Release Notes - Digium Gateways G100/G200/G400/G800

Version 3.0.1 - December 17, 2018

Improvements Made

Update to support new manufacturing process and hardware revision

Version 2.4.7 - October 16, 2018

Bug Fixes

Interconnect is now compatible with Switchvox 6.7.1

Version 2.4.6 - September 26, 2018

Improvements Made

Gateway no longer appears as an option on phone server selection menus.
Updated internal encryption methods

Version 2.4.5 - July 12, 2018

Improvements Made

SIP 183 response is never sent when Progress Inband is set to never
Interconnect choices are differentiated for FXO and FXS ports.

Bug Fixes

Correct issue where Interconnect configuration did not occur over VLAN connections
If Outbound Proxy is configured, use it to match peers even if Host is resolvable.
Do not strip leading 0s from the Number Begins field in Calling Rules.

Version 2.4.2 - January 23, 2018

Improvements Made

Added Interconnect capability to allow simplified connection and configuration with a Switchvox system.
LibPRI updatesd to V1.4.15.
mDNS and Avahi services added.

Version 2.3.1 - April 29, 2015

Improvements Made

Set the Facility-based ISDN Supplementary Service to be active in the default configuration. (Insures name is passed for spans configured for QSIG switches by default)

Bug Fixes

Fix lock up problem that occurs when span events occur during call setup and tear down.
Fix the TLS Certificate Common name NULL byte exploit (Ref: AST-2015-003)

Version 2.3.0 - March 2, 2015

Improvements Made

TDM channels can now be recorded for troubleshooting purposes
GUI users can now be authenticated using Radius servers
Call Briding events can be sent out to syslog. A single line of text identifies both ends of the call.
Syslog Output can now be redirected to a USB stick. (Note: Enable Advanced Settings must be set to YES for this to work.)
SIP endpoint status is showing more information on the call ID session.
SIP user names can now include '*' and '.'

Bug Fixes

SIP Bind ports can now differ from 5060
NFAS groups are prevented from being assigned to the same physical spans due to editing the NFAS span names
Explicit Caller ID assignment in calling rule is reflected on outgoing calls when incoming is set to prohibited
Prohibited incoming Caller ID is no longer passed to T1/E1 span

Version 2.2.5 - December 18, 2014

Bug Fixes

Resolve issue where timing slips and e-bit errors were incurred due to problems with glocal clock configuration

Version 2.2.4 - October 27, 2014

Bug Fixes

Remove SSLv2 and SSLv3 from HTTPS negotiations in the graphical interface.
Fixed how SIP endpoint data is parsed in the Connection Status page.

Version 2.2.3.2 - September 18, 2014

Improvements Made

Added the ability to recover the clock on spans that are set in remote loopback.

Bug Fixes

Fixed bug that prevented using more than 49 analog channels (less than 3 spans) as a target for a calling rule.
Fixed issue where D4 Framing and AMI Coding did not work in CAS signaling.
Fixed a bug where DTMF detected before channel structures were initialized caused a system crash.
Debug logs of gateway functionality are no longer affected by how remote syslog is set.
Fixed a bug wehere E1 spans connected to SIP clients offering non-aLaw codecs in addition to aLaw were causing one way audio.
Cleaned up display during software updates so false errors would be flagged.
Remote loopback diagnostics indicates spans are in loopback even when they are not.
Remove SSL2 and SSL3 from TLS SIP signaling options. (They are not supported.)
Fix bug with binary image that prevented upgrade from v2.2.0 (October 2)

Version 2.2.1.1 - March 10, 2014

Improvements Made

Added more system information in the debug tarballs

Bug Fixes

Fixed issue where DOS attack could cause all available file descriptors to be used. (Ref: AST-2014-002)
Fixed a bug where the gateway would not function if an NFAS group were defined without spans
Fixed a bug where NFAS group name editing would cause a duplicate NFAS group to be defined

Version 2.2.0 - February 24, 2014

Improvements Made

Non-Facility Associated Signalling (NFAS) is now supported
Spans can be removed from the list of spans used to be considered for clock recovery.
Remote option pull via DHCP option 66 now made the default behavior
Timezone files have been updated to the 2013i version

Bug Fixes

Fixed a bug where E1 status showed spans to be up, active and in alarm when they actually were not in alarm.

