<html><head></head><body bgcolor="#FFFFFF"><div><br><br>Denis at mobile.</div><div><br>Begin forwarded message:<br><br></div><blockquote type="cite"><div><b>From:</b> Asterisk Development Team <<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>><br><b>Date:</b> 27 de janeiro de 2012 17:10:00 BRST<br><b>To:</b> <a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a><br><b>Subject:</b> <b>[asterisk-dev] Asterisk 1.8.9.0 Now Available</b><br><b>Reply-To:</b> Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br><br></div></blockquote><div></div><blockquote type="cite"><div><span>The Asterisk Development Team is pleased to announce the release of</span><br><span>Asterisk 1.8.9.0. This release is available for immediate download at</span><br><span><a href="http://downloads.asterisk.org/pub/telephony/asterisk/">http://downloads.asterisk.org/pub/telephony/asterisk/</a></span><br><span></span><br><span>The release of Asterisk 1.8.9.0 resolves several issues reported by the</span><br><span>community and would have not been possible without your participation.</span><br><span>Thank you!</span><br><span></span><br><span>The following is a sample of the issues resolved in this release:</span><br><span></span><br><span>* AST-2012-001: prevent crash when an SDP offer</span><br><span> is received with an encrypted video stream when support for video</span><br><span> is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)</span><br><span> Reported by: Catalin Sanda</span><br><span></span><br><span>* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing</span><br><span> to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop</span><br><span> causes the loop to exit prematurely. This causes a variety of negative side</span><br><span> effects, depending on when the loop exits. This patch handles the frame by</span><br><span> essentially swallowing the frame in the local loop, as the current channel</span><br><span> drivers expect the RTP bridge to handle the frame, and, in the case of the</span><br><span> local bridge loop, no additional action is necessary.</span><br><span> (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested</span><br><span> by: Matt Jordan</span><br><span></span><br><span>* Fix timing source dependency issues with MOH. Prior to this patch,</span><br><span> res_musiconhold existed at the same module priority level as the timing</span><br><span> sources that it depends on. This would cause a problem when music on </span><br><span> hold was reloaded, as the timing source could be changed after</span><br><span> res_musiconhold was processed. This patch adds a new module priority</span><br><span> level, AST_MODPRI_TIMING, that the various timing modules are now loaded</span><br><span> at. This now occurs before loading other resource modules, such</span><br><span> that the timing source is guaranteed to be set prior to resolving</span><br><span> the timing source dependencies. </span><br><span> (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,</span><br><span> Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont</span><br><span> Patched by elguero</span><br><span></span><br><span>* Fix RTP reference leak. If a blind transfer were initiated using a </span><br><span> REFER without a prior reINVITE to place the call on hold, AND if Asterisk</span><br><span> were sending RTCP reports, then there was a reference leak for the </span><br><span> RTP instance of the transferrer.</span><br><span> (closes issue ASTERISK-19192) Reported by: Tyuta Vitali</span><br><span></span><br><span>* Fix blind transfers from failing if an 'h' extension</span><br><span> is present. This prevents the 'h' extension from being run on the</span><br><span> transferee channel when it is transferred via a native transfer</span><br><span> mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported</span><br><span> by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by</span><br><span> Mark Michelson (license 5049)</span><br><span></span><br><span>* Restore call progress code for analog ports. Extracting sig_analog</span><br><span> from chan_dahdi lost call progress detection functionality. Fix </span><br><span> analog ports from considering a call answered immediately after </span><br><span> dialing has completed if the callprogress option is enabled. </span><br><span> (closes issue ASTERISK-18841)</span><br><span> Reported by: Richard Miller Patched by Richard Miller</span><br><span></span><br><span>* Fix regression that 'rtp/rtcp set debup ip' only works when a port</span><br><span> was also specified. </span><br><span> (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:</span><br><span> Walter Doekes</span><br><span></span><br><span>For a full list of changes in this release candidate, please see the ChangeLog:</span><br><span></span><br><span><a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0</a></span><br><span></span><br><span>Thank you for your continued support of Asterisk!</span><br><span></span><br><span></span><br><span>--</span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br><span></span><br><span>asterisk-dev mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a></span><br></div></blockquote></body></html>