Netborder Express Gateway Release NotesVersion 2.0 General Availability : May 29, 2009
1 Product Compatibility
Here are some of the major compatibility points.
- Systems with Intel Based Processors only. AMD processors are not supported
- Operating Systems Supported:
- Microsoft Windows XP 32 bit
- Microsoft Windows Vista 32 bits
- Microsoft Windows 2003 Server 32 bits
- Microsoft Windows 2008 Server 32 bits
- Sangoma Telephony Cards Supported:
- AFT A101/2/4/8 T1/E1 with hardware echo cancellation (PCI / PCI-Express)
- AFT A200 FXO with hardware echo cancellation (PCI / PCI-Express)
- Sangoma Software Release Versions supported:
- 6.0.9.12 (included in gateway software package)
- SIP 3261 compliant endpoints using either TCP or UDP as the transport protocol
- DTMF relay as per IETF RFC 2833.
- RTP/RTCP as per IETF RFC 3550/3551
- Minimum Server requirements: Intel Core Duo 2 processor or later with a minimum 512
MB of RAM.
Feature Support
| Feature |
Notes |
| PSTN-initiated calling |
- Support FXO analog interface
- Support ISDN-PRI Q931 (DMS100, 4ESS, 5ESS, National ISDN 2) terminal and network sides.
- NFAS for DMA100, 4ESS, 5ESS and National ISDN 2 variants terminal and network sides.
- FXS Analog is not supported
- NFAS with D-Channel backup is not supported
- CAS is not supported
|
| SIP-initiated callling |
The Gateway listens on port 5066 by default. |
| Support for 3xx redirect primitives |
Includes “hybrid” redirect (redirecting to either SIP or PSTN endpoint) |
| SIP Registration |
Allows to register the gateway to a third party SIP registrar |
RTP processing as per RFC 3550 and RTCP as per RFC 3551 |
G.711 codecs (uLaw and A-law) with law conversion. |
| DTMF per RFC 2833 |
Both DTMF relay (PSTN to SIP) and DTMF re-generation (SIP to PSTN) |
Mapping of PSTN calls to SIP endpoints through rules, including DNIS-based routing |
Configurable routing rules |
Mapping of SIP calls to PSTN ports, trunks and DN through rules |
Configurable routing rules |
CallerID/ANI/DNIS and custom information element |
Available in SIP message |
| Packaged as a Windows Service |
|
| Integrated to Windows Event Viewer |
|
| Configurable logging per sub-system |
|
| Call logs |
Per call information |
| Web Service Interface for management |
|
2 Limitations and Known Problems
Here is the list of known problems and limitations.
2.1 Hardware & driver related limitations
- AFT-400 boards are NOT supported.
- The gateway does not support T.38 Fax relay.
- Support Sangoma Software version 6.0.9.12. The gateway has been tested and
validated with Sangoma software 6.0.9.12. The gateway validates this version and generates an error if the version is different. The gateway will also fail to start.
- Echo cancellation tail length is fixed to 128ms for all calls. The echo cancellation is
performed by the hardware. Thus, having support for shorter tail length will have no impact of the overall performance of the gateway.
2.2 Other limitations
- Documentation (User Guide)
- Include Quick Start and Tone Configuration guides only
- Analog disconnect supervision
- Low amplitude telephony tone are not always detected (Bug 1601)
- Disconnect tone is not supported
- Gateway Web User Interface
- Supported web browsers : Internet Explorer, Mozilla Firefox. Google Chrome is not
yet supported
- FXO caller-ID
- Support is limited to caller-ID extraction as described by Bellcore FSK 1200bps
Caller-ID standards in SDMF or MDMF which is used in Australia, Canada, China, Hong Kong, New Zealand, Singapore and USA. The gateway extracts only the caller number from the caller-ID in SDMF mode and extract caller number and caller name in MDMF mode. ETSI FSK caller ID and caller name is also supported
- Service shutdown while waiting to register/unregister to a SIP registrar may cause
shutdown timeout: If the feature of registering the gateway with a SIP registrar is used, and the gateway is waiting for a reply from a registrar that is particularly slow or down, it is possible that a service shutdown request times out in Windows before we can complete the operation (register or unregister). The impact is simply that the service shutdown is not very elegant.
