., ‘Tel to IP
Routing’ table, ‘Gatekeeper IP Address’, etc.). Two different IP addresses can be
assigned to the same hostname. If the hostname isn’t found in this table, the gateway
communicates with an external DNS server. Up to 10 hostnames can be configured.
Relevant parameter: Dns2IP.
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Enhanced Tel to IP routing selection - Selection of destination IP address and IP Profiles
(optional), can now be performed according to both Destination and Source numbers.
Relevant parameter: Prefix.
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Enhanced IP to Tel routing selection - Selection of trunk groups and IP Profiles (optional)
can now be performed according to Destination number, Source Number and Source IP
address.
Relevant parameter: PSTNPrefix.
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Enhanced Number Manipulation support - In all four manipulation tables, the following
functionalities were added:
ƒ
Can now select an entry according to both destination and source numbers.
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Can now apply the "Digits to add" and "Digits to remove" manipulation rules also on
number suffixes in addition to number prefixes.
Relevant parameters: NumberMapTel2IP, NumberMapIP2Tel,
SourceNumberMapTel2IP, SourceNumberMapIP2Tel.
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An option to configure the Calling Number Presentation (Allowed or Restricted) per Tel to
IP call was added (using the Source Number Manipulation table).
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The Called Number Manipulation table was increased to 50 rows. The Calling Number
Manipulation table was increased to 20 rows.
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Alternative routing for released calls, for both Tel to IP and IP to Tel calls. Users can now
define several call release reasons, to be used for alternative routing. If a new call is
3Com VCX V7122 SIP VoIP Gateway Release Notes
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released as a result of one of these reasons, the gateway tries to find an alternative
routing rule to that call. If such a rule is found, the gateway immediately performs a new
call according to that rule. In the current release, only one alternative rule can be defined.
Note 1: If there is no response from the remote party the call is released “internally” with
a 408 reason. This “internal” reason can be also used to initiate an alternative call. The
timeout for “no response” decision depends on the alternative IP addresses:
ƒ
If the resolution of the called domain name results with two IP addresses, the “no
response” timeout will be according to the number of “Hot-Swap” retransmissions
using the parameter ‘ProxyHotSwapRtx’ (default = 3 retransmissions).
ƒ
Otherwise the “no response” timeout will be according to the usual number of the SIP
retransmissions (7 – default).
Note 2: For Tel to IP calls, this feature is relevant only if the internal Tel to IP routing
table is used to route the calls.
Relevant parameters: AltRouteCauseIP2Tel, AltRouteCauseTel2IP, PSTNPrefix.
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A new Status Only mode was added to the Alternative Routing feature - The new IP
Connectivity screen can be used to display the status of IP address connections, using
Ping and QoS results, without enabling/disabling the routing rules.
Relevant parameter: AltRoutingTel2IPEnable.
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IP DiffServ code can now be configured for SIP signaling protocol in addition to RTP
Diffserv.
Relevant parameter: ControlIPDiffServ.
SIP New Features
Release 4.4
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Locating SIP Proxy servers – The gateway can now use DNS Service Record (SRV)
queries to discover Proxy servers. If the Proxy IP address parameter contains a domain
name without port definition (e.g., ProxyIP = domain.com), an SRV query is performed (if
enabled). The SRV query returns up to four Proxy host names and their weights. The
gateway then performs DNS A-record queries for each Proxy host name (according to
the received weights) to locate up to four Proxy IP addresses. Therefore, if the first SRV
query returns 2 A-records, and the A-record queries return 2 IP addresses each, no more
searches are performed.
If the Proxy IP address parameter contains a domain name with port definition (e.g.,
ProxyIP = domain.com:5080), the gateway performs a regular DNS A-record query.
Note: This mechanism is applicable only if ‘EnableProxyKeepAlive = 1’.
Relevant parameter: EnableProxySRVQuery.
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Connect on Progress Indicator – After receiving a 183 Session Progress, the gateway
now sends a Connect message to the ISDN side. This enables the opening of a voice
channel for receiving announcements from the IP.
3Com VCX V7122 SIP VoIP Gateway Release Notes
Relevant parameter: ConnectOnProgressInd.
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Carrier Identification Code (CIC) feature – An option was added to relay the CIC from IP
to ISDN in Transit IE. The CIC code (4 digits) is received in the Invite Request-URI.
Relevant parameter: EnableCIC.
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Symmetric Response Routing (according to RFC 3581) is now supported. The gateway
adds a ‘rport’ parameter to the Via header field of each SIP message. The first Proxy that
receives this message sets the ‘rport’ value of the response to the actual port from which
the request was received. This method is used, for example, to enable the gateway to
identify its port mapping outside a NAT.
ƒ
Registration:
ƒ
An option was added to configure the gateway’s registration name that is used in
Register messages.
ƒ
A registrar domain name can now be used instead of an IP address.
ƒ
Users can now determine the registration timing (in percentage) of the re-register
timing that is set by the Registrar.
Relevant parameters: RegistrationTimeDivider, GWRegistrationName, RegistrarName.
ƒ
Support for ‘Path Extension Header’ according to RFC 3327 was added. The gateway
adds a “Path” parameter to the Supported header field of Register messages. This field
allows to accumulate the list of Proxy IP addresses between the gateway and the
Registrar. The gateway can also receive the Path header in a response.
ƒ
IP Alert Timeout – Users can now define a timer for the gateway to wait for a 200 OK
response from the called party (IP side). If the timer expires, the call is released.
Relevant parameter: IPAlertTimeout.
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Session Timer – If no response is received from the remote side except for 100 Trying
after the session timer expires, the gateway issues a Cancel message. If the gateway
receives a 408/481 response, it also issues a Cancel message. If the gateway receives a
4xx (except for 408/481) / 5xx / 6xx response, it issues a Bye message.
ƒ
If the gateway receives a SIP Invite message with an RPID header in which the “rpiprivacy” parameter equals “full”, the gateway now removes the Calling Display Name IE
from the PRI Setup message.
ƒ
Users can now use the SDP attribute (“a=sendonly”) to place the remote party on hold, in
addition to the use of the IP address of 0.0.0.0 and the attribute (“a=inactive”).
Relevant parameter: HoldFormat.
ƒ
Asserted Identity – P-asserted or P-preferred headers are now sent in 180 Ringing and
200 OK messages if received in the initial Invite message.
ƒ
RFC 2833 Negotiation – If the remote side doesn’t include the “telephone-event”
parameter in the SDP attributes, the gateway now keeps sending DTMF digits using
transparent mode as part of the voice RTP.
3Com VCX V7122 SIP VoIP Gateway Release Notes
15
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Support for the ‘Transparent’ coder was added. The ‘Transparent’ coder can be used in
TDM tunneling applications to relay the TDM signaling bearers.
Relevant parameters: Coder = Transparent, TransparentPayloadType.
ƒ
If the coder G.729 is used with silence suppression enabled, the gateway now includes
the string “annex b” in the SDP.
Release 4.4 Beta
ƒ
Supports the latest SIP2QSIG IETF draft-ietf-sipping-qsig2sip-04.txt, including
interworking between 180/183 responses with SDP and Q.931 Progress messages.
ƒ
Support for SIP UPDATE method according to RFC 3311 was added (the gateway
doesn’t initiate UPDATE messages but responds to them).
ƒ
Network Asserted Identity (RFC 3325) supporting both P-Asserted and P-Preferred
Identity headers.
Relevant parameters: AssertedIdMode, IsTrustedProxy.
ƒ
Registration retry time can now be configured.
Relevant parameter: RegistrationRetryTime.
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Support for the Privacy header (RFC 3323 and RFC 3325) was added. If Caller ID is
restricted, the INVITE message will include a Privacy header with “id” parameter
(privacy: id). The privacy header is used together with P-asserted or P-preferred
headers.
ƒ
Proxy Domain Name(s) can now be obtained from a DHCP server according to RFC
3361.
ƒ
On-the-fly Registration / Unregistration to Proxy/Registrar using the Embedded Web
Server’s Re-Register button. Users can now unregister and reregister after
authentication parameters (e.g., username, password) were modified.
ƒ
Now supports reception of 180 Ringing messages after 183 messages with SDP are
received. The gateway ignores the early media and generates a local Ringback tone.
ƒ
Can now configure the sip:URI host part in the OPTIONS message to be either the
gateway’s IP address or the "gatewayname" parameter.
Relevant parameter: UseGatewayNameForOPTIONS.
SNMP and Web Server New Features
Release 4.4
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A ‘SIP Channel Status’ screen was added to the Embedded Web Server. This screen
can be accessed via the ‘Channel Status’ screen. It contains SIP static information and
associated calls information of the selected port.
3Com VCX V7122 SIP VoIP Gateway Release Notes
Release 4.4 Beta
ƒ
The gateway’s Web Interface appearance was updated and enhanced.
ƒ
A new Web wizard guides the user through the process of software upgrade – selection
of files and loading them to the gateway. The wizard also enables the user to upgrade
the software and to maintain the existing configuration.
ƒ
Enable private labeling of the Web browser’s title when a graphical logo is used.
ƒ
A radio button was added alerting the user whether to burn or not to burn changes to
flash during reset.
ƒ
Adding the capability to provision the table of authorized SNMP managers.
ƒ
In addition to acBoard MIB, a new set of MIBs for configuration and status is introduced.
The new MIBs are divided by functionality (Media, Control, System).
ƒ
New SNMP MIB for collection and monitoring system performance.
ƒ
Introduction of a carrier-grade alarm system with the following characteristics:
ƒ
Allows an EM to determine which alarms are currently active (active alarm table).
ƒ
Allows an EM to detect lost alarm raise and clear traps.
ƒ
Allows an EM to recover lost alarm raise and clear traps (alarm history table).
Miscellaneous New Features
Release 4.4
ƒ
Support for prerecorded Call Progress Tones was added. Using the TrunkPack
Downloadable Conversion Utility, users can now create a file that contains prerecorded
tones. Each tone is assigned with a tone type. After loading it to the device, the
prerecorded tones are played as regular Call Progress Tones according to the tone
types. No detection is supported for these tones. The prerecorded tones file can be
burned to the non-volatile memory.
Relevant parameter: PrerecordedTonesFileName = ‘filename’.
ƒ
Users can now instruct the gateway to load a new software (cmp) file and / or
configuration files from a preconfigured TFTP server after a Web / SNMP reset.
Therefore, the gateway can now obtain its networking parameters from BootP or DHCP
servers and its software and configuration files from a different (preconfigured) TFTP
server.
The ini file can be loaded according to a specific gateway’s MAC address enabling easy
configuration for different gateways.
Relevant parameters: PrerecordedTonesFileName = ‘filename’.IniFileURL, CmpFileURL.
ƒ
The error message that indicates an invalid ini file configuration now contains the line
number of the invalid parameter in the ini file.
3Com VCX V7122 SIP VoIP Gateway Release Notes
17
Release 4.4 Beta
ƒ
NTP support. The time of day can now be obtained from a standard NTP server.
Relevant parameters: NTPServerIP, NTPServerUTCOffset, NTPUpdateInterval.
ƒ
When NTP is enabled, a timestamp string [hour:minutes:seconds] is added to all Syslog
messages.
ƒ
DHCP client improvements. The DHCP client now supports limited IP leasing time and
performs lease renewal. In addition, the time server and SIP DHCP options are now
supported.
ƒ
Operation in a multiple routers network was improved. The gateway now learns the
network topology by responding to ICMP redirections and caching them as routing rules
(with expiration time).
ƒ
Support was added for loading and retrieving encoded ini files from the gateway instead
of clear text files. Files are encoded / decoded using the TrunkPack Downloadable
Conversion utility.
ƒ
The mechanism for burning configuration files in non-volatile memory was improved. The
new mechanism enables users to maintain their configuration when upgrading the
software version. Users should note the following changes:
ƒ
ƒ
Saving the entire configuration (parameters and files) in non-volatile memory is now
controlled by a single parameter – SaveConfiguration (default = 1).
ƒ
‘BurnCallProgressToneFile’ and ‘BurnCASFile’ parameters are no longer supported.
Sending of in-band and out-of-band DTMF digits (RFC 2833) in parallel is now
supported.
Relevant parameters: If DisableAutoMuteDTMF=1, in-band DTMF transmission is set
according to the DTMFTransportType parameter.
Resolved Constraints (Release 4.4 Beta)
ƒ
The G.729A internal processing mechanism was enhanced to achieve better
performance results on high load situations.
ƒ
Can now handle 401/407 “authentication required” responses for all SIP requests.
ƒ
Passes the called display name to INVITE messages, if it appears in the Refer-To
header in a REFER request.
ƒ
Supports the compact header (x) for Session expires.
ƒ
Session timer is now supported also for T.38 faxes and for Held calls.
ƒ
Enables SIP destination port configuration for the entire UDP range.
ƒ
Static NAT is now supported for local IP calls.
18
3Com VCX V7122 SIP VoIP Gateway Release Notes
ƒ
Reliable sending of DTMF digits using INFO messages. The gateway now waits for
200OK before sending new DTMF digits.
ƒ
‘SIPDestinationPort’, if used, only affects the destination of the INVITE requests, unless
‘IsAlwaysUseProxy=1’, forcing all SIP messages to be sent to this port.
ƒ
Several SNMP managers can now be configured to access the gateway concurrently.
ƒ
DHCP now supports limited IP leasing time. The gateway performs lease renewal and
initiates a new DHCP request when the lease time expires.
ƒ
All request URIs for mid dialog requests issued by the gateway contain all URI
parameters received in contact/record route.
ƒ
Send an immediate NOTIFY (with 100 trying) as a result of a received REFER request.
ƒ
Requests URIs for INVITE request issued as a result of REFER\3xx will contain all URI
parameters and new headers received in the REFER to\contact headers.
New and Modified Parameters
Release 4.4
Most new parameters (described in Table 2) can be configured with the ini file and via the
Embedded Web Server. Note that only those parameters contained within square brackets
are configurable via the Embedded Web Server.
Table 2
Release 4.4 GA ini File [Web Browser] Parameter Name
ini File [Web Interface] Parameter
Name
EnableProxySRVQuery
[Enable Proxy SRV Queries]
Description
Enables the use of DNS Service Record (SRV) queries to discover
Proxy servers.
0 = Disabled (default).
1 = Enabled.
If enabled and the Proxy IP address parameter contains a domain
name without port definition (e.g., ProxyIP = domain.com), an SRV
query is performed. The SRV query returns up to four Proxy host
names and their weights. The gateway then performs DNS A-record
queries for each Proxy host name (according to the received weights)
to locate up to four Proxy IP addresses. Therefore, if the first SRV
query returns 2 A-records, and the A-record queries return 2 IP
addresses each, no more searches are performed.
If the Proxy IP address parameter contains a domain name with port
definition (e.g., ProxyIP = domain.com:5080), the gateway performs a
regular DNS A-record query.
Note: This mechanism is applicable only if
‘EnableProxyKeepAlive = 1’.
3Com VCX V7122 SIP VoIP Gateway Release Notes
19
ini File [Web Interface] Parameter
Name
ProxyIP
[Proxy IP Address]
Description
IP address of the primary Proxy server you are using.
Enter the IP address as FQDN or in dotted format notation (for
example 201.10.8.1).
You can also specify the selected port in the format:
<IP Address>:<port>.
This parameter is applicable only if you select ‘Yes’ in the ‘Is Proxy
Used’ field. If you enable Proxy Redundancy (by setting
EnableProxyKeepAlive=1), the gateway can work with up to three
Proxy servers. If there is no response from the primary Proxy, the
gateway tries to communicate with the redundant Proxies. When a
redundant Proxy is found, the gateway either continues working with
it until the next failure occurs or reverts to the primary Proxy (refer to
the ‘Redundancy Mode’ parameter). If none of the Proxy servers
respond, the gateway goes over the list again.
