Release Notes for 3.1.3 -- 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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IMPORTANT NOTICE:  
- This and all future releases DO NOT allow firmware downgrade 
  to software versions prior to 1.0.30.
- Starting from release 3.1.3, the (NT) feature lable is removed from
  the firmware version string. All the NT specific features are
  included in 3.1.3 and later releases
   

Feature Enhancement
===================================
-- since 3.1.2 --

1.  Support external conference bridge for n-way conference calls 
    (where n > 2) instead of mixing audio locally. To enable this feature, configure the
    following 2 new [Line 1/2] paramters:
    <Conference Bridge URL> : Such as "conf@myservie.com:12345", or "conf" (use the < Proxy >
                             as the domain). Maximum length is 79 characters. Default is
                             blank which disables this feature
    <Conference Bridge Ports> : Maximum number of conference participants. Default is 3.
    
2.  New [Line 1/2] parameter < SIP Proxy-Require >.

3.  New [Line 1/2] parameter < SIP Remote-Party-ID >. 

4.  Support parsing of SIP P-Asserted-Identity header in inbound INVITE and 200 response to INVITE.

5.  New [SIP] paramater < Stats In BYE >. This is a boolean parameter. If set to "yes", 
    the unit will include P-RTP-Stat header a BYE or response to a BYE message. 
    Default setting is "no".
    This header contains RTP statistics of the current call. The format of this header is 
    (mirrors MGCP connection paramters):
       P-RTP-State:  PS=< packets sent >,OS=< octets sent >,PR=< packets received >,OR=< octets received >,
                     PL=< packets lost >,JI=< jitter in ms >,LA=< delay in ms >,DU=< call duration in s >,
		     EN=< encoder>,DE=< decoder>
    NOTES: 		
    - <encoder> and  can be one of the following names:
      G711u,G711a,G729a,G723,G726-16,G726-32,G726-24,G726-40
    - delay value is valid only if both end-points support RTCP. Otherwise the
      delay will be 0.

6.  New [SIP] parameter <Escape Display Name>. This is a boolean parameter. If set to
    "yes", the unit will enclosed the string configured in <Dislay Name> in  pair of double 
    quotes in outbound SIP messages. Any occurences of " or \ in the string will be escaped 
    with \" and \\ inside the pair of double quotes. Default setting is "no"


7.  New [Line 1/2] parameter < Refer-To Target Contact >. This is a boolean parameter.
    If set to "yes", the unit, as the refeor, will use the refer target's Contact in 
    the Refer-To header when sending out a REFER request to the referee during a
    attended call transfer operation. If the option is "no", the well-known address 
    of the target is used. Default value is "no".

8.  New [Line 1/2] parameters to fine control call transfer operations:
    < Referor Bye Delay >, < Referee BYE Delay > , .
    All values are in seconds

9.  Allow an optional profile rule parameter in NOTIFY event header for resync.
    Format: Event: resync[;profile=< profile-rule >]
    where <profile-rule> follows the syntax of a standard profile fule.
    Example: Event: resync;profile=tftp://tftpserver/$MAC.cfg
    NOTES: 
    - %xx escape is allowed in the <profile-rule> parameter.   
    - quoted-string is not allowed for <profile-rule> parameter

10. Added < Sticky 183 > boolean paramter under Line 1/2 tap. If set to
    "yes", SPA will ignore further 180 SIP responses after receiving the 
    first 183 SIP response for an outbound INVITE. 

11  Extract caller name or number from the From header if P-Asserted-Identity
    blocks caller name or number

12. Allocate a fresh codec list if inbound reINVITE codec list mismatches
    current list

13. Reply 481 to in-dialog SIP OPTIONS requests if no matching dialog
    found