Release Notes - Digium Gateways G100/G200

Version 1.1.7 - August 14, 2013

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.

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.

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

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.

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.

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.

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.

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

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.

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.)

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

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