The gateway also provides real time switching (hotswap mode),
between the primary and redundant proxies (‘IsProxyHotSwap=1’).
This mode supports only two proxies. If the first Proxy doesn’t
respond to Invite message, the same Invite message is immediately
sent to the second Proxy.
Note 1: If ‘EnableProxyKeepAlive=1’, the gateway monitors the
connection with the Proxies by using keep-alive messages
("OPTIONS").
Note 2: To use Proxy Redundancy, you must specify one or more
redundant Proxies using multiple ’ProxyIP= <IP address>’ definitions.
Note 3: When port number is specified, DNS SRV queries aren’t
performed, even if ‘EnableProxySRVQuery’ is set to 1.
ProxyIP
[Redundant Proxy IP Address]
IP addresses of the redundant Proxies you are using.
Enter the IP address as FQDN or in dotted format notation (for
example 192.10.1.255). You can also specify the selected port in the
format: <IP Address>:<port>.
Note 1: This parameter is available only if you select “Yes” in the ‘Is
Proxy Used’ field.
Note 2: When port number is specified, DNS SRV queries aren’t
performed, even if ‘EnableProxySRVQuery’ is set to 1.
ini file note: The IP addresses of the redundant Proxies are defined
by the second and third repetition of the ini file parameter ‘ProxyIP’.
EnableDigitDelivery2IP
[Enable Digit Delivery to IP]
0 = Disabled (default).
1 = Enable digit delivery to IP.
The digit delivery feature enables sending of DTMF digits to the
destination IP address after the TelÆIP call was answered.
To enable this feature, modify the called number to include at least
one ’p’ character.
The gateway uses the digits before the ‘p’ character in the initial Invite
message. After the call was answered the gateway waits for the
required time (# of ‘p’ * 1.5 seconds) and then sends the rest of the
DTMF digits using the method chosen (in-band, out-of-band).
Note: The called number can include several ‘p’ characters (1.5
seconds pause).
For example, the called number can be as follows: pp699, p9p300.
20
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter
Name
Description
MaxCallDuration
[Max Call Duration (sec)]
Defines the maximum call duration in seconds. If this time expires,
both sides of the call are released (IP and Tel).
The default time is 0 seconds (no limitation).
RadiusAccountingType
[RADIUS Accounting Type]
Determines when a RADIUS accounting report is issued.
0 = At the Release of the call only (default).
1 = At the Connect and Release of the call.
2 = At the Setup and Release of the call.
RegretTime
Determines the time period (in seconds) the gateway waits for an
MFC R2 Resume (re-Answer) signal once a Suspend (Clear back)
signal was received from the PBX. If this timer expires, the call is
released.
The valid range is 0 to 255. The default value is 0.
Applicable only for MFC R2 CAS Brazil variant.
HeldTimeout
Determines the time period the gateway can stay on Hold. If a
Resume (un-hold Re-Invite) message is received before the timer
expires, the call is renewed. If this timer expires, the call is released.
-1
= Indefinitely (default).
0 - 2400 = Time to wait in seconds.
Currently applicable only to MFC R2 CAS variants.
HoldFormat
Determines the format of the hold request.
0 = The connection IP address in SDP is 0.0.0.0 (default).
1 = The last attribute of the SDP contains the following “a=sendonly”.
ConnectOnProgressInd
0 = Connect message isn’t sent after 183 Session Progress is
received (default).
1 = Connect message is sent after 183 Session Progress is received.
This feature enables the play of announcements from IP to PSTN
without the need to answer the TelÆIP call. It can be used with PSTN
networks that don’t support the opening of a TDM channel before an
ISDN Connect message is received.
EnableCIC
0 = Do not relay the Carrier Identification Code (CIC) to ISDN
(default).
1 = CIC is relayed to ISDN in Transit IE.
If enabled, the CIC code (received in an Invite Request-URI) is
included in a TNS IE in ISDN Setup message. For example: INVITE
sip:555666;cic=2345@100.2.3.4 sip/2.0.
Note: Currently this feature is supported only in SIPÆISDN direction.
PIForDisconnectMsg
[Send PI in Disconnect Message]
Defines the gateway’s behavior when a Disconnect message is
received from the ISDN before a Connect message was received.
“Not configured” = Sends a 183 message according to the received
PI in the ISDN Disconnect message. If PI = 1 or 8, the gateway sends
a 183 response, enabling the PSTN to play a voice announcement to
the IP side. If there isn’t a PI in the Disconnect message, the call is
released (default).
0 = Do not send a 183 message to IP. The call is released.
1, 8 = Sends 183 message to IP.
3Com VCX V7122 SIP VoIP Gateway Release Notes
21
ini File [Web Interface] Parameter
Name
Description
LocalISDNRBToneSource
[Local ISDN RBT Source]
Determines whether Ringback tone is played to the ISDN by the PBX
/ PSTN or by the gateway.
0 = PBX / PSTN (default).
1 = Gateway.
This parameter is applicable to ISDN protocols. It is used
simultaneously with the parameter ’PlayRBTone2TEL’.
SupportRedirectInFacility
0 = Not Supported (default).
1 = Supports reception of Redirect Number in Facility IE of ISDN
Setup messages.
TrunkTransferMode_X
0 = Not supported (default).
1 = Supports CAS NFA DMS-100 transfer.
Note: A specific NFA CAS table is required.
IniFileURL
Specifies the name of the ini file and the location of the TFTP server
from which the gateway loads the ini and configuration files.
For example:
tftp://192.168.0.1/filename
tftp://192.10.77.13/config<MAC>
Note: The optional string “<MAC>” is replaced with the gateway’s
MAC address.
Therefore, the gateway requests an ini file name that contains its
MAC address. This option enables the load of different configuration
for specific gateways.
CmpFileURL
Specifies the name of the cmp file and the location of the TFTP
server from which the gateway loads a new cmp file and updates
itself. For example: tftp://192.168.0.1/filename
Note 1: When this parameter is set in the ini file, the gateway always
loads the cmp file after it is reset.
Note 2: The version of the loaded cmp file isn’t checked.
TransparentPayloadType
Specifies the payload type that is used when the selected coder is set
to ‘Transparent’.
The valid range is 96-120. The default value is 56.
GWRegistrationName
Defines the user name that is used in From and To headers of
Register messages.
If ‘GWRegistrationName’ isn’t specified (default), the ’Username’
parameter is used instead.
RegistrarName
[Registrar Name]
Registrar Domain Name.
If specified, the name is used as Request-URI in Register messages.
If it isn’t specified (default), the Registrar IP address or Proxy name
or Proxy IP address is used instead.
RegistrationTimeDivider
Defines the re-registration timing (in percentage). The timing is a
percentage of the re-register timing set by the Registration server.
The valid range is 50 to 100. The default value is 50.
For example: If ‘RegistrationTimeDivider = 70’ (%) and Registration
Expires time = 3600, the gateway resends its registration request
after 3600 x 70% = 2520 sec.
IPAlertTimeout
[Tel2IP No Answer Timeout]
Defines the time (in seconds) the gateway waits for a 200 OK
response from the called party (IP side) after sending an Invite
message. If the timer expires, the call is released.
The valid range is 0 to 3600. The default value is 180.
22
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter
Name
Description
MINSE
[Minimum Session-Expires]
Defines the time (in seconds) that is used in the Min-SE header field.
This field defines the minimum time that the user agent supports for
session refresh.
The valid range is 10 to 100000. The default value is 90.
MaxActiveCalls
[Max Number Of Active Calls]
Defines the maximum number of calls that the gateway can have
active at the same time. If the maximum number of calls is reached,
new calls are not established.
The default value is max available channels (no restriction on the
maximum number of calls). The valid range is 0 to 240.
UseGatewayNameForOptions
[Use Gateway Name for OPTIONS]
0 = Use the gateway’s IP address in keep-alive OPTIONS messages
(default).
1 = Use ‘GatewayName’ in keep-alive OPTIONS messages.
The OPTIONS Request-URI host part contains either the gateway’s
IP address or a string defined by the parameter ‘Gatewayname’.
The gateway uses the OPTIONS request as a keep-alive message to
its primary and redundant Proxies.
IsUserPhoneInFrom
[Use “user=phone” in From header]
0 = Doesn’t use ";user=phone" string in From header (default).
1 = ";user=phone" string is part of the From header.
UseSourceNumberAsDisplay
Name
[Use Source Number as Display
Name]
0 = Leave Display Name empty (default).
1 = Set Display Name to Source Number.
Applicable to TelÆIP calls. If enabled and calling party name is not
defined (CallerDisplayInfoX = <name> is not specified per gateway’s
x B-channel), the calling number is used instead.
SIP183Behavior
[Behavior of 183 message]
Defines the ISDN message that is sent when 183 Session Progress
message is received for IPÆTel calls.
0 = Progress message (default).
1 = Alert message.
When set to 1, the gateway sends an Alert message (after the receipt
of a 183 response) instead of an ISDN Progress message.
3Com VCX V7122 SIP VoIP Gateway Release Notes
23
ini File [Web Interface] Parameter
Name
NSEMode
Description
Cisco compatible fax and modem bypass mode
0 = NSE disabled (default)
1 = NSE enabled
Note 1: This feature can be used only if VxxModemTransportType=2
(Bypass)
Note 2: If NSE mode is enabled the SDP contains the following line:
“a=rtpmap:100 X-NSE/8000”
Note 3: To use this feature:
ƒ
The Cisco gateway must include the following definition: "modem
passthrough nse payload-type 100 codec g711alaw".
ƒ
Set the Modem transport type to Bypass mode
(‘VxxModemTransportType = 2’) for all modems.
ƒ
Configure the gateway parameter NSEPayloadType= 100
In NSE bypass mode the gateway starts using G.711 A-Law (default)
or G.711µ-Law, according to the parameter
‘FaxModemBypassCoderType’. The payload type used with these
G.711 coders is a standard one (8 for G.711 A-Law and 0 for G.711
µ-Law). The parameters defining payload type for the “old”
3ComsBypass mode, ‘FaxBypassPayloadType’ and
‘ModemBypassPayloadType’, are not used with NSE Bypass. The
bypass packet interval is selected according to the parameter
‘FaxModemBypassBasicRtpPacketInterval’.
NSEPayloadType
24
NSE payload type for Cisco Bypass compatible mode.
The valid range is 96-127. The default value is 105.
Note: Cisco gateways usually use NSE payload type of 100.
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter
Name
PlayRBTone2Tel
[Play Ringback Tone to TEL]
Description
0 (Don’t play) = The ISDN / CAS gateway doesn’t play a Ringback
Tone (RBT). No PI is sent to the ISDN, unless the parameter
‘Progress Indicator to ISDN’ is configured differently.
1 (Play) = The CAS gateway plays a local RBT to PSTN after receipt
of a 180 ringing response (with or without SDP).
Note: Reception of a 183 response doesn’t cause the CAS gateway
to play an RBT (unless ‘SIP183Behavior = 1’).
The ISDN gateway functions according to the parameter
‘LocalISDNRBToneSource’:
ƒ
If the ISDN gateway receives a 180 ringing response (with or
without SDP) and ‘LocalISDNRBToneSource = 1’, it plays a RBT
and sends an Alert with PI = 8 (unless the parameter ‘Progress
Indicator to ISDN’ is configured differently).
ƒ
If ‘LocalISDNRBToneSource = 0’, the ISDN gateway doesn’t play
an RBT and an Alert message (without PI) is sent to the ISDN. In
this case, the PBX / PSTN should play the RBT to the originating
terminal by itself.
Note: Reception of a 183 response doesn’t cause the ISDN gateway
to play an RBT; the gateway issues a Progress message (unless
‘SIP183Behavior = 1’).
If ‘SIP183Behavior = 1’, the 183 response is treated the same way as
a 180 ringing response.
2 = Play according to “early media” (default).
If a 180 response is received and the voice channel is already open
(due to a previous 183 early media response or due to an SDP in the
current 180 response), the ISDN / CAS gateway doesn’t play the
RBT; PI = 8 is sent in an ISDN Alert message (unless the parameter
‘Progress Indicator to ISDN’ is configured differently).
If a 180 response is received but the “early media” voice channel is
not opened, the CAS gateway plays an RBT to the PSTN; the ISDN
gateway functions according to the parameter
‘LocalISDNRBToneSource’:
ƒ
If ‘LocalISDNRBToneSource = 1’, the ISDN gateway plays an
RBT and sends an ISDN Alert with PI = 8 to the ISDN (unless the
parameter ‘Progress Indicator to ISDN’ is configured differently).
ƒ
If ‘LocalISDNRBToneSource = 0’, the ISDN gateway doesn’t play
an RBT.
No PI is sent in the ISDN Alert message (unless the parameter
‘Progress Indicator to ISDN’ is configured differently). In this
case, the PBX / PSTN should play an RBT tone to the originating
terminal by itself.
Note: Reception of a 183 response results in an ISDN Progress
message (unless ‘SIP183Behavior = 1’).
If ‘SIP183Behavior = 1’ (183 is handled in the same way as a
180+SDP), the gateway sends an Alert message with PI = 8, without
playing an RBT.
3Com VCX V7122 SIP VoIP Gateway Release Notes
25
ini File [Web Interface] Parameter
Name
Description
EnableAOC
0 = Not used (default).
1 = ISDN Advice of Charge (AOC) messages are interworked to SIP.
The gateway supports reception of ISDN (Euro ISDN) AOC
messages. AOC messages can be received during a call (Facility
messages) or at the end of a call (Disconnect or Release messages).
The gateway converts the AOC messages into SIP Info (during a call)
and Bye (end of a call) messages using a proprietary AOC SIP
header. The gateway supports both Currency and Pulse AOC
messages.
EnableTDMoverIP
0 = Disabled (default).
1 = TDM Tunneling is enabled.
When TDM Tunneling is enabled, the originating 3Com VCX V7122
automatically initiates SIP calls from all enabled B-channels
belonging to the E1/T1/J1 spans that are configured with the
‘Transparent’ protocol. The called number of each call is the internal
phone number of the B-channel that the call originates from. The IP
to Trunk Group routing table is used to define the destination IP
address of the terminating 3Com VCX V7122. The terminating 3Com
VCX V7122 gateway automatically answers these calls if its E1/T1
protocol is set to ‘Transparent’ (ProtocolType = 5).
CASTransportType
0 = Disable CAS relay (default).
1 = Enable CAS relay mode using RFC 2833.
The CAS relay mode can be used with the TDM tunneling feature to
enable tunneling over IP for both voice and CAS signaling bearers.
ISDNDMSTimerT310
Overrides the T310 timer for the DMS-100 ISDN variant.
This parameter enables users to increase the 10 seconds timeout
from call Setup until Alert is received up to 30 seconds.
The valid range is 10 to 30. The default value is 10 (seconds).
Note: Applicable only to Nortel DMS and Nortel MERIDIAN PRI
variants (ProtocolType = 14 and 35).
ISDNJapanNTTTimerT3JA
T3_JA timer (in seconds).
This parameter overrides the internal PSTN T3 timeout on the Users
Side (TE side).
If an outgoing call from the 3Com VCX V7122 to an ISDN subscriber
is not answered during this timeout, the call is released.
The valid range is 10 to 180. The default value is 50.
Applicable only to Japan NTT PRI variant (ProtocolType = 16).
PrerecordedTonesFileName
The name (and path) of the file containing the Prerecorded Tones.
Release 4.4 Beta
Most new parameters (described in Table 3) can be configured with the ini file and via the
Embedded Web Server. Note that only those parameters contained within square brackets
are configurable via the Embedded Web Server.