Version 2.1.3.3 - February 11, 2014

Improvements Made

Dial Failover rules now drops the call when the user rejects the call. (The call does not fail over to other spans)
Feature access codes *,#,a,b,c,d are now passed to the PRI spans.

Bug Fixes

Jitter implementation for adaptive jitter buffer has been fixed
DTMF via SIP INFO messages no longer hangs with the DTMF tone.being active
SIP calls thet are set for only g729 transcoding no longer prematurely hangup.

Version 2.1.0.1 - November 21, 2013

Bug Fixes

Fix bug where recovery USB stick image did not work with G100/200

Version 2.1.0 - November 12, 2013

New Features

Migrate the G100/G200 with the G400/G800 including all the new capabilities from 2.0.1.1 and 2.0.2.1
Add the ability to set SIP endpoints to use an adaptive Jitter Buffer
Add the ability to recover the DID from the SIP To: header
Add gateway IP address to the connection status display
Provide the ability to refresh DNS for SIP endpoint addresses

Bug Fixes

Fix bugs in Connection Status display and access control lists.
Fixed bug where RTP port range was not able to be set away from the default.
Fixed problem where debug log would not save data if remote logging was enabled.
Fix problem where setting a calling rule trim digit to '0' caused a calling rule to fail.

Version 2.0.2.1 - August 7, 2013

Bug Fixes

G400 now can recover the clock from the incoming spans.
Fixed bug where IP configuration changes were not maintained over reboots.

Version 2.0.1.1 - July 10, 2013 (G400/G800 Only)

New Features

This software is the initial release of the G400/G800 gateways. The software contains the functionality of the G100/200 with the addition of:

Increased number of spans.
Dual physical Ethernet interfaces
VLAN Support (2 per physical interface).
Physical Reset Switch/GUI password reset.
SIP TLS support.
Ability to set spans into network line and network payload loopback states.
Backup restore while maintaining current network configuration.

Known Issues

Access Control Rules do not allow for disabling the Web interface.
T1/E1 span connection status may falsely report "Down, Active" even though it is operating correctly. A successful call over the span will clear the error.
T1/E1 span connection status clear error may not clear all error logs.
Ethernet Port may show in Connection Status even though it is disabled.

Version 1.1.7 - August 14, 2013 (G100/G200 Only)

Bugs Fixed

Fixed a bug that prevented the upgrade from 1.0.x firmware due to the inability to reset the DSP via command.
Increased the number of calling rules that code be assocviated with a call source from 128 to > 500.

Version 1.1.6.1 - July 29, 2013 (G100/G200 Only)

Bugs Fixed

Updated media processing with new fax processing stack to improve fax performance
Fixed bug in ordering of calling rules by insuring that numerical sort (not alphabetical) is being used.
Fixed bug where overlap dialing calls were being disconnected by adding Proceeding message
Fixed bug in syslog output to insure that messages are being output when remote syslog is disabled.

Version 1.1.5 - March 25, 2013 (G100/G200 Only)

Bugs Fixed

Fixed a crash where e&m calls caused the media processing to reset
Fixed e&m wink time default to be set at 150 mS. The range for wink times is now 50 to 300 mS
e&m media is now sent through to SIP endpoints before the call is connected. (allows for ringback to be heard on the SIP endpoints.)
Fixed a problem where the order of failovers was not maintained
Fixed upgrade that caused PRI dialplan prefixes to be properly displayed
Fixed upgrade issues that improperly imported e&m feature set trunks

Version 1.1.3 - December 12, 2012 (G100/G200 Only)

Bugs Fixed

Fixed regression that prevented g729 to be used on SIP calls introduced in 1.1.0.3
DTMF Emulation has been disabled to prevent DTMF getting stuck on T1 to T1 calls
Sysinfo.txt is included in debug tarballs to assist in system troubleshooting.
Minor pulse mask tweaks to better fit profile.

