Release Notes for -- Linksys SPA-3000 3.1.20(GW) 3000 -- 1 Port FXS, 1 Port FXO, 1 Ethernet Interface Copyright (C) 2007 by Linksys, a Division of Cisco Systems, Inc. All Rights Reserved. * * * * * * * * * * IMPORTANT * * * * * * * * * * * * * * * * * * Use of Proprietary Information and Copyright Notice: * * This release note document contains proprietary information * * that is to be used only by Sipura Technology, Linksys(R), * * and Cisco Systems, Inc. customers. Any unauthorized * * disclosure, copying, distribution, or use of this * * information is prohibited. This restriction includes * * ALL Internet based discussion forums, e.g. DSLreports. * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * /** NOTICE **/ official release ======================================== New Features ======================================== ### Since 3.1.20(GW) - Removed Polarity Reversal from the "DTMF(Denmark)" caller-id method. Instead added a new "DTMF(Denmark) With PR" caller-id method that behaves the same as the old "DTMF(Denmark)" method. - dtmf tx mode: strict and normalfor pap2t, the minimum duration for detection is as follows:AVT strict mode: 70 msAVT normal mode: 30 msSIP info strict mode: 80 msSIP info normal mode: 30 ms - Added option. If set to "yes", all (redundant encoded) AVT tone pacekts will have the Marker bit set; otherwise only the first packet of each DTMF digit is set. Default is "yes". - If REGISTER results in a 301 response with a Contact header that has a maddr URI parameter, and if the is an IP address, the SPA will change the outproxy proxy address to the value of the maddr address. This value will remain valid until the next 301 response, if any, or will restore to the originally configured value upon reboot. - When swapping calls in a call-waiting or similar situation, the SPA will order the operations to make sure that call hold is invoked before call resume. ### Since 3.1.18-GW - Avoid reboot following resync if only changes involve Syslog_Server, Debug_Server, Debug_Level. - XML configuration profiles can now specify parameter values using 'value' attributes in empty tags, instead of enclosing the value within start and end tags. Example: ======================================== Bug Fixes ======================================== - Fixed this problem: Unit stops playing incoming RTP audio once it receives a 180/183 w/o SDP; regardless setting of parameter - Fixed this problem: Unit does not use the same Authorization header fields in the ACK as in the corresponding INVITE, per RFC 3261 - Fixed CSCsh44588: ACK uses different nc (nonce count) from INVITE if 100rel (PRACK) is enabled. - SDP in INVITE response, includes annexb=yes/no if present in INVITE message - Fixed this problem: A,B,C,D digits are dropped when dialing - SPA should escape occurrences of '#' in the dial string with %23 in outbound INVITE - bug fix: Bogus IP may be used if dns srv failed when querying the proxy server. ### Since 3.1.18-GW - avt marker bit incorrectly inserted in every avt rtp packet, should only appear in the first packet - Fixed this problem: nonce count not incremented when reINVITE to hold/resume inside a dialog - Fixed this problem: If first host in route has maddr parameter specified as host name instead of IP address, we will ignore it. - Fixed this problem: Object Statistics debug msg contains garbage letters for the subscription dialog object pool - Cold reboot on resync if Primary or Seconday DNS params changed. - Added config option. Default is yes. If set to "yes", unit will include c=0.0.0.0 syntax in SDP when reINVITE to hold. If set to "no", unit will not include c=0.0.0.0. syntax in SDP when reINVITE to hold. It will always include a=sendonly syntax when reINVITE to hold - If request uid not matching an exisiting server transaction's request uid, should try to match it against locally configured user id. If it matches, should still accept that request as belong to the same server transaction. - Include "replaces" in Supported header in outbound SIP messages and accept "replaces" in Require header in inbound SIP messages - Fixed this bug: NOTIFY request for refer event during call transfer does not include subscription-state header that is required per RFC 3265 - Fixed this bug: SPA does not include RTP-Stat header in BYE after the BYE is challenged - Fixed this bug: When unit fails over to secondary proxy during INVITE, the correspnding ACK might be sent to the primary if expires between the INVITE and the ACK. -- since 3.1.7 -- . Fixed URL/IP address configured in speed dial problem . If inbound INVITE does not have contact header, SPA will send 400, but followed by 100, and 180 which is not correct