26
3Com VCX V7122 SIP VoIP Gateway Release Notes
Table 3
Release 4.4 Beta ini File [Web Browser] Parameter Name
ini File [Web Interface] Parameter
Name
Description
ControlIPDiffServ
[Signaling DiffServ]
Defines the value of the 'DiffServ' field in the IP header for the
signaling session.
The valid range is 0 to 63. The default value is 0.
RegistrationRetryTime
[Registration Retry Time]
Defines the time period (in seconds) after which a Registration
request is resent if registration fails with 4xx, or there is no response
from the Proxy/Registrar.
The default is 30 seconds. The range is 10 to 3600.
AssertedIdMode
[Asserted Identity Mode]
0 = None (default).
1 = P-asserted.
2 = P-preferred.
The Asserted ID mode defines the header that is used in the
generated INVITE request. The header also depends on the calling
Privacy: allowed or restricted.
The P-asserted (or P-preferred) headers are used if the originating
party has a Caller ID name. The Caller ID name is presented as a
display name in the P-asserted (or P-preferred) headers. P-asserted
(or P-preferred) headers are used together with the Privacy header. If
Caller ID is restricted the “Privacy: id” will be included. Otherwise for
allowed Caller ID the “Privacy: none” will be used. If Caller ID
(received from PSTN) is restricted, the From header is set to
<anonymous@anonymous.invalid>.
IsTrustedProxy
[Is Proxy Trusted]
0 = The SIP Proxy is not Trusted.
1 = SIP Proxy is Trusted (default).
If Proxy is not Trusted, the P-asserted header is not used.
AddTON2RPI
0 = TON/PLAN parameters aren’t included in the RPID header.
1 = TON/PLAN parameters are included in the RPID header (default).
If RPID header is enabled (EnableRPIHeader = 1) and
‘AddTON2RPI=1’, it is possible to configure the calling and called
number type and number plan using the Number Manipulation tables
for TelÆIP calls.
T38UseRTPPort
Defines that the T.38 packets will be received using the same Rx port
as RTP packets.
0 = Use the RTP port +2 to receive T.38 packets (default).
1 = Use the same port as the RTP port to receive T.38 packets.
3Com VCX V7122 SIP VoIP Gateway Release Notes
27
ini File [Web Interface] Parameter
Name
IPProfile_ID
[IP Profile Settings]
Description
IPProfile_<Profile ID> =
<Profile Name>,<Preference>,<Coder Group ID>,
<IsFaxUsed *>,<DJBufMinDelay *>, <DJBufOptFactor *>,
<IPDiffServ *>,<ControlIPDiffServ *>,<EnableSilenceCompression>,
<RTPRedundancyDepth>
Preference = (1-10) The preference option is used to determine the
priority of the Profile. If both IP and Tel profiles apply to the same call,
the coders and other common parameters of the preferred Profile will
be applied to that call. If the Preference of the Tel and IP Profiles is
identical, the Tel Profile parameters will be applied.
For example:
IPProfile_1 = name1,2,1,0,10,13,15,44,1,1
IPProfile_2 = name2,$$,$$,$$,$,$$,$$,$$,$$,1
$$ = Not configured, the default value of the parameter is used.
(*) = Common parameter used in both IP and Tel profiles.
Note 1: The IP ProfileID can be used in the Tel2IP and IP2Tel routing
tables (Prefix and PSTNPrefix parameters).
Note 2: ‘Profile Name’ assigned to a ProfileID, enabling users to
identify it intuitively and easily.
Note 3: This parameter can appear up to 5 times.
TelProfile_ID
[Tel Profile Settings]
TelProfile_<Profile ID> =
<Profile Name>,<Preference>,<Coder Group ID>,
<IsFaxUsed *>,<DJBufMinDelay *>, <DJBufOptFactor *>,
<IPDiffServ *>,<ControlIPDiffServ*>,<DtmfVolume>,<InputGain>,
<VoiceVolume>, <EnableDigitDelivery>, <ECE>
Preference = (1-10) The preference option is used to determine the
priority of the Profile. If both IP and Tel profiles apply to the same call,
the coders and other common parameters of the preferred Profile will
be applied to that call. If the Preference of the Tel and IP Profiles is
identical, the Tel Profile parameters will be applied.
For examples:
TelProfile_1 = FaxProfile,1,2,0,10,5,22,33,2,22,34,1,1
TelProfile_2 = ModemProfile,0,10,13,$$,$$,$$,$$,$$,0,$$,0,1
$$ = Not configured, the default value of the parameter is used.
(*) = Common parameter used in both IP and Tel profiles.
Note 1: The Tel ProfileID can be used in the Trunk Group table
(TrunkGroup_x parameter).
Note 2: ‘Profile Name’ assigned to a ProfileID, enabling users to
identify it intuitively and easily.
Note 3: This parameter can appear up to 5 times.
28
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter
Name
TrunkGroup_x
[Trunk Group Table]
Description
TrunkGroup_x = T/a-b,c,d
x = Trunk group ID (1 to 99).
T = Physical trunk number (0 to 7).
a = Starting B-channel (from 1).
b = Ending B-channel (up to 31).
c = Phone number associated with the first channel (optional).
d = Optional Tel Profile ID (1 to 5).
For example:
TrunkGroup_1 = 0/1-31,1000 (for E1 span).
TrunkGroup_1 = 1/1-31,$$,1.
TrunkGroup_2 = 2/1-24,3000 (for T1 span).
Trunk group is the recommended method to configure the gateway's
B-channels. The parameter ’ChannelList’ (although still supported)
mustn’t be used simultaneously with Trunk Groups.
Note 1: An optional Tel Profile ID (1 to 5) can be applied to each
group of B-channels.
Note 2: Parameters can be skipped by using the sign "$$".
3Com VCX V7122 SIP VoIP Gateway Release Notes
29
ini File [Web Interface] Parameter
Name
CoderName_ID
[Coder Group Settings]
Description
Coder list for Profiles (up to five Coders).
The CoderName_ID parameter (ID from 1 to 4) provides groups of
coders that can be associated with IP or Tel profiles.
You can select the following coders:
g711Alaw64k
– G.711 A-law.
g711Ulaw64k
– G.711 µ-law.
g7231
– G.723 6.3 kbps (default).
g7231r53
– G.723 5.3 kbps.
g726
– G.726 ADPCM 32 kbps (Payload Type = 2).
g729
– G.729A.
NetCoder6_4
– NetCoder 6.4 kbps.
NetCoder7_2
– NetCoder 7.2 kbps.
NetCoder8
– NetCoder 8.0 kbps.
NetCoder8_8
– NetCoder 8.8 kbps.
Transparent
– Transparent coder.
The RTP packetization period (ptime, in msec) depends on the
selected Coder name, and can have the following values:
g711 family
g729
g723 family
G.726 family
NetCoder family
– 10, 20, 30, 40, 50, 60, 80, 100 (default=20).
– 10, 20, 30, 40 (default=20).
– 30, 60, 90, 120, 150 (default = 30).
– 10, 20, 30, 40, 50, 60, 80, 100 (default=20)
– 20, 40, 60, 80, 100 (default=20).
Note 1: If not specified, the ptime gets a default value.
Note 2: Each coder should appear only once.
Note 3: The ptime specifies the maximum packetization time the
Gateway will receive.
Note 4: G.729B is supported if the coder G.729 is selected and
‘EnableSilenceCompression’ is enabled.
ini file note 1: This parameter (CoderName) can appear up to 5
times.
ini file note 2: The coder name is case-sensitive.
ini file note 3: Enter in the format: CoderName,ptime.
For example, the following three coders belong to coder group with
ID=1:
CoderName_1 = g711Alaw64k,20
CoderName_1 = g711Ulaw64k,40
CoderName_1 = g7231,90
DisableAutoDTMFMute
Enables / disables the automatic mute of DTMF digits when out-ofband DTMF transmission is used.
0 = Auto mute is used (default).
1 = No automatic mute of in-band DTMF.
When ‘DisableAutoDTMFMute=1’, the DTMF transport type is set
according to the parameter ‘DTMFTransportType’ and the DTMF
digits aren’t muted if out-of-band DTMF mode is selected
(’IsDTMFUsed =1’). This enables the sending of DTMF digits in-band
(transparent of RFC 2833) in addition to out-of-band DTMF
messages.
Note: Usually this mode is not recommended.
30
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter
Name
DNS2IP
[Internal DNS Table]
Description
Internal DNS table, used to resolve host names to IP addresses. Two
different IP addresses (in dotted format notation) can be assigned to
a hostname.
DNS2IP = <Hostname>, <first IP address>, <second IP address>
Note 1: If the internal DNS table is configured, the gateway first tries
to resolve a domain name using this table. If the domain name isn’t
found, the gateway performs a DNS resolution using an external DNS
server.
Note 2: This parameter can appear up to 10 times.
AltRouteCauseTel2IP
[Reasons for Alternative Routing
Table]
Table of call failure reason values received from the IP side. If a call
is released as a result of one of these reasons, the gateway tries to
find an alternative route to that call in the ‘Tel to IP Routing’ table.
For example:
AltRouteCauseTel2IP = 486 (Busy here).
AltRouteCauseTel2IP = 480 (Temporarily unavailable).
AltRouteCauseTel2IP = 408 (No response).
Note 1: The 408 reason can be used to specify that there was no
response from the remote party to the INVITE request.
Note 2: This parameter can appear up to 5 times.
AltRouteCauseIP2Tel
[Reasons for Alternative Routing
Table]
Table of call failure reason values received from the pstn side (in
Q.931 presentation). If a call is released as a result of one of these
reasons, the gateway tries to find an alternative hunt group to that call
in the ‘IP to Hunt Group Routing’ table.
For example:
AltRouteCauseIP2Tel = 3 (Not route to destination).
AltRouteCauseIP2Tel = 1 (Unallocated number).
AltRouteCauseIP2Tel = 17 (Busy here).
Note 1: This parameter can appear up to 5 times.
Note 2: If the 3Com VCX V7122 fails to establish a call to the PSTN
because it has no available channels in a specific trunk group (e.g.,
all of the trunk group’s channels are occupied, or the trunk group’s
spans are disconnected or out of sync), it will use the internal release
cause ‘3’ (no route to destination). This cause can be used in the
‘AltRouteCauseIP2Tel’ table to define routing to an alternative trunk
group.
Prefix
[Tel to IP Routing Table]
Prefix = <Destination Phone Prefix>, <IP Address>,
<Src Phone Prefix>,<IP Profile ID>
Selection of IP address (for Tel To IP calls) is according to destination
and source prefixes.
Note: An optional IP ProfileID (1 to 5) can be applied to each routing
rule.
3Com VCX V7122 SIP VoIP Gateway Release Notes
31
ini File [Web Interface] Parameter
Name
PSTNPrefix
[IP to Trunk Group Routing Table]
Description
PSTNPrefix = a,b,c,d,e
a = Destination Number Prefix
b = Trunk group ID (1 to 99)
c = Source Number Prefix
d = Source IP address
e = IP Profile ID (1 to 5)
Selection of trunk groups (for IP to Tel calls) is according to
destination number, source number and source IP address.
Note 1: To support the ‘in call alternative routing’ feature, users can
use two entries that support the same call, but assign them with
different trunk groups. The second entree functions as an alternative
selection if the first rule fails as a result of one of the release reasons
listed in the AltRouteCauseIP2Tel table.
Note 2: An optional IP ProfileID (1 to 5) can be applied to each
routing rule.
Note 3: The Source IP Address can include the “x” wildcard to
represent single digits. For example: 10.8.8.xx represents all IP
addresses between 10.8.8.10 to 10.8.8.99.
NumberMapTel2IP
[Destination Phone Number
Manipulation Table for TelÆIP
calls]
Manipulates the destination number for Tel to IP calls.
NumberMapTel2IP = a,b,c,d,e,f,g
a = Destination number prefix
b = Number of stripped digits from the left, or (if brackets are used)
from the right. A combination of both options is allowed.
c = String to add as prefix, or (if brackets are used) as suffix. A
combination of both options is allowed.
d = Number of remaining digits from the right
e = Number Plan used in RPID header
f = Number Type used in RPID header
g = Source number prefix
The ‘b’ to ‘f’ manipulations rules are applied if the called and calling
numbers match the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and
‘c’.
Parameters can be skipped by using the sign "$$", for example:
NumberMapTel2IP=01,2,972,$$,0,0,$$
NumberMapTel2IP=03,(2),667,$$,0,0,22
32
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter
Name
SourceNumberMapTel2IP
[Source Phone Number
Manipulation Table for TelÆIP
calls]
Description
SourceNumberMapTel2IP = a,b,c,d,e,f,g,h
a = Source number prefix
b = Number of stripped digits from the left, or (if in brackets are used)
from right. A Combination of both options is allowed.
c = String to add as prefix, or (if in brackets are used) as suffix. A
Combination of both options is allowed.
d = Number of remaining digits from the right
e = Number Plan used in RPID header
f = Number Type used in RPID header
g =Destination number prefix
h =Calling number presentation (0 to allow presentation, 1 to restrict
presentation)
The ‘b’ to ‘f’ and ‘h’ manipulation rules are applied if the called and
calling numbers match the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and
‘c’.
Parameters can be skipped by using the sign "$$", for example:
SourceNumberMapTel2IP=01,2,972,$$,0,0,$$,1
SourceNumberMapTel2IP=03,(2),667,$$,0,0,22,0
NumberMapIP2Tel
[Destination Phone Number
Manipulation Table for IPÆTel
calls]
Manipulate the destination number for IP to Tel calls.
NumberMapIP2Tel = a,b,c,d,e,f,g,h,i
a = Destination number prefix
b = Number of stripped digits from the left, or (if brackets are used)
from the right. A combination of both options is allowed.
c = String to add as prefix, or (if brackets are used) as suffix. A
combination of both options is allowed.
d = Number of remaining digits from the right
e = Q.931 Number Plan
f = Q.931 Number Type
g = Source number prefix
h = Not applicable, set to $$
I = Source IP address
The ‘b’ to ‘f’ manipulation rules are applied if the called and calling
numbers match the ‘a’, ‘g’ and ‘i’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and
‘c’.
Parameters can be skipped by using the sign "$$", for example:
NumberMapIP2Tel =01,2,972,$$,0,$$,034
NumberMapIP2Tel =03,(2),667,$$,$$,0,22,$$,10.13.77.8
Note: The Source IP address can include the “x” wildcard to
represent single digits. For example: 10.8.8.xx represents all the
addresses between 10.8.8.10 to 10.8.8.99.
3Com VCX V7122 SIP VoIP Gateway Release Notes
33
ini File [Web Interface] Parameter
Name
SourceNumberMapIP2Tel
[Source Phone Number
Manipulation Table for IPÆTel
calls]
Description
Manipulate the source number for IP to Tel calls.
SourceNumberMapIP2Tel = a,b,c,d,e,f,g,h
a = Source number prefix
b = Number of stripped digits from the left, or (if brackets are used)
from the right. A combination of both options is allowed.
c = String to add as prefix, or (if brackets are used) as suffix. A
combination of both options is allowed.
d = Number of remaining digits from the right
e = Q.931 Number Plan
f = Q.931 Number Type
g = Destination number prefix
h =Calling number presentation (0 to allow presentation, 1 to restrict
presentation)
The ‘b’ to ‘f’ and ‘h’ manipulation rules are applied if the called and
calling numbers match the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and
‘c’.
Parameters can be skipped by using the sign "$$", for example:
SourceNumberMapIP2Tel =01,2,972,$$,0,$$,034
SourceNumberMapIP2Tel =03,(2),667,$$,$$,0,22
CDRSyslogServerIP
[CDR Server IP Address]
Defines the destination IP address for CDR logs.
The default value is a null string that causes the CDR messages to
be sent with all Syslog messages.