Version 1.1.2 - November 19, 2012 (G100/G200 Only)

Bugs Fixed

Fix deadlock where simultaneous SIP calls could lock up SIP processing for subsequent dialogs.
PRI Dial Plan for Dialed Number are properly displayed in the GUI.
Deleting an Administrator is immediately committed to the configuration archive and survives rebooting.
Adding 10+ failovers to calling rules no longer causes system slowdown.

Version 1.1.1 - October 19, 2012 (G100/G200 Only)

Known Issues

Deleting an Administrator may not be saved to the configuration archive. A reset of the unit may cause the user to reappear. A workaround is to save another configuration element after deleting the administrator.

Bugs Fixed

Fixed bug where caller ID was being turned off on upgrades from earlier versions.
NOTE: Users who have already upgraded to version 1.1.0.3 users must manually re-enable caller ID from the T1/E1 Port -> Advanced Signaling tab.
SIP setting "shrink caller id" shows actual setting instead of just the default value.

Version 1.1.0.3 - October 10, 2012 (G100/G200 Only)

New Features

Analog CAS channel allocation
The gateway now supports mixed analog signaling on a single span
Added the ability to change DID number prefixes.
T.38 FAX now can traverse NATs.
The GUI API has been improved.
Firefox and Chrome browser interoperability has been improved.
FAX ECM and TCF options have been added to FAX Settings for SIP Endpoints
Enhanced Syslog capabilities added.
Patloop test added for diagnostics
Debug and Back UP file names have been changed to include user readable date information.
Temperature sensor information is now displayed in the GUI.
Number of tone locales has been increased.
span error information has been added to T1/E1 Connection Status
Upgrade to use asterisk V1.8.11

Bugs Fixed

E-M modes have been fixed.
Call routing rules does not match dialed number length correctly.
Network configuration is now updated after a restore of a backup configuration
Asymetric DTMF payload types is supported
T.17 training failures on some fax machines is now supported with local TCF.
Caller ID is now supported in analog signaling modes.
Fixed bug where setting manual time causes timeout in GUI sessions.

Version 1.0.5.3 - July 19, 2012 (G100/G200 Only)

Bugs Fixed

Configurations are now archived after a database upgrade.
Allow for configuring 2 endpoints going to the same asterisk server when the gateway registers to the asterisk server
When stopping a debug capture then rebooting the box, the capture appears on the gui.
Syslog turns off when disabled in the GUI.
Switching from manual to default NTP settings restores the original NTP server.
Call Routing Rules can be moved from one page to another
The gateway provides for a destination number when the gateway registers to an endpoint.
The bug that improperly set tones for a non-US Locale has been fixed.

Improvements Made

Allow for SIP authentication based solely on IP address through the option "Authenticate Incoming Calls".
This is set in the SIP Endpoints page under Call Settings -> Advanced Signaling Settings. (This requires Advanced Options to be enabled.)

Version 1.0.4.2 - June 25, 2012 (G100/G200 Only)

Bugs Fixed

CallerID number is set to anonymous for calls received with callerID name set as restricted
Faxes and modem calls are failing when being routed from one T1 port to the other.
One way audio on voice calls
Span Yellow Alarms not being handled correctly when determining span clock sources

Version 1.0.3.2 - June 4, 2012 (G100/G200 Only)

Bugs Fixed

Can not access downloads.digium.com for software update when the configured DNS is not a public DNS
Gateway appears to not be sending early media (ringing) from SIP provider on through to PRI
Outbound/Inbound DTMF fails in some cases
One-way audio starting at the first SIP re-invite
Inbound caller ID name is only displaying as phone number, or as Anonymous
T.38 Faxing fails
Unable to change the UDP/TCP port number on an endpoint that is not configured to register
Sporadic one-way audio issue; PSTN (external) side cannot hear the SIP (internal) side
Setting a port for E1, then enabling CRC4, then setting it back for T1 leaves CRC4 enabled
Switching Timing order of ports does not take effect until a system restart
SIP Remote Party-ID settings are not activated
Document behavior of booting up automatically after being shutdown on the Maintenance -> System reset page
Fix a few mistakes in help pages

Improvements Made

Debug logging now includes DTMF messages
Increase TCP traffic maximum file size on debug sessions from 8MB to 16MB.