- Gateway Does Not Monitor the Via or Max hops Headers for Self-Loops: If users
design ill-formed routing rules, it could happen that they re-direct incoming SIP calls to the gateway’s SIP user agent. The gateway does not currently ensure that the ‘via’ header is different from the source of the call nor that ‘maxhops’ is not violated. This could cause an infinite loop of SIP calls.
- Limitations to the use of arbitrary SIP headers in the routing rules:
- If two headers of the same name are specified in the sip.out.header out parameters,
only the last one is used
- If a “known” SIP header (automatically generated by the gateway, as described in a
point below) is used in sip.out.header, the header internally generated will not be overridden, creating two headers that have a great chance of confusing the remote SIP user agent.
- Known SIP headers, automatically generated by the gateway, cannot be used as
sip.in.header.* parameters. The list of all known headers follows:
VIA, FROM, TO, CSEQ, CALLID, CONTENTLENGTH, ACCEPTENCODING, ACCEPT, ACCEPTLANGUAGE, ALERTINFO, ALLOW, ALLOWEVENTS, AUTHENTICATE, AUTHENTICATIONINFO, AUTHORIZATION, CALLINFO, CCDIVERSION, CONTACT, CONTENTDISPOSITION, CONTENTENCODING, CONTENTTYPE, DATE, ENCRYPTION, ERRORINFO, EVENT, EXPIRES, HIDE, INREPLYTO, MAXFORWARDS, MIMEVERSION, MINEXPIRES, MINSE, ORGANIZATION, PRIORITY, PROXYAUTHENTICATE, PROXYAUTHORIZATION, PROXYREQUIRE, RACK, RSEQ, RECORDROUTE, REFERTO, REFERREDBY, REPLACES, REQUIRE, RESPONSEKEY, RETRYAFTER, ROUTE, SERVER, SESSIONEXPIRES, SESSION, SUBJECT, SUBSCRIBESTATE, SUPPORTED, TIMESTAMP, UNKNOWN, UNSUPPORTED, USERAGENT, WWWAUTHENTICATE, WARNING.
3 Changes Since Last Release
3.1 Release 2.0 General Availability
- This software supports both analog FXO & digital PRI telephony interfaces.
- In band tone detection is now performed when required in PRI outbound calls.
- The SIP PRACK request is supported in both directions.
- Added support of ISDN Network side (act like a telco switch) for all ISDN variants
supported by the gateway.
- Added caller name support for all ISDN variants.
- Improved Web User Interface (UI)
- All gateway configuration files can be edited through the UI
3.2 Release 2.0 Limited Availability
- Extended of FXO connectivity to support most countries around the world (please consult
the Web User Interface to get the list of these countries). However, the user has to edit tone definition files and the .RAM files used to regenerate call progress tones for all countries except AUSTRIALIA, CANADA and USA. Consult the Tone_Configuration_guide.pdf for more details.
- Added support to set the Type Of Service (TOS) field in the IP header of the RTP and
RTCP packets transmitted by the gateway via the parameter “Netborder.media.ip.tos” in the gw.properties file.
- Extended FXO disconnect supervision to support battery removal and reverse battery
disconnect detectors.
- Improved audio quality.
3.3 Release 2.0 Beta
- This release is the first to offer FXO analog PSTN connectivity limited to North America countries.
- The Gateway Web User Interface has been redesigned and its capabilities have been greatly augmented :
- The Gateway service can be started/stopped from the Web User Interface
- Initial gateway configuration can be generated by a Web UI wizard
- Most telephony configuration parameters can be modified through the Web UI.
3.4 Release 1.6.2
- Fixed problem where the Windows user interface was not responding to user input as
soon as the user had started the gateway WEB interface on server where the gateway is running. This problem was observed only systems having a single core CPU.
3.5 Release 1.6.1
- Fixed a problem with the DTMF detection on B-Channel 23 of T1 spans
- Fixed gateway crash that could occurs with some NFAS configurations.
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