AltRoutingTel2IPEnable
[Enable Alt Routing Tel to IP]
Operation modes of the Alternative Routing mechanism:
0 = Disabled (default).
1 = Enabled.
2 = Enabled for status only, not for routing decisions.
NTPServerIP
IP address (in dotted format notation) of the NTP server.
The default IP address is 0.0.0.0 (the internal NTP client is disabled).
NTPServerUTCOffset
Defines the UTC (Universal Time Coordinate) offset (in seconds) from
the NTP server.
The default offset is 0. The offset range is –43200 to 43200 seconds.
NTPUpdateInterval
Defines the time interval (in seconds) the NTP client requests for a
time update.
The default interval is 86400 seconds (24 hours). The range is 0 to
214783647 seconds.
Note: It isn’t recommended to be set beyond one month (2592000
seconds).
SaveConfiguration
Set to 1 to store the configuration files (e.g., Call Progress Tones) in
the non-volatile memory.
Note: The parameters ‘BurnCallProgressToneFile’ and
‘BurnCoeffFile’ are no longer supported.
34
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter
Name
BootPSelectiveEnable
Description
Enables the Selective BootP mechanism.
1 = Enabled.
0 = Disabled (default).
The Selective BootP mechanism enables the gateway’s integral
BootP client to filter unsolicited BootP/DHCP replies (accepts only
BootP replies that contain the text “AUDC" in the vendor specific
information field). This option is useful in environments where
enterprise DHCP server responds to gateway BootP requests.
Note1: When working with DHCP (EnableDHCP=1) the selective
BootP feature must be disabled.
Note 2: The BootPSelectiveEnable is a special "Hidden" parameter.
Once defined and saved in the flash memory, it is used even if it
doesn't appear in the ini file.
PSTNAlertTimeout
Alert Timeout in seconds (ISDN T2 timer) for outgoing calls to PSTN.
The default is 180 seconds. The range is 0 to 240.
Note: The PSTN stack T2 timer can be overridden by a lower value,
but it can’t be increased.
SNMP Parameters
SNMPTrustedMGR_x
Up to five IP addresses of remote trusted SNMP managers from
which the SNMP agent accepts and processes get and set requests.
Note 1: If no values are assigned to these parameters any manager
can access the device.
Note 2: Trusted managers can work with all community strings.
SNMPReadOnlyCommunityStrin
g_x
Read-only community string (up to 19 chars).
The default string is “public”.
SNMPReadWriteCommunityStrin
g_x
Read-write community string (up to 19 chars).
The default string is “private”.
SNMPTrapCommunityString_x
Community string used in traps (up to 19 chars).
The default string is “trapuser”.
3Com VCX V7122 SIP VoIP Gateway Release Notes
35
36
3Com VCX V7122 SIP VoIP Gateway Release Notes
CHAPTER 2: SIP AND PSTN
COMPATIBILITY
PSTN to SIP Interworking
The 3Com VCX V7122/SIP Gateway supports various ISDN PRI protocols such as
EuroISDN, North American NI2, Lucent 5ESS, Nortel DMS100, Meridian1 DMS100, Japan
J1, as well as QSIG (basic call). PRI support includes User Termination or Network
Termination side. ISDN-PRI protocols can be defined on an E1/T1 basis (i.e., different
variants of PRI are allowed on different E1/T1 spans).
In addition, it supports numerous variants of CAS protocols for E1 and T1 spans, including
MFCR2, E&M wink start, E&M immediate start, E&M delay dial/start, loop-start, and ground
start. CAS protocols can be defined on an E1/T1 basis (i.e., different variants of CAS are
allowed on different E1/T1 spans).
The 3Com VCX V7122 simultaneously supports different variants of CAS and PRI protocols
on different E1/T1 spans (no more than four simultaneous PRI variants).
PSTN to SIP and SIP to PSTN Called and Calling numbers can be optionally modified
according to rules that are defined in Gateway’s ini file.
Supported Interworking Features
ƒ
Definition and use of Trunk Groups for routing IPÆPSTN calls.
ƒ
B-channel negotiation for PRI spans.
ƒ
ISDN Non Facility Associated Signaling (NFAS).
ƒ
Supports the latest SIP2QSIG IETF draft-ietf-sipping-qsig2sip-04.txt, including
interworking between 180/183 responses with SDP and Q.931 Progress message.
ƒ
PRI to SIP Interworking of Q.931 Display (Calling name) information element.
ƒ
PRI (NI-2) to SIP interworking of Calling Name using Facility IE in Setup and Facility
messages.
ƒ
Configuration of Numbering Plan and Type for IPÆISDN calls.
ƒ
Interworking of PSTN to SIP release causes.
ƒ
Interworking of ISDN redirect number to SIP diversion header (according to IETF draftlevy-sip-diversion-05.txt).
3Com VCX V7122 SIP VoIP Gateway Release Notes
37
ƒ
Optional change of redirect number to called number for ISDNÆ IP calls.
ƒ
Interworking of ISDN calling line Presentation & Screening indicators using RPID header
<draft-ietf-sip-privacy-04.txt>.
ƒ
Interworking of Q.931 Called and Calling Number Type and Number Plan values using
the RPID header.
ƒ
Supports ISDN en-block or overlap dialing for incoming TelÆIP calls.
ƒ
Supports routing of IPÆTel calls to predefined trunk groups.
ƒ
Supports a configurable channel select mode per trunk group.
ƒ
Supports various number manipulation rules for IPÆTel and TelÆIP, called and calling
numbers.
ƒ
Option to configure ISDN Transfer Capability (per Gateway).
ƒ
Supports MFC R2 “Clear Back” feature – keeps a Regret Timer when a Suspend
message is received from the PBX and a Hold message is sent to the IP. If the timer
expires, a Release message is sent to the PBX.
Unsupported Interworking Features
ƒ
Q.931 and QSIG supplementary services.
ƒ
Overlap sending (only en-bloc sending is used).
ƒ
QSIG and NI-2 Calling name identification.
ƒ
QSIG and PRI connected line identification.
Supported SIP Features
The 3Com VCX V7122 SIP main features are:
ƒ
Reliable UDP transport, with retransmissions.
ƒ
T.38 real time Fax (using SIP).
Note: If the remote side includes the fax maximum rate parameter in the SDP body of
the Invite message, the gateway returns the same rate in the response SDP.
ƒ
Works with Proxy or without Proxy, using an internal routing table.
ƒ
Fallback to internal routing table if Proxy is not responding.
ƒ
Supports four Proxy servers. If the primary Proxy fails, the 3Com VCX V7122
automatically switches to a redundant Proxy.
ƒ
Supports Proxy discovery using DNS SRV records.
38
3Com VCX V7122 SIP VoIP Gateway Release Notes
ƒ
Proxy or Registrar Registration, such as:
REGISTER sip:servername SIP/2.0
VIA: SIP/2.0/UDP 212.179.22.229;branch=z9hG4bRaC7AU234
From: <sip:GWRegistrationName@sipgatewayname>;tag=1c29347
To: <sip:GWRegistrationName@sipgatewayname>
Call-ID: 10453@212.179.22.229
Seq: 1 REGISTER
Expires: 3600
Contact: sip:GWRegistrationName@212.179.22.229
Content-Length: 0
The "servername" string is defined according to the following rules:
ƒ
The "servername" is equal to "RegistrarName" if configured. The "RegistrarName" can
be any string.
ƒ
Otherwise, the "servername" is equal to "RegistrarIP" (either FQDN or numerical IP
address), if configured.
ƒ
Otherwise the "servername" is equal to "ProxyName" if configured. The "ProxyName"
can be any string.
ƒ
Otherwise the "servername" is equal to "ProxyIP" (either FQDN or numerical IP
address).
The parameter ‘GWRegistrationName’ can be any string. If the parameter is not defined,
the parameter ‘UserName’ is used instead.
The "sipgatewayname" parameter (defined in the ini file or set from the Web browser),
can be any string. Some Proxy servers require that the "sipgatewayname" (in Register
messages) is set equal to the Registrar / Proxy IP address or to the Registrar / Proxy
domain name.
The Register message is sent to the Registrar’s IP address (if configured) or to the
Proxy’s IP address. The message is sent once per gateway. The registration request is
resent according to the parameter ‘RegistrationTimeDivider’. For example, if
‘RegistrationTimeDivider = 70’ (%) and Registration Expires time = 3600, the gateway
resends its registration request after 3600 x 70% = 2520 sec. The default value of
‘RegistrartionTimeDivider’ is 50%.
ƒ
Proxy and Registrar Authentication (handling 401 and 407 responses) using Basic or
Digest methods.
ƒ
Supported methods: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, INFO,
REFER, NOTIFY, PRACK, UPDATE and SUBSCRIBE.
ƒ
Modifying connection parameters in a call (re-INVITE).
ƒ
Working with Redirect server and handling 3xx responses.
ƒ
Early media (supporting 183 Session Progress).
ƒ
PRACK reliable provisional responses <RFC 3262>.
ƒ
Call Hold and Transfer Supplementary services using REFER, Refer-To, Referred-By,
Replaces and NOTIFY messages.
3Com VCX V7122 SIP VoIP Gateway Release Notes
39
ƒ
Session Timer <draft-ietf-sip-session-timer-13.txt>.
ƒ
Network asserted identity and privacy (RFC 3325 and RFC 3323).
ƒ
Can now obtain Proxy Domain Name(s) from a DHCP server according to RFC-3361.
ƒ
RFC 2833 Relay for DTMF Digits, including payload type negotiation.
ƒ
DTMF out of band transfer using:
ƒ
INFO method <draft-choudhuri-sip-info-digit-00.txt>.
ƒ
INFO method, compatible with CIisco gateways.
ƒ
NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt>.
ƒ
SIP URL: sip:”phone number”@IP address (such as 1225556@10.1.2.4, where “122556”
is the phone number of the source or destination) or sip:”phone_number”@”domain
name”, such as 122556@myproxy.com. Note that the SIP URI host name can be
configured differently per called number.
ƒ
Can negotiate coder from a list of given coders.
ƒ
Supported coders:
ƒ
G.711 A-law (10, 20, 30, 40, 50, 60, 80, 100 msec).
ƒ
G.711 µ-law (10, 20, 30, 40, 50, 60, 80, 100 msec).
ƒ
G.723 (5.3, 6.3 kbps, 30, 60, 90, 120, 150 msec).
ƒ
G.729A (8 kbps, 10, 20, 30, 40, 50, 60, 80, 100 msec), G.729B is supported if
Silence Suppression is enabled.
ƒ
G.726 (32 kbps, 10, 20, 30, 40, 50, 60, 80, 100 msec).
ƒ
NetCoder (6.4, 7.2, 8.0 and 8.8 kbps, 20, 40, 60, 80, 100, 120 msec).
ƒ
Supports RFC 3327 – Adding “Path” to Supported header.
ƒ
Supports RFC 3581 – Symmetric Response Routing.
Unsupported SIP Features
The following SIP features are NOT supported:
ƒ
MESSAGE method
ƒ
Preconditions (RFC 3312)
ƒ
SDP - Simple Capability Declaration (RFC 3407)
ƒ
Proxy discovery using NAPTR DNS records
ƒ
Multicast
ƒ
TCP, TLS and SIPs.
40
3Com VCX V7122 SIP VoIP Gateway Release Notes
SIP Compliance Tables
The 3Com VCX V7122/SIP Gateways comply with RFC 3261, as shown in the following
sections.
SIP Functions
Table 4
Supported SIP Functions
Function
Supported
User Agent Client (UAC)
Yes
User Agent Server (UAS)
Yes
Proxy Server
Third-party only (Checked with Ubiquity, Delta3,
Microsoft, 3Com, and Snom Proxies)
Redirect Server
Third-party
Registrar Server
Third -party
SIP Methods
Table 5
Supported SIP Methods
Method
Supported
INVITE
Yes
ACK
Yes
BYE
Yes
CANCEL
Yes
REGISTER
Yes
REFER
Yes
NOTIFY
Yes
INFO
Yes
OPTIONS
Yes
PRACK
Yes
UPDATE
Yes
3Com VCX V7122 SIP VoIP Gateway Release Notes
Comments
Send only
Receive only
41
SIP Headers
Release 4.4 of the 3Com VCX V7122/SIP Gateways supports the following SIP Headers:
Table 6
Supported SIP Headers
Header Field
Supported
Accept
Yes
Accept–Encoding
Yes
Alert-Info
Yes
Allow
Yes
Also
Yes
Asserted-Identity
Yes
Authorization
Yes
Call-ID
Yes
Call-Info
Yes
Contact
Yes
Content-Encoding
Yes
Content-Length
Yes
Content-Type
Yes
Cseq
Yes
Date
Yes
Diversion
Yes
Encryption
No
Expires
Yes
Fax
Yes
From
Yes
Max-Forwards
Yes
Messages-Waiting
Yes
MIN-SE
Yes
Organization
No
Priority
No
42
3Com VCX V7122 SIP VoIP Gateway Release Notes
Header Field
Supported
Proxy- Authenticate
Yes
Proxy- Authorization
Yes
Proxy- Require
Yes
Prack
Yes
Record- Route
Yes
Refer-To
Yes
Referred-By
Yes
Remote-Party-ID
Yes
Replaces
Yes
Require
Yes
Remote-Party-ID
Yes
Response- Key
Yes
Retry- After
Yes
Route
Yes
Rseq
Yes
Session-Expires
Yes
Server
Yes
Subject
Yes
Supported
Yes
Timestamp
Yes
To
Yes
Unsupported
Yes
User- Agent
Yes
Via
Yes
Voicemail
Yes
Warning
Yes
WWW- Authenticate
Yes
3Com VCX V7122 SIP VoIP Gateway Release Notes
43
SDP Headers
Release 4.4 of the 3Com VCX V7122/SIP Gateways supports the following SDP Headers:
Table 7
Supported SDP Headers
SDP Header Element
Supported
v - Protocol version
Yes
o - Owner/ creator and session identifier
Yes
a - Attribute information
Yes
c - Connection information
Yes
d - Digit
Yes
m - Media name and transport address
Yes
s - Session information
Yes
t - Time alive header
Yes
b - Bandwidth header
Yes
u - Uri Description Header
Yes
e - Email Address header
Yes
i - Session Info Header
Yes
p - Phone number header
Yes
y - Year
Yes
SIP Responses
Release 4.4 of the 3Com VCX V7122/SIP Gateways support the following SIP responses:
1xx Response - Information Responses.
2xx Response - Successful Responses.
3xx Response - Redirection Responses.
4xx Response - Request Failure Responses.
5xx Response - Server Failure Responses.
6xx Response - Global Responses.
44
3Com VCX V7122 SIP VoIP Gateway Release Notes
1xx Response – Information Responses
Table 8
Supported 1xx SIP Responses
1xx
Response
Supported
Comments
100
Trying
Yes
The SIP Gateway generates this response upon receiving
of Proceeding message from ISDN or immediately after
placing a call for CAS signaling.
180
Ringing
Yes
The SIP Gateway generates this response for an incoming
INVITE message. On receiving this response, the Gateway
waits for a 200 OK response.
181
Call is being
forwarded
Yes
The SIP Gateway does not generate these responses.
However, the Gateway does receive them. The Gateway
processes these responses the same way that it processes
the 100 Trying response.
182
Queued
Yes
The SIP Gateway generates this response in Call Waiting
service. When SIP Gateway receives 182 response, it
plays a special waiting Ringback tone to TEL side.
183
Session
Progress
Yes
The SIP Gateway generates this response if Early Media
feature is enabled and if the Gateway plays a Ringback
tone to IP
2xx Response – Successful Responses
Table 9
Supported 2xx SIP Responses
2xx
Response
Supported
200
OK
Yes
202
Accepted
Yes
Comments
3xx Response – Redirection Responses
Table 10
Supported 3xx SIP Responses
3xx
Response
Supported
Comments
300
Multiple
Choice
Yes
The Gateway responds with an Ack and resends the
request to first in the contact list, new address.
301
Moved
Permanently
Yes
The Gateway responds with an Ack and resends the
request to new address.
302
Moved
Temporarily
Yes
The SIP Gateway generates this response when call
forward is used, to redirect the call to another destination. If
such response is received, the calling Gateway initiates an
INVITE message to the new destination.
305
Use Proxy
Yes
The Gateway responds with an Ack and resends the
request to new address.
3Com VCX V7122 SIP VoIP Gateway Release Notes
45
3xx
Response
Supported
Comments
380
Alternate
Service
Yes
"
4xx Response – Request Failure Responses
Table 11
Supported 4xx SIP Responses
4xx
Response
Supported
Comments
400
Bad Request
Yes
The Gateway does not generate this response. On
reception of this message, before a 200 OK has been
received, the gateway responds with an ACK and
disconnects the call.
401
Unauthorized
Yes
Authentication support for Basic and Digest. On receiving
this message the GW issues a new request according to
the scheme received on this response
402
Payment
Required
Yes
The Gateway does not generate this response. On
reception of this message, before a 200 OK has been
received, the gateway responds with an ACK and
disconnects the call.
403
Forbidden
Yes
The Gateway does not generate this response. On
reception of this message, before a 200 OK has been
received, the gateway responds with an ACK and
disconnects the call.
404
Not Found
Yes
The SIP Gateway generates this response if it is unable to
locate the callee. On receiving this response, the Gateway
notifies the User with a Reorder Tone.
405
Method Not
Allowed
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
406
Not Acceptable
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
407
Proxy
Authentication
Required
Yes
Authentication support for Basic and Digest. On receiving
this message the GW issues a new request according to
the scheme received on this response.
408
Request
Timeout
Yes
The Gateway generates this response if no-answer timer
was expired. On reception of this message, before a
200OK has been received, the gateway responds with an
ACK and disconnects the call.
409
Conflict
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
46
3Com VCX V7122 SIP VoIP Gateway Release Notes
4xx
Response
Supported
Comments
410
Gone
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
411
Length
Required
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
413
Request Entity
Too Large
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
414
Request-URL
Too Long
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
415
Unsupported
Media
Yes
If the Gateway receives a 415 Unsupported Media
response, it notifies the User with a Reorder Tone.
The gateway generates this response in case of SDP
mismatch.
420
Bad Extension
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
480
Temporarily
Unavailable
Yes
If the Gateway receives a 480 Temporarily Unavailable
response, it notifies the User with a Reorder Tone.
This response is issued if there is no response from
remote.
481
Call
Leg/Transaction
Does Not Exist
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
482
Loop Detected
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
483
Too Many Hops
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
484
Address
Incomplete
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
3Com VCX V7122 SIP VoIP Gateway Release Notes
47
4xx
Response
Supported
Comments
485
Ambiguous
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
486
Busy Here
Yes
The SIP Gateway generates this response if the called
party is off hook and the call cannot be presented as a call
waiting call. On receiving this response, the Gateway
notifies the User and generates a busy tone.
487
Request
Canceled
Yes
This response indicates that the initial request is
terminated with a BYE or CANCEL request.
488
Not Acceptable
Yes
The Gateway does not generate this response. On
reception of this message, before a 200OK has been
received, the gateway responds with an ACK and
disconnects the call.
5xx Response – Server Failure Responses
Table 12
Supported 5xx SIP Responses
5xx Response
500
Internal Server Error
501
Not Implemented
502
Bad Gateway
503
Service Unavailable
504
Gateway Timeout
505
Version Not Supported
Comments
On reception of any of these Responses, the GW
releases the call, sending appropriate release cause
to PSTN side.
The GW generates 5xx response according to
PSTN release cause coming from PSTN.
6xx Response – Global Responses
Table 13
Supported 6xx SIP Responses
6XX Response
600
Busy Everywhere
603
Decline
604
Does Not Exist Anywhere
606
Not Acceptable
48
Comments
On reception of any of these Responses, the GW releases
the call, sending appropriate release cause to PSTN side.
3Com VCX V7122 SIP VoIP Gateway Release Notes
CHAPTER 3: KNOWN CONSTRAINTS
SIP Constraints
ƒ
When using out of band DTMF transport (IsDTMFUsed=1), the ‘DTMFTransportType’
parameter should be set to 0 (erase digits from voice stream).
ƒ
If the (first) incoming INVITE message contains both audio and T.38 coders, the gateway
will reply with the first media in SDP and not with an audio coder as was in 4.21 version.
ƒ
Channel parameters, such as, Voice/DTMF gain, silence suppression and Jitter buffer
are collectively configured in the ini file on a per gateway usage (not on a per call basis).
By using Profiles this limitation can be overcome.
ƒ
G.726, 16 kbps, 24 kbps and 40 kbps coders are not supported. Only G.726/32 kbps is
supported.
ƒ
Single ptime parameter is used in SDP message to define the packetization period for
multiple coders. For example, if G.711 and G.723 coders are used, the ptime is set to 30
msec.
ƒ
Only the ptime (packetization time) of the first coder in the defined coder list is declared
in Invite/200 OK SDP, even if multiple coders are defined. Therefore, in the Coders
screen in the Web Interface only the ptime of the first coder in the list is relevant.
Gateway Constraints
ƒ
The VXML-based Calling Card application is not supported in the current version (will be
supported in release 4.4 fix).
ƒ
RFC 2198 redundancy mode with RFC 2833 is not supported (that is, if a complete
DTMF digit was lost, it is not reconstructed). The current RFC 2833 implementation does
support redundancy for inter-digit information lost.
ƒ
Date and Time should be set after each Gateway power reset unless NTP (Network
Time Protocol) is used.
ƒ
Coder names in ini file are case-sensitive.
ƒ
The gateway only supports symmetrical coders – the same coder is used for transmit
and for receive (though different ptime is supported).
ƒ
When using G.711 coder with 10 msec packetization time (without silence suppression)
the 3Com VCX V7122 can support up to 360 channels, each gateway module supporting
up to 180 channels.
ƒ
Usually when using E1 protocols, it is necessary to set the PCMLawselect parameter to
A-law, while when using T1 protocols the PCMLawselect parameter should be set to
3Com VCX V7122 SIP VoIP Gateway Release Notes
49
µ-Law (The parameter can be set from the ini file or via the Web Interface in Trunk
Settings page).
ƒ
It is not valid to configure the board to auto-negotiate mode while the opposite port is set
manually to full-duplex (either 10 Base-T or 100 Base-T). It is also not valid to set the
board to one of the manual modes while the opposite port is configured differently.
ƒ
It is strongly recommended to use 100 Base-T switches. Use of 10 Base-T LAN hubs
should be avoided.
Web Constraints
ƒ
Domain names in the ‘Tel to IP Routing’ table are limited to 15 characters.
ƒ
Not all parameters can be changed on-the-fly from the Web browser. Parameters that
can’t be changed on-the-fly are noted with (!). To change these parameters, reset the
board, using the Web browser reset button.
ƒ
When changing Gateway parameters from the Web browser, the new parameters are
permanently stored in flash memory only after the Gateway is reset from the Web or after
"Save Configuration" button is pressed.
SNMP Constraints
ƒ
The performance monitoring sections in the Trunk MIB are not supported.
ƒ
Configuration alarm does not clear.
ƒ
The following RTP MIB objects are not supported: rtpRcvrSRCSSRC, rtpRcvrSSRC,
rtpSenderSSRC, rtpRcvrLostPackets, rtpRcvrPackets, rtpSenderPackets, rtpRcvrOctets,
rtpSenderOctets.
ƒ
The CAS tables cannot be correctly set via SNMP.
ƒ
The range of the faxModemRelayVolume MIB object is wrong. Instead of 0 to 15, it
should be -18 to -3, corresponding to an actual volume of (-18.5 dBm) to (-3.5 dBm).
ƒ
Only one SNMP manager can access the device simultaneously.
ƒ
Channel status is limited to the number of B-channels (determined by the number of
trunks) and not by the number of logical channels.
50
3Com VCX V7122 SIP VoIP Gateway Release Notes
CHAPTER 4: 3COM VCX V7122 SIP
SUPPLIED SOFTWARE KIT
Table 14 describes the standard supplied software kit for 3Com VCX V7122 SIP Gateways.
The supplied documentation includes this Release Notes and the 3Com VCX V7122 VoIP
SIP User Manual.
Supplied Software
Table 14
3Com VCX V7122 SIP Supplied Software Kit
File Name
Description
Ram.cmp file
Mediant_SIP_xxx.cmp
Image file containing the software for the 3Com VCX V7122 Gateway.
ini files and utilities
Mediant_T1.ini
Sample ini file for 3Com VCX V7122 E1 gateways.
Mediant_E1.ini
Sample ini file for 3Com VCX V7122 T1 gateways.
Usa_tones_xx.dat
Default downloadable Call Progress Tones dat file.
Usa_tones_xx.ini
Call progress Tones ini file (used to create dat file).
bootp.exe
BootP/TFTP configuration utility
DConvert240.exe
TrunkPack Downloadable Conversion Utility
ACSyslog08.exe
Syslog server.
CAS Protocol Files
Used for various signaling types, such as E_M_WinkTable.dat.
MIBs
MIB library for SNMP browser
3Com VCX V7122 SIP VoIP Gateway Release Notes
51
52
3Com VCX V7122 SIP VoIP Gateway Release Notes
CHAPTER 5: RECENT REVISION HISTORY
Revision 4.2 Rev 03
General New Features (Version 4.2 Rev 03)
ƒ
The Embedded Web Server’s username and password can now be changed on-the-fly.
A warning message is displayed if the entered password exceeds 7 characters.
ƒ
Network parameters (IP address, Default Gateway and Subnet) entered from the Web
are now checked for validity. A warning message is displayed if parameter’s value is
incorrect.
ƒ
An option to enable / disable call release when an RTP broken connection is detected,
using the parameter ‘DisconnectOnBrokenConnection’.
ƒ
An option to manipulate source number according to destination number prefix.
ƒ
T.38 fax relay SIP session can now be initiated also when a CED answering tone is
detected (using the parameter ‘DetFaxOnAnswerTone =1’). Note that this operational
mode is not recommended. It is only necessary for specific originating fax machines that
require the reception of a CED tone.
ƒ
The Gateway now supports two stage dialing option for IPÆTel calls: placing a call and
then sending DTMF digits, using the parameter ‘EnableDigitDelivery = 1’.
ƒ
ISDN Transfer Capability for IPÆPSTN calls can now be configured.
ƒ
Can now play dial tone to the ISDN user side in Overlap dialing, if an empty called
number is received, and ‘ISDNINCallsBehavior = 65536 (bit #16) causing the Progress
Indicator to be included in the SETUPACK ISDN message.
ƒ
The Gateway now opens voice if an ISDN DISCONNECT message with PI is received.
ƒ
Supports configuration of ISDN overlap dialing per 3Com VCX V7122 trunk.
ƒ
An optional ISDN IE can now be configured. This IE is used in ISDN SETUP message. It
is also possible to configure the specific E1/T1 span(s) from where this IE is sent.
ƒ
Support for Meridian 1 DMS100 PRI variant was added.
SIP New Features (Version 4.2 Rev 03)
ƒ
T.38 fax now supports the reception of T.38 capabilities in the first INVITE.
ƒ
Supports REINVITE for mid-call change of T.38 fax session parameters.
3Com VCX V7122 SIP VoIP Gateway Release Notes
53
ƒ
A new x-channel header is added. This header provides information on the E1/T1
physical trunk/B-channel on which the call is received or placed. For example: “xchannel: DS/DS1-5/22”. This header is generated by the 3Com VCX V7122 and is sent
in the following messages: INVITE and 183/180/200 OK responses. To enable this
feature set the parameter ‘XChannelHeader’.
ƒ
Diversion header is now supported also for IPÆ ISDN calls. It is used to generate a
redirect number in ISDN SETUP.
ƒ
Supports the ‘maddr’ parameter in ‘refer_to’ URI, and using it in the generated INVITE
SIP:URI.
ƒ
Max-Forwards header was added to the Gateway generated INVITE messages. The
default value is set to 70.
ƒ
Reception of 180 Ringing after 183 response is received, now causes a Ringback tone to
be played to the PSTN. ISDN ALERT (with Progress Indicator) is sent after the reception
of 183 response. If subsequent 180 Ringing message is received and ALERT was
already sent, the 3Com VCX V7122 Gateway plays a Ringback tone.
ƒ
The parameter ‘User=Phone’, can now be included also in the FROM header (in addition
to INVITE URI). Configure ‘IsUserPhoneInFrom = 1’ in the ini file.
Resolved Constraints (Version 4.2 Rev 03)
From Version 4.200 to Version 4.202
ƒ
HTTP download of CAS ini file(s) is fixed.
ƒ
Single page T.38 fax problem is solved (new DSP version).
ƒ
SNMP memory leaks were solved.
ƒ
Asserted Identity is now supported (for IPÆPRI) also if there are no “user=phone”
parameters in the Asserted Identity header.
ƒ
Now supports REFER-To SIP URI, without userpart, for example: “Refer-To:
sip:10.3.1.35”.
ƒ
The Gateway now supports UDP port in REFER-To SIP URI, such as
“sip:123@10.3.1.35:5080”.
ƒ
Can now send PRACK (and other methods) to the IP address that is provided in the
180/183 Record Route header.
ƒ
Supports DTMF INFO messages while the Gateway is in Hold state.
ƒ
Can now send ACK/BYE messages to “maddr IP” in Contact header.
ƒ
Authorization bug was fixed - if “qop = Auth, Auth-INT”, the response contained “Auth,
without the right quotation mark.
ƒ
Remote expire registration time, was updated according to min-expires that is received
from the remote side.
54
3Com VCX V7122 SIP VoIP Gateway Release Notes
ƒ
Mid call authentication (empty username & wrong URI) is fixed.
ƒ
The missing ‘CRLF’ at the end of 200 OK message that is sent in response to REINVITE
message, was added.
ƒ
During call transfer, if the terminating Gateway is in alert state, the Gateway now plays a
Ringback tone.
ƒ
Registration expire timeout bug was fixed.
From Version 4.202 to Version 4.2101
ƒ
The Gateway now releases the allocated Gateway internal sessions in an erroneous call
scenarios.
ƒ
Can now properly handle T2 timer with provisional responses. Reception of 100 trying in
response to non-INVITE SIP requests (such as REGISTER, BYE and others) does not
stop the retransmission.
ƒ
Now sends 481 response if BYE or other SIP messages that are not expected are
received.
ƒ
Now sends 405 “method not allowed” response if MESSAGE method is received.
ƒ
Can now handle 301/302 responses with contact URI: port number (with or without
maddr). The Gateway sends INVITE to the proper IP/port.
ƒ
The Gateway now supports the reception of first INVITE or 200 OK with 0.0.0.0 in SDP
(holds the call from its beginning).
ƒ
When first INVITE with 0.0.0.0 in SDP is received, the received PRACK message (sent
to acknowledge the 180 Ringing) is now acknowledged with 200 OK.
ƒ
The crash that occurred due to the following scenario was fixed: The Gateway receives
an INVITE message with 0.0.0.0, it replies with 200 OK, but no ACK is received. After the
retransmission of 200 OK is finished, the call is not released; the Gateway crashes when
call is released from the Tel side.
ƒ
The missing local port (other than 5060) in Via and Contact headers of REGISTER
message was added.
ƒ
There is now also support for Session timer REINVITE keep alive messages during a call
in Hold state, and for a T.38 fax call as well.
ƒ
Can now send ACK message with the same Branch as was received in 481 response (if
481 was sent as a response to the Gateway’s initiated INVITE).
ƒ
The Gateway now supports up to 10 coders (instead of 5) in received SDP.
ƒ
A bug that increased the value of Cseq by 2, between INVITEs (after authentication) is
fixed.
ƒ
G.723 coder ptimes can now be configured from the Web interface (ranged from 30 to
150 msec).
ƒ
G.723 coder SDP negotiation issue was fixed, for other than 30 msec ptimes.
3Com VCX V7122 SIP VoIP Gateway Release Notes
55
ƒ
CANCEL request is now sent with the same URI as the URI in the INVITE message, if it
is initiated before a 200 OK message is received.
ƒ
The Gateway now supports SIP URI without userpart, in Contact header, for example:
Contact: <sip:192.168.1.34:5060>.
ƒ
The Gateway now uses the ‘maddr’ parameter in an INVITE URI, if it appears in Refer-to,
Record-route or Contact headers.
ƒ
DNS resolution, for REFER-to domains isn’t used, if ‘Send all INVITEs to proxy’ or
‘always use Proxy’ features are enabled.
ƒ
Bug fix: when a call was forwarded on ‘no reply’, a Ringback tone wasn’t played.
ƒ
Double quotes are now added to the names in the FROM headers, even if the names in
the received ISDN DISPLAY header has no quotes.
ƒ
Now delays sending of BYE message after call transfer is complete. This enables the
party that initiated the REFER to also send BYE message.
ƒ
Now supports ‘A’-‘D’ DTMF digits for out of band (INFO) signaling.
ƒ
The bug that filtered all the text between the (< >) signs in the Web’s Message log is
fixed. SIP messages are displayed correctly.
ƒ
Proxy and Gateway names were increased to 50 characters each.
ƒ
The IP address field in the Tel to IP routing table was increased to accept up to 30
characters.
ƒ
The destination prefix field in the IP to Tel manipulation table was increased to accept
strings up to 40 characters.
ƒ
Can now handle two ‘c= IP address’ lines in SDP, including video and audio. Using one
“c=IP address” that is associated with audio.
ƒ
DMS100 ISDN Protocol violation is fixed. Progress indicator is not sent in ISDN
PROCEEDING message (for both TEÆNT and NTÆTE), and not in ALERTING
message (for TEÆNT).
ƒ
NI-2 ISDN protocol violation is fixed. Progress indicator is not sent in ISDN
PROCEEDING message.
ƒ
Bug fix: the “Add Trunk Group as Prefix” and Overlap dialing features now interoperate
correctly.
ƒ
The Calling Number Type / Plan for T1 PRI protocols can now be configured from the ini
file and from the Web. In the previous version it was automatically set according to the
calling number’s length.
ƒ
Redirect number interworking for DMS100, NI-2 and 4ESS/5ESS protocols, is now
supported for both NTÆTE and TEÆNT calls. For Euro ISDN it is supported only for
NTÆTE direction.
56
3Com VCX V7122 SIP VoIP Gateway Release Notes
ƒ
Calling name (Display) interworking for DMS-100 and 4ESS/5ESS protocols is now
supported for both NTÆTE and TEÆNT calls. For Euro ISDN it is supported only for
NTÆTE direction. NI-2 is currently not supported.
ƒ
The following parameters were added to the ‘Protocol Definition’ screen on the Web
interface:
ƒ
ƒ
RFC 2833 Payload Type
ƒ
NTT Caller ID Type
The G.726 unsupported coders were removed from the ‘Protocol Definition’ screen in the
Web interface. Only G.726 / 32 kbps coder is used with SIP.
New Parameters (Version 4.2 Rev 03)
Most new parameters (described in Table 15) can be configured with the ini file and via the
Embedded Web Server. Note that only those parameters contained within square brackets
are configurable via the Embedded Web Server.
Table 15
ini File [Web Browser] Parameter Name
ini File [Web Interface]
Parameter Name
Description
DisconnectOnBrokenConnection
0 = Don’t release the call.
1 = Call is released If RTP packets are not received for a predefined
timeout (default).
Note 1: If enabled, the timeout is set by the parameter
‘BrokenConnectionEventTimeout’, in 100 msec resolution. The default
timeout is 10 seconds: (BrokenConnectionEventTimeout =100).
Note 2: This feature is applicable only if RTP session is used without
Silence Compression. If Silence Compression is enabled, the
Gateway doesn’t detect that the RTP connection is broken.
Note 3: During a call, if the source IP address (from where the RTP
packets were sent) is changed without notifying the Gateway. The
Gateway will filter these RTP packets. To overcome this issue, set
‘DisconnectOnBrokenConnection=0’; the Gateway doesn’t detect RTP
packets arriving from the original source IP address, and will switch
(after 300 msec) to the RTP packets arriving from the new source IP
address.
3Com VCX V7122 SIP VoIP Gateway Release Notes
57
ini File [Web Interface]
Parameter Name
Description
EnableDigitDelivery
The digit delivery feature enables sending of DTMF digits to the
Gateway’s B-channel after the call is answered.
0 = Disabled (default).
1 = Enable Digit Delivery feature for 3Com VCX V7122 (two stage
dialing).
Note: For incoming IPÆTel calls, if the called number includes the
characters ‘w’ or ‘p’, the 3Com VCX V7122 Gateway places a call with
the first part of the called number, and plays DTMF digits after the call
is answered.
If the character ‘p’ (pause) is used, the 3Com VCX V7122 waits for
1.5 seconds before playing the next DTMF digit.
If the character ‘w’ is used, the 3Com VCX V7122 waits for detection
of dial tone before it starts playing DTMF digits. The character ‘w’ can
appear once in the called number, and must precede any ‘p’
character. The ‘p’ character can appear several times.
For example: if the number “1007766p100” is defined as the called
number, the 3Com VCX V7122 places a call with 1007766 as the
destination number, then, after the call is answered, it waits for 1.5
seconds and plays the rest of the number (100) as DTMF digits.
Other examples: 1664wpp102, 66644ppp503, 7774w100pp200.
ScreeningInd2IP
The parameter can overwrite the calling number screening indication
for ISDN TelÆIP calls.
-1 = not configured (interworking from ISDN to IP) or set to 0 for CAS.
0 = user provided, not screened.
1 = user provided, verified and passed.
2 = user provided, verified and failed.
3 = network provided.
SourceMapModeIP2Tel
Source number manipulation option for IPÆTel calls, applicable for
MP-1xx/FXS with Caller ID.
0 = Regular mapping (default).
1 = Source number is changed according to destination number’s
prefix (using source number manipulation table).
SourceMapModeTel2IP
Source number manipulation option for TelÆIP calls.
0 = Regular mapping (default).
1 = Source number is changed according to destination number prefix
(using source number manipulation table).
58
3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface]
Parameter Name
Description
PlayRBTone2Tel
[Play Ringback Tone to TEL]
0 = Don’t play Ringback tone (default).
The Gateway doesn’t play Ringback tone. No PI is sent to ISDN,
unless the parameter ‘Progress Indicator to ISDN’ is configured
differently.
1 = Play. The Gateway plays local Ringback tone to PSTN, after
180/183 response is received. The PRI Gateway sends PI = 8 to the
ISDN, unless the parameter ‘Progress Indicator to ISDN’ is configured
differently.
2 = Ringback tone is played according to 180/183.
For CAS and PRI protocols:
If ‘183 session progress’ with SDP is received, the Gateway cuts
through the voice channel and doesn’t play Ringback tone; PI=8 is
sent in ISDN ALERT message (unless the parameter ‘Progress
Indicator to ISDN’ is configured differently).
If ‘180 ringing’ is received, the CAS Gateway plays Ringback tone to
PSTN; the ISDN Gateway doesn’t play Ringback tone to PSTN. PI is
not sent (unless the parameter ‘Progress Indicator to ISDN’ is
configured differently).
3 = Play Ringback tone if 180 Ringing is received.
Ringback tone is played in all Gateways (including PRI Gateways), if
180 Ringing is received. The PRI Gateway sends PI=8 in ISDN
ALERT message.
ISDNRxOverlap_x
Enable / disable Rx ISDN overlap per trunk ID (x = 0 to 7).
0 = Disabled (default).
1 = Enabled.
Note 1: If enabled, the 3Com VCX V7122 receives ISDN called
number that is sent in the "Overlap" mode.
Note 2: The SETUP message to IP is sent only after the number
(including the ‘Sending Complete’ Info Element) was fully received
(via SETUP and/or subsequent INFO Q.931 messages).
Note3: The ‘MaxDigits’ parameter can be used to limit the length of
the collected number for 3Com VCX V7122 ISDN overlap dialing (if
sending complete was not received).
ISDNTransferCapability
ISDN Transfer Capability can be configured for IPÆISDN calls.
0 = Audio 3.1 (default)
1 = Speech
2 = Data
XChannelHeader
0 = x-channel header is not used (default).
1 = x-channel header is generated, with trunk/B-channel information.
The header provides information on the E1/T1 physical trunk/Bchannel on which the call is received or placed. For example “xchannel: DS/DS1-5/22”. This header is generated by the 3Com VCX
V7122 and is sent in the following messages: INVITE and
183/180/200OK responses.
AddIEinSetup
This parameter enables to add an optional Information Element data
(in hex format) to ISDN SETUP message.
For example: to add the following IE: “0x20,0x02,0x00,0xe1”, define:
“AddIEinSetup = 200200e1”.
Note: This IE is sent from the Trunk Group IDs defined by the
parameter ‘SendIEonTG’.
3Com VCX V7122 SIP VoIP Gateway Release Notes
59
ini File [Web Interface]
Parameter Name
Description
SendIEonTG
A list of Trunk Group IDs (up to 50 characters) from where the
optional ISDN IE, defined by the parameter ‘AddIEinSetup’, is sent.
For example: "SendIEonTG = 1,2,4,10,12,6”.
Revision 4.2
SIP New Features (Version 4.2)
ƒ
T.38 Fax is now supported - according to draft-sip-t38callflows-00 and draft-ietf-sippingrealtimefax-01.
The following call scenario is supported:
Voice call is established→Called side detects a Fax Preamble→REINVITE for
T.38→T.38 fax session.
Relevant parameters:
IsFaxUsed – enables fax on both the caller and the called sides.
CNGDetectorMode – Set to 2 to start fax session on the caller side after CNG tone is
detected (not recommended). Use preamble detection on the fax’s answering side
instead.
For detailed information about additional parameters used to configure the fax/modem
transfer methods, refer to the 3Com VCX V7122 VoIP SIP User's Manual.
ƒ
Call Waiting – If the 3Com VCX V7122 receives a 182 response, it plays Call Waiting
Ringback Tone to the PSTN side.
ƒ
Support for placing call on Hold using one of the two following modes:
ƒ
Locally playing a Hold tone, when a REINVITE message with either the IP address
0.0.0.0 or the “inactive” string in SDP is received.
ƒ
Stop sending RTP packets if “sendonly” is received in REINVITE SDP message. In
this mode it is expected that on hold music (or any other hold tone) will be played
over IP by the remote party.
ƒ
Call Pickup - The Gateway performs Directed Call Pickup when it receives REFER
message with REPLACES header.
ƒ
Full Support for SIP "REPLACES" header (used in transfer message) as defined in draftietf-sip-replaces-02.txt.
ƒ
Supports RFC 2833 DTMF relay payload type negotiation in the SDP.
ƒ
Supports out of band DTMF transport via INFO message.
ƒ
Support for Notify and Subscribe methods for out of band DTMF transport was added
(according to draft-mahy-sipping-signaled-digits-01.txt).
ƒ
When using out of band DTMF (IsDTMFUsed=1), the “DTMFTransportType” is
automatically set to 0, to erase the DTMF digits from the RTP stream.
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3Com VCX V7122 SIP VoIP Gateway Release Notes
ƒ
ƒ
Support for several operational modes with Outbound-Proxy server:
ƒ
SIP RFC 3261: standard rules are used to define which messages are sent directly
to the Proxy server (“IsProxyUsed = 1” and “ProxyIP = <IPaddress>”).
ƒ
All INVITE messages including those generated as a result of Transfer or Redirect
are sent to Proxy server (“IsProxyUsed = 1”, “ProxyIP = <IPaddress>” and
“SendInvitetoProxy = 1”).
ƒ
All SIP messages and responses are sent via Proxy server. (“IsProxyUsed = 1”,
“ProxyIP = <IPaddress>” and “AlwaysSendtoProxy = 1”).
Proxy Hot-Swap mode – If the main Proxy server doesn't respond to an INVITE message
that was retransmitted for a configurable number of times, the call is routed to a
secondary Proxy server.
Relevant parameters: IsProxyHotSwap, ProxyHotSwapRtx.
ƒ
Proxy redundancy parking and homing modes — In homing mode, the Gateway always
tries to work with the primary Proxy server (switches back to the main Proxy whenever it
is available), while in parking mode the Gateway continues working with the last active
Proxy until the next Proxy failure.
Relevant parameter: ProxyRedundancyMode.
ƒ
Support for Sendonly/Recvonly/Inactive attributes in received SDP messages. According
to RFC 3264.
ƒ
Enhanced coder negotiation – if an SDP reply from a remote Gateway includes more
than one coder, the coder is selected (by the receiving gateway) according to order of
appearance (of the coder) in the SDP.
ƒ
Support for G.726 32 kbps coder (instead of G.726 16 kbps) was added.
ƒ
PRACK (Provisional Response Acknowledge) mechanism mode for 1XX reliable
responses — support for calling and called sides (according to RFC 3262). For requests
initiated by the Gateway, PRACK can be configured to: no, optional, and mandatory.
Relevant parameter: PRACKMode.
ƒ
The "User-Agent" header (with software version, board type, etc.) is now added to all
transmitted messages.
ƒ
Supports Network Asserted Identity (RFC 3325) header for IPÆTel calls.
General New Features (Version 4.2)
ƒ
Providing comprehensive Accounting over RADIUS server support.
ƒ
Enhanced Dialing Plan capabilities – Allows entering ranges of numbers, fixed and
opened numbers and the use of wild card characters. Applies to the four manipulation
tables, TelÆIP Routing table and to IPÆTrunk Group Routing table.
When entering a number in the ‘Prefix’ column, you can create an entry that represents
multiple numbers using the following notation:
3Com VCX V7122 SIP VoIP Gateway Release Notes
61
ƒ
[n-m] represents a range of numbers
ƒ
[n,m,l] represents multiple numbers
ƒ
x represents any single digit
ƒ
# represents the end of a number
For example:
ƒ
[5551200-5551300]# represents all of the numbers from 5551200 to 5551300
ƒ
[2221,2231,2241] represents three numbers: 2221, 2231 and 2241
ƒ
54324 represents any number that starts with 54324
ƒ
54324xx# represents a 7 digit number that starts with 54324
ƒ
123[100-200]# represents all of the numbers from 123100 to 123200
ƒ
Call Restriction – when the internal routing table is used (and Proxy isn’t used), reject all
TelÆIP calls that are associated with the destination IP address: 0.0.0.0 in the Tel to IP
routing table.
ƒ
Filter Calls to IP – When Proxy is used, the Gateway checks the Tel to IP routing table
before a telephone number is routed to the Proxy. If the number is not allowed (number
isn’t listed or a call restriction routing rule was applied), the call is released.
Relevant parameter: “FilterCalls2IP = 1”.
ƒ
Alternative Routing (e.g., to implement PSTN Fallback) feature using Tel to IP routing –
For PSTN to IP calls, when the internal routing table (Tel2IP) is used, an alternative route
can now be added. Call is sent to the alternative IP address when no ping to the initial
destination is available or when poor QoS (delay or packet loss, calculated according to
previous calls) is detected.
