Denis at mobile.
Begin forwarded message: > From: Asterisk Development Team <asteriskt...@digium.com> > Date: 27 de janeiro de 2012 17:10:00 BRST > To: asterisk-...@lists.digium.com > Subject: [asterisk-dev] Asterisk 10.1.0 Now Available > Reply-To: Asterisk Developers Mailing List <asterisk-...@lists.digium.com> > > The Asterisk Development Team is pleased to announce the release of > Asterisk 10.1.0. This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/ > > The release of Asterisk 10.1.0 resolves several issues reported by the > community and would have not been possible without your participation. > Thank you! > > The following is a sample of the issues resolved in this release: > > * AST-2012-001: prevent crash when an SDP offer > is received with an encrypted video stream when support for video > is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) > Reported by: Catalin Sanda > > * Allow playback of formats that don't support seeking. ast_streamfile > previously