Release Notes for 2.0.13(g) -- Sipura Phone Adapter

SPA-1000 -- 1 Port FXS, 1 Ethernet Interface
SPA-2000 -- 2 Port FXS, 1 Ethernet Interface

Copyright (C) 2003-2005 Sipura Technology Inc.

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* Use of Proprietary Information and Copyright Notice:         *
* This release note document contains proprietary information  *
* that is to be used only by Sipura Technology customers.      *
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* of this information is prohibited. This restriction includes * 
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IMPORTANT NOTICE:  
- This and all future releases DO NOT allow firmware downgrade 
  to software versions prior to 1.0.30.

Bug Fixes
==================================
 . SPA to SPA call may stuck in playing back out-of-band DTMF tones, or
   lost audio both-ways when both ends press DTMF at the same time. Hook flash
   twice on either end can restore normal audio.

 . URI parameter in Authorization header may not matching request uri.

 . SPA ignores STUN responses if the message is larger than 64 bytes

 . OOB DTMF via SIP INFO still transmitted when call is on hold;
   result is that holding peer can hear DTMF as caller dialed 
   3rd party in a 3-way call.

 . SPA does not include all available codecs when transferred; it only includes a 
   trimmed down set as a result of the original call establishment. 

 . SPA does not support compact form SIP EVENT header 'o'. 

 . SPA does not support changes to SDP in successive 183 responses.


Feature Enhancements
===================================
 . Change factory default <Time Zone> to GMT-8:00

 . Increased max allowed SIP URL parameter length to 318 characters
   in To, From, Contact, Route, and Refer-To headers.

 . Increased max allowed SIP URL user-id length to 142 characters
   in To, From, Contact, Route, and Refer-To headers.

 . Support RTP keep alive at the interval specified in <NAT Keep Alive Intvl>
   and is enabled if <NAT Keep Alive Enable> is "yes". 
 
 . Accept up to 199 chars of Call-ID and Branch values in inbound SIP messages.

 . Support <No UDP Checksum> SIP option for outbound  RTP packets