The alternative routing is configured by adding an additional entry for the same
number/prefix in the Tel2IP routing table.
Note: If the alternative routing destination is the Gateway itself, the call can be
configured to be routed back to one of the Gateway trunk groups and back into the PSTN
(PSTN Fallback).
ƒ
Supported by the 3Com Element Management System (EMS).
ƒ
Channel Select Mode – Several methods for trunk B-channel allocation for IP to TEL
calls. Determined per whole Gateway (ChannelSelectMode) or separately for each trunk
group (Trunk Group Settings table).
ƒ
Routing calls according to DNS host names – In “Tel to IP Routing Table” Fully Qualified
Domain Names (FQDN), such as 3Com.com, can be used instead of IP addresses.
To use this feature you must configure the IP addresses of the primary and secondary
(optional) DNS servers.
ƒ
A new G.168-2000 compliant Echo Canceller, with support for up to 128 msec of echo
tail, has been added. Refer to the new configuration parameter
.MaxEchoCancellerLength.
Note: When set to 64 msec or more, the number of available gateway channels is
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3Com VCX V7122 SIP VoIP Gateway Release Notes
reduced (by a factor of 5/6). For example: Gateway with 8 E1 spans capacity is reduced
to 6 spans (180 channels), while Gateway with 8 T1 spans capacity remains the same
(192 channels).
ƒ
ISDN Overlap receiving — the 3Com VCX V7122 can now receive PSTN phone
numbers that are sent in the "Overlap" mode (refer to ISDNRxOverlap, ini file
parameter).
Note: request to IP is sent only after the number (including “Sending Complete” Info
Element) was fully received (via SETUP and/or subsequent INFO Q.931 messages).
ƒ
Transparent Protocol support — if trunks are configured to transparent protocol, then call
is established without applying any PSTN protocol, e.g., trunks under SS7 signaling
control. RTP is sent and received on the TDM slot.
ƒ
Gateway’s channel internal number can now be used as a ‘destination number’ for
TelÆIP calls, if called number, was not received from PSTN.
(ReplaceEmptyDstWithPortNumber).
ƒ
“Add Port as prefix” feature — For TelÆIP incoming calls, trunk ID number (1-8) is added
as prefix to the called phone number. Can be used to define various routing rules.
ƒ
Syslog CDR support enhanced — A Call Detail Record (CDR) can now be sent at both
the end and start of a call (after INVITE message was sent/received) to Syslog server.
ƒ
RTP Broken Connection — Call is disconnected if RTP packets aren’t received for a
configurable time period during the call (BrokenConnectionEventTimeout).
ƒ
Robust reception of RTP streams using a new filtering mechanism. This new mechanism
filters out unwanted RTP streams that are sent to the same port on the board. These
multiple RTP streams can result from traces of previous calls, call control errors or
deliberate attacks. As a result, a port may accept packets only from one known source at
a time.
ƒ
Support for reception of RTP packets with the header Padding bit set to 1.
ƒ
Support for reordered RTP packets — a new algorithm was implemented to handle
reordered RTP packets. This feature improves the voice quality on a network which
suffers from packet reordering problems.
ƒ
Configurable Default Release Cause (to IP or to PSTN) — when the Gateway terminates
a call, and if an explicit matching cause for this termination isn’t found, a default release
cause can be configured (DefaultReleaseCause):
The default release cause is:
Other common values are:
NO_ROUTE_TO_DESTINATION (3).
NO_CIRCUIT_AVAILABLE (34) or
NETWORK_OUT_OF_ORDER (38), etc.
Note: The default release cause is described in the Q.931 notation, and is translated to
the corresponding SIP 4xx and 5xx values.
ƒ
Option to delay Gateway’s operation – After a reset cycle, the Gateway’s operation can
now be delayed for a specified time (according to the GWAppDelayTime parameter).
This feature helps to overcome connection problems caused by specific routers.
3Com VCX V7122 SIP VoIP Gateway Release Notes
63
ƒ
Common Debug Level Parameter – Syslog Debug levels can now be configured via a
single parameter, GwDebugLevel, instead of separate "logger objects". Usually set to 5 if
debug traces are needed.
Embedded Web Server New Features (Version 4.2)
ƒ
Online loading of CAS tables via the Embedded Web Server is now available.
ƒ
Number Plan and Number Type values are presented in Number Manipulation tables
according to Table 1 in ETS 300 189 standard.
The following Plan, Type values are supported:
ƒ
0,0 – Unknown, Unknown
ƒ
9,0 – Private, Unknown
ƒ
9,1 – Private, Level 2 Regional
ƒ
9,2 – Private, Level 1 Regional
ƒ
9,3 – Private, PISN Specific
ƒ
9,4 – Private, Level 0 Regional (local)
ƒ
1,0 – Public(ISDN/E.164), Unknown
ƒ
1,1 – Public(ISDN/E.164), International
ƒ
1,2 – Public(ISDN/E.164), National
ƒ
1,3 – Public(ISDN/E.164), Network Specific
ƒ
1,4 – Public(ISDN/E.164), Subscriber
ƒ
1,6 – Public(ISDN/E.164), Abbreviated
ƒ
Invalid parameter value warning alert – invalid parameters are colored red and a short
warning message is displayed.
ƒ
Users can now retrieve the Gateway’s configuration in an ini file format; the ini file
(downloaded from the Gateway via the Embedded Web Server) contains all parameters
that are different from their default value. The ini file can then be uploaded to a second
Gateway to apply the same configuration.
ƒ
Message log page – adds the option to watch the error logs directly without an external
Syslog server.
ƒ
Save Configuration button - burning the current configuration to the flash memory without
resetting the Gateway. Resetting the board should be done before activating traffic (or at
a low traffic period).
ƒ
New administration feature - Logo images upload. User can change the logo that
appears in the Web pages.
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3Com VCX V7122 SIP VoIP Gateway Release Notes
ƒ
A number of parameters have been upgraded with on-the-fly capability. In the Embedded
Web Server, parameters that can be changed on-the-fly are noted with an asterisk (*).
SNMP New Features (Version 4.2)
ƒ
Updated SNMP MIB files for SIP parameters and other gateway (ACL MIB) parameters.
ƒ
New Trap Manager Table - SNMPManagers – providing online configuration for up to
three Managers used for receiving SNMP Traps. Each of the following parameters can
be configured separately for each Manager: IP address, port number and whether it is
active or not. (Related parameters: SNMPManagerTableIP, SNMPManagerTrapPort,
SNMPManagerIsUsed and SNMPManagerTrapSendingEnable).
To enable SNMP Traps set “SNMPManagerIsUsed=1” in the ini file.
ƒ
Traps - Traps are sent when major problems are encountered. Clear trap is sent when
problem is solved.
Traps Include:
ƒ
ƒ
General Fatal Error
ƒ
Configuration Error
ƒ
Controller lost (Proxy)
ƒ
Call resources low (EnableRAI parameter must be enabled)
ƒ
Gateway Overload
Community Strings for Get and Set are configurable via ini file parameter
SetCommunityString. The same string (up to 20 characters) is used for Set and for Get.
Resolved Constraints (Version 4.2)
ƒ
Responses are now sent to source IP address and not to the IP address specified in the
Via header (RFC 3261).
ƒ
CANCEL, ACK and PRACK messages are now sent to the correct IP addresses.
ƒ
The CANCEL message, sent before 200 OK response is received, now gets the same
URI as the originating INVITE message.
ƒ
Configurable ProxyKeepAliveTime and MaxRtx.
ƒ
DomainName was enlarged to 30 characters.
ƒ
Sending 481 "Call/Transaction does not exist" message in response to a re-INVITE or
INFO audit requests for session that does not exist.
ƒ
Setting the T38MaxBitRate parameter in SDP during fax negotiation is set according to
"FaxRelayMaxRate" parameter.
ƒ
Support for received SIP messages of up to 1700 bytes.
ƒ
Static NAT support.
3Com VCX V7122 SIP VoIP Gateway Release Notes
65
ƒ
Various call hold and transfer services using REFER and REPLACES were fixed.
ƒ
Out of band DTMF INFO/NOTIFY messages are also sent if Call is in hold state (no RTP
is sent).
ƒ
Second Registrar request with MD5 response is sent without the “To” tag, same as the
first Registrar request.
ƒ
ISDN NFAS (Non Facility Associated Signaling) support.
ƒ
IPÆTrunk group routing table was increased to support 24 rules.
ƒ
Only one simultaneous source of incoming RTP packets is allowed per channel.
ƒ
MFCR2 supports release causes (Busy, congestion, etc.) for PSTNÆIP calls.
New Parameters (Version 4.2)
Most new parameters (described in Table 16) can be configured with the ini file and via the
Embedded Web Server. Note that only those parameters contained within square brackets
are configurable via the Embedded Web Server.
Table 16
ini File [Web Browser] Parameter Name
ini File [Web Interface] Parameter Name
Description
ProtocolType
Support for additional protocols:
4 = T1_TRANSPARENT
5 = E1_TRANSPARENT_31
6 = E1_TRANSPARENT_30
15 = J1_TRANSPARENT
ISDNRxOverlap
0 = Disabled (default)
1 = Enabled
Note 1: If enabled the 3Com VCX V7122 receives ISDN
called number that is sent in the "Overlap" mode.
Note 2: The SETUP to IP is sent only after the number
(including “Sending Complete” Info Element) was fully
received (via SETUP and/or subsequent INFO Q.931
messages).
AlwaysSendtoProxy
0 = Use standard SIP routing rules (default)
1 = All SIP messages and Responses are sent to Proxy
server
Note: Applicable only if Proxy server is used.
SendInviteToProxy
0 = INVITE messages, generated as a result of Transfer or
Redirect, are sent directly to the URL (according to the referto header in the REFER message or contact header in 30x
response).
1 = All INVITE messages, including those generated as a
result of Transfer or Redirect are sent to Proxy.
Note: Applicable only if Proxy server is used and
“AlwaysSendtoProxy=0”.
EnableProxyKeepAlive
[Enable Proxy Keep Alive]
0 = Disable (default)
1 = Keep alive with Proxy, by sending "OPTIONS" SIP
message every “ProxyKeepAliveTime”.
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3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter Name
Description
ProxyKeepAliveTime
Defines the Proxy keep-alive time interval (in seconds)
between OPTIONS messages.
The default value is 60 seconds.
SIPMaxRtx
[SIP MAX Rtx]
Number of UDP retransmissions of SIP messages.
The range is 1 to 7.
The default value is 7.
EnableRPIHeader
[Enable Remote Party ID]
Enable-Remote-Party-ID Headers for calling and called
numbers for TelÆIP calls.
0 = Disable (default)
1 = Enable
IsProxyHotSwap
[Enable Proxy HotSwap]
Enable Proxy Hot Swap redundancy mode.
0 = Disabled (default)
1 = Enabled
If Hot Swap is enabled, SIP INVITE message is first sent to
the primary Proxy server. If there is no response from the
primary Proxy server for “ProxyHotSwapRtx”
retransmissions, the INVITE message is resent to the
redundant Proxy server.
ProxyHotSwapRtx
[Number of RTX before HotSwap]
Number of retransmitted INVITE messages before call is
routed (hot swap) to another Proxy.
Range: 1-30
Default: 3
ProxyRedundancyMode
[Redundancy Mode]
0 = Parking mode: Gateway continues working with the last
active Proxy until the next failure. (default)
1 = Homing mode: Gateway always tries to work with the
primary Proxy server (switches back to the main Proxy
whenever it is available).
PRACKMODE
[PRACK Mode]
PRACK mechanism mode for 1XX reliable responses:
0 = Disabled
1 = Supported (default)
2 = Required
Note 1: The Supported and Required headers contain the
“100rel” parameter respectively.
Note 2: 3Com VCX V7122 sends PRACK message if
180/183 response is received with “100rel” in the Supported
or the Required headers.
xferPrefix
Defined string that is added, as a prefix, to the transferred
called number, using REFER message.
Note 1: The number manipulation rules apply to the user
part of the “REFER-TO” URL before it is sent in the INVITE
message.
Note 2: The xferprefix parameter can be used to apply
different manipulation rules to differentiate the transferred
number from the original dialed number.
ReplaceEmptyDstWithPortNumber
0 = Disabled (default).
1 = Enabled, Internal channel number is used as a
destination number if called number is missing.
Note: Applicable only to TelÆIP calls, if called number is
missing.
3Com VCX V7122 SIP VoIP Gateway Release Notes
67
ini File [Web Interface] Parameter Name
Description
MaxEchoCancellerLength
Maximum Echo Canceler Length in msec:
0 = Internal decision to keep max channel capacity (currently
32 msec)
4 = 32 msec
5 = 35 msec
6 = 40 msec
7 = 45 msec
8 = 50 msec
9 = 55 msec
10 = 60 msec
11 = 64 msec, reduced channels (max channels capacity is
200)
22 = 128 msec, reduced channels (max channels capacity is
200)
The default value is 0.
AlwaysUseRouteTable
[Use Routing Table For Host Names]
Use the internal routing table to obtain the URL Host name,
even if Proxy server is used.
0 = Don’t use (default)
1 = Use
Note: This Domain name is used, instead of Proxy name or
Proxy IP address, in the INVITE SIP URL.
IsFaxUsed
0 = T.38 Fax relay disabled (default)
1 = Enable T.38 Fax Relay
Note: FaxTransportMode can be set to 0 (transparent). The
gateway automatically changes the Fax transport mode to
T.38 if “IsFaxUsed=1” and fax is detected.
“IsFaxUsed=0” fax can be sent (transparently) if G.711
coder is used.
CngDetectorMode
0 = Don’t detect CNG (default)
2 = Detect CNG on caller side and start fax session (if
IsFaxUsed=1)
Usually T.38 fax session starts when the “preamble” signal
is detected by the answering side. Some SIP gateways do
not support the detection of this fax signal on the answering
side. For these cases it is possible to configure the
Gateways to start the T.38 fax session when the CNG tone
is detected by the originating side. However this mode is not
recommended.
AltRoutingTel2IPEnable
[Enable Alt Routing Tel2IP]
0 = Alternative Routing is disabled (default)
1 = Alternative Routing is enabled
AltRoutingTel2IPMode
[Alt Routing Tel2IP Mode]
Alternative routing is performed if:
0 = Alternative routing according to PING and QoS is
disabled
1 = Ping to initial destination failed
2 = QOS, poor quality of service was detected
3 = Both, either Ping to initial destination failed, or poor
quality of service was detected (default)
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3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter Name
Description
ChannelSelectMode
[Channel Select Mode]
Defines port allocation algorithm for IP to TEL calls.
Note: Replaces the obsolete parameter “IsUseFreeChannel”
(Cyclic Ascending mode provides a similar functionality to
the “IsUseFreeChannel” parameter).
ƒ
0 = By phone number — Select the Gateway port
according to the called number (called number is
defined in the ‘Trunk Group’ table).
ƒ
1 = Cyclic Ascending — Select the next available
channel in an ascending cycle order. Always select the
next higher channel number in the Hunt Group. When
the Gateway reaches the highest channel number in the
Hunt Group, it will select the lowest channel number in
the Hunt Group and then start ascending again
(default).
ƒ
2 = Ascending — Select the lowest available channel.
Always start at the lowest channel number in the Hunt
Group and if that channel is not available, select the
next higher channel.
ƒ
3 = Cyclic Descending — Select the next available
channel in descending cycle order. Always select the
next lower channel number in the Hunt Group. When
the Gateway reaches the lowest channel number in the
Hunt Group, it will select the highest channel number in
the Hunt Group and then start descending again.
ƒ
4 = Descending — Select the highest available channel.
Always start at the highest channel number in the Hunt
Group and if that channel is not available, select the
next lower channel.
ƒ
5 = Number + Cyclic Ascending — First select the
Gateway port according to the called number. If the
called number isn’t found, then select the next available
channel in ascending cyclic order. Note that if the called
number is found, but the port associated with this
number is busy, the call is released.
TrunkGroupSettings
New table named:
[Hunt Group Settings]
Define rules for port allocation for specific Hunt Groups. If
such rules do not exist, the global rule defined by
ChannelSelectMode applies.
a, b
a = Trunk Group ID number
b = Channel select mode for that Trunk Group.
Available values are identical to those defined by the
ChannelSelectMode parameter.
DNSPriServerIP
[DNS Primary Server IP]
IP address of the primary DNS server in dotted format
notation.
DNSSecServerIP
[DNS Secondary Server IP]
IP address of the primary DNS server in dotted format
notation.
3Com VCX V7122 SIP VoIP Gateway Release Notes
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ini File [Web Interface] Parameter Name
Description
DefaultReleaseCause
Default Release Cause (to IP), used when the Gateway
initiates a call release, and if an explicit matching cause for
this release isn’t found, a default release cause can be
configured. The default release cause is described in the
Q.931 notation, and translated to corresponding SIP
equivalent response value
The default release cause is:
NO_ROUTE_TO_DESTINATION (3).
Other common values are: NO_CIRCUIT_AVAILABLE (34)
or NETWORK_OUT_OF_ORDER (38), etc.
FilterCalls2IP
[Filter Calls To IP]
0 = Disabled (default)
1 = Enabled
Note: If filter calls to IP feature is enabled, then when Proxy
is used, the Gateway checks first the TelÆIP routing table
before a telephone number is routed to the Proxy. If the
number is not allowed (number isn’t listed or a negative
routing rule was applied), the call is released.
RxDTMFOption
[Rx DTMF Option]
Defines the supported Receive DTMF negotiation method.
0 = Don’t declare RFC 2833 Telephony-event parameter in
SDP
1 = n/a
2 = n/a
3 = Declare RFC 2833 “Telephony-event” parameter in SDP
(default)
The GW is designed to always be receptive to RFC 2833
DTMF relay packets. Therefore, it is always correct to
include the “Telephony-event” parameter as a default in the
SDP. However some gateways use the absence of the
“telephony-event” from the SDP to decide to send DTMF
digits inband using G.711 coder. If this is the case you can
set “RxDTMFOption=0”.
TxDTMFOption
[DTMF RFC2833 Negotiation]
0 = No negotiation, DTMF digit is sent according to the
“DTMFTransportType” parameter
4 = Enable RFC 2833 payload type (PT) negotiation
Note 1: This parameter is applicable only if
“IsDTMFUsed=0” (out of band DTMF is not used).
Note 2: If enabled, the Gateway:
ƒ
Negotiates RFC 2833 payload type using local and
remote SDPs.
ƒ
Sends DTMF packets using RFC 2833 PT according to
the received SDP.
ƒ
Expects to receive RFC 2833 packets with the same PT
as configured by the “RFC2833PayloadType”
parameter.
Note 3: If the remote party doesn’t support the RFC 2833
DTMF relay, the Gateway uses the same PT for send and
receive.
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3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter Name
Description
OutOfBandDTMFFormat
The exact method to send out of band DTMF digits
1 = INFO format (Nortel)
2 = INFO format (Cisco) - (default)
3 = NOTIFY format <draft-mahy-sipping-signaled-digits01.txt>
Note 1: To use out of band DTMF, set “IsDTMFUsed=1” or
“Enable DTMF = yes”.
Note 2: When using out of band DTMF, the
“DTMFTransportType” parameter is automatically set to 0, to
erase the DTMF digits from RTP path.
AddPortAsPrefix
[Add Port as Prefix]
0 = Don’t add (default)
1 = Add trunk ID number (single digit in the range 1 to 8) as
a prefix to the called phone number for TelÆIP incoming
calls.
This option can be used to define various routing rules.
GWAppDelayTime
Defines the amount of time (in seconds) the Gateway’s
operation is delayed after a reset cycle.
The default value is 0 seconds
Note: This feature helps to overcome connection problems
caused by some LAN routers.
DisableNAT
0 = NAT is enabled
1 = NAT is disabled (default)
If NAT is enabled, then the source IP address, of the first
received RTP packet on a new session, is compared to the
remote IP address, stated when session was opened; if they
are not identical, then destination IP address of the outgoing
RTP packets will be the source IP address of the first
incoming packet.
SNMP Managers - a device that is used for
receiving SNMP Traps.
SNMPManagerISUsed_x
Up to three parameters, each controls the validity of the
parameters (IP address and port number) of the
corresponding SNMP Manager.
0 = Disabled (default)
1 = Enabled
(SNMPManagerIsUsed_0, SNMPManagerIsUsed_1,
SNMPManagerIsUsed_2)
SNMPManagerTrapSendingEnable_x
Up to three parameters, each controls the
activation/deactivation of sending traps to the corresponding
SNMP Manager.
0 = Sending is disabled
1 = Sending is enabled (default)
(SNMPManagerTrapSendingEnable_0,
SNMPManagerTrapSendingEnable_1,
SNMPManagerTrapSendingEnable_2)
3Com VCX V7122 SIP VoIP Gateway Release Notes
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ini File [Web Interface] Parameter Name
Description
SNMPManagerTableIP_x
Up to three IP addresses of remote hosts that are used as
an SNMP Managers.
(SnmpManagerIP_0, SnmpManagerIP_1,
SnmpManagerIP_2)
Enter the IP address in dotted format notation, for example
108.10.1.255.
Note: This parameter replaces the obsolete parameter
SNMPManagerIP.
SNMPManagerTrapPort_x
Up to three parameters used to define the Port numbers of
the remote SNMP Managers (SNMPManagerTrapPort_0,
SNMPManagerTrapPort_1, and SNMPManagerTrapPort_2).
Note: This parameter replaces the obsolete parameter
SNMPTrapPort.
The default SNMP trap port is 163.
The SNMP trap port must be less than 4000.
SetCommunityString
SNMP community string (up to 20 chars).
Default community strings are “public” for read, and “private”
for set & get.
ModemRtpByPassPayloadType
Modem Bypass dynamic payload type (range 0-127).
The default value is 103.
FaxModemBypassBasicRtpPacketInterval
0 = set internally (default)
1 = 5 msec (not recommended)
2 = 10 msec
3 = 20 msec
NSEMode
Cisco compatible modem bypass mode
0 = NSE disabled (default)
1 = NSE enabled
Note 1: This feature can be used only if
VxxModemTransportType=2 (Bypass)
Note 2: If NSE mode is enabled the SDP contains the
following line:
“a=rtpmap:100 X-NSE/8000”
Note 3: To use this feature:
ƒ
The Cisco gateway must include the following definition:
"modem passthrough nse payload-type 105 codec
g711alaw".
ƒ
Set the Modem transport type to Bypass mode
(“VxxModemTransportType = 2”) for all modems.
NSEPayloadType
NSE payload type (range 96-127)
The default value is 105.
Note: The Cisco gateways usually use NSE payload type of
100.
BrokenConnectionEvent
Timeout
The amount of time (in 100 msec units) an RTP packet isn’t
received, after which a call is disconnected.
The default value is 100 (10 seconds).
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3Com VCX V7122 SIP VoIP Gateway Release Notes
ini File [Web Interface] Parameter Name
Description
GwDebugLevel
[Debug Level]
Defines the Syslog logging level (usually set to 5 if debug
traces are needed).
0 = Debug is disabled (default)
1 = Flow debugging is enabled
2 = Flow and board interface debugging are enabled
3 = Flow, board interface and stack interface debugging are
enabled
4 = Flow, board interface, stack interface and session
manager debugging are enabled
5 = Flow, board interface, stack interface, session manager
and board interface expanded debugging are enabled.
Usually set to 5 if debug traces are needed.
EnableRADIUS
0 = RADIUS server is disabled (default).
1 = RADIUS server is enabled.
MaxRADIUSSessions
Number of concurrent calls that can communicate with the
RADIUS server (optional).
Range: 0-240.
The default value is 0.
SharedSecret
“Secret” used to authenticate the Gateway to the RADIUS
server. It should be a cryptographically strong password.
RADIUSRetransmission
Number of retransmission retries.
Range: 1-10.
The default value is 3.
RadiusTO
The interval between each retry (measured in seconds).
Range: 1-30.
The default value is 10.
RADIUSAccServerIP
IP address of accounting server.
RADIUSAccPort
Port number of accounting server.
The default value is 1646.
AAAIndications
0 = No indications (default).
3 = Accounting only
3Com VCX V7122 SIP VoIP Gateway Release Notes
73
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3Com VCX V7122 SIP VoIP Gateway Release Notes
APPENDIX A: OBTAINING SUPPORT FOR
YOUR 3COM PRODUCTS
3Com offers product registration, case management, and repair services through
eSupport.3com.com. You must have a user name and password to access these services,
which are described in this appendix.
Register Your Product to Gain Service Benefits
To take advantage of warranty and other service benefits, you must first register your
product at:
http://esupport.3com.com/
3Com eSupport services are based on accounts that are created or that you are authorized
to access.
Solve Problems Online
3Com offers these support tools:
ƒ
3Com Knowledgebase — Helps you to troubleshoot 3Com products. This query-based
interactive tool is located at:
http://knowledgebase.3com.com/
It contains thousands of technical solutions written by 3Com support engineers.
ƒ
Connection Assistant — Helps you to install, configure, and troubleshoot 3Com
desktop and server network interface cards (NICs), wireless cards, and Bluetooth
devices. This diagnostic software is located at:
http://www.3com.com/connectionassistant
Purchase Extended Warranty and Professional Services
To enhance response times or extend your warranty benefits, you can purchase value-added
services such as 24x7 telephone technical support, software upgrades, onsite assistance, or
advanced hardware replacement.
3Com VCX V7122 SIP VoIP Gateway Release Notes
75
Experienced engineers are available to manage your installation with minimal disruption to
your network. Expert assessment and implementation services are offered to fill resource
gaps and ensure the success of your networking projects. For more information on 3Com
Extended Warranty and Professional Services, see:
http://www.3com.com/
Contact your authorized 3Com reseller or 3Com for additional product and support
information. See the table of access numbers later in this appendix.
Access Software Downloads
You are entitled to bug fix / maintenance releases for the version of software that you initially
purchased with your 3Com product. To obtain access to this software, you need to register
your product and then use the Serial Number as your login. Restricted Software is available
at:
http://esupport.3com.com/
To obtain software releases that follow the software version that you originally purchased,
3Com recommends that you buy an Express or Guardian contract, a Software Upgrades
contract, or an equivalent support contract from 3Com or your reseller. Support contracts
that include software upgrades cover feature enhancements, incremental functionality, and
bug fixes, but they do not include software that is released by 3Com as a separately ordered
product. Separately orderable software releases and licenses are listed in the 3Com Price
List and are available for purchase from your 3Com reseller.
Contact Us
3Com offers telephone, internet, and e-mail access to technical support and repair services.
To access these services for your region, use the appropriate telephone number, URL, or email address from the table in the next section.
Telephone Technical Support and Repair
To obtain telephone support as part of your warranty and other service benefits, you must
first register your product at:
http://esupport.3com.com/
When you contact 3Com for assistance, please have the following information ready:
ƒ
Product model name, part number, and serial number
ƒ
A list of system hardware and software, including revision level
ƒ
Diagnostic error messages
ƒ
Details about recent configuration changes, if applicable
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3Com VCX V7122 SIP VoIP Gateway Release Notes
To send a product directly to 3Com for repair, you must first obtain a return materials
authorization number (RMA). Products sent to 3Com without authorization numbers clearly
marked on the outside of the package will be returned to the sender unopened, at the
sender's expense. If your product is registered and under warranty, you can obtain an RMA
number online at http://esupport.3com.com/. First-time users must apply for a user name and
password.
Telephone numbers are correct at the time of publication. Find a current directory of 3Com
resources by region at:
http://csoweb4.3com.com/contactus/
Country
Telephone Number
Country
Telephone Number
Asia, Pacific Rim — Telephone Technical Support and Repair
Australia
1 800 678 515
Pakistan
+61 2 9937 5083
Hong Kong
800 933 486
Philippines
1235 61 266 2602 or
1800 1 888 9469
India
+61 2 9424 5179 or
000800 650 1111
P.R. of China
800 810 3033
Indonesia
001 803 61009
Singapore
800 6161 463
Japan
00531 616 439 or 03
5977 7991
S. Korea
080 333 3308
Malaysia
1800 801 777
Taiwan
00801 611 261
New Zealand
0800 446 398
Thailand
001 800 611 2000
You can also obtain support in this region at this e-mail address: apr_technical_support@3com.com
Or request a return material authorization number (RMA) by FAX using this number: +61 2 9937 5048
Europe, Middle East, and Africa — Telephone Technical Support and Repair
From anywhere in these regions, call: +44 (0)1442 435529
From the following countries, call the appropriate number:
Austria
01 7956 7124
Luxembourg
342 0808128
Belgium
070 700 770
Netherlands
0900 777 7737
Denmark
7010 7289
Norway
815 33 047
Finland
01080 2783
Poland
00800 441 1357
France
0825 809 622
Portugal
707 200 123
Germany
01805 404 747
South Africa
0800 995 014
Hungary
06800 12813
Spain
9 021 60455
3Com VCX V7122 SIP VoIP Gateway Release Notes
77
Country
Telephone Number
Country
Telephone Number
Ireland
01407 3387
Sweden
07711 14453
Israel
1800 945 3794
Switzerland
08488 50112
Italy
199 161346
U.K.
0870 909 3266
You can also obtain support in this region using this URL:
http://emea.3com.com/support/email.html
Latin America — Telephone Technical Support and Repair
Antigua
1 800 988 2112
Guatemala
AT&T +800 998 2112
Argentina
0 810 444 3COM
Haiti
57 1 657 0888
Aruba
1 800 998 2112
Honduras
AT&T +800 998 2112
Bahamas
1 800 998 2112
Jamaica
1 800 998 2112
Barbados
1 800 998 2112
Martinique
571 657 0888
Belize
52 5 201 0010
Mexico
01 800 849CARE
Bermuda
1 800 998 2112
Nicaragua
AT&T +800 998 2112
Bonaire
1 800 998 2112
Panama
AT&T +800 998 2112
Brazil
0800 13 3COM
Paraguay
54 11 4894 1888
Cayman
1 800 998 2112
Peru
AT&T +800 998 2112
Chile
AT&T +800 998 2112
Puerto Rico
1 800 998 2112
Colombia
AT&T +800 998 2112
Salvador
AT&T +800 998 2112
Costa Rica
AT&T +800 998 2112
Trinidad and Tobago
1 800 998 2112
Curacao
1 800 998 2112
Uruguay
AT&T +800 998 2112
Ecuador
AT&T +800 998 2112
Venezuela
AT&T +800 998 2112
Dominican Republic
AT&T +800 998 2112
Virgin Islands
57 1 657 0888
You can also obtain support in this region in the following ways:
ƒ
Spanish speakers, enter the URL: http://lat.3com.com/lat/support/form.html
ƒ
Portuguese speakers, enter the URL: http://lat.3com.com/br/support/form.html
ƒ
English speakers in Latin America, send e-mail to: lat_support_anc@3com.com
US and Canada — Telephone Technical Support and Repair
All locations:
78
Network Jacks; Wired or Wireless Network Interface Cards:
1 847-262-0070
All other 3Com products:
1 800 876 3266
3Com VCX V7122 SIP VoIP Gateway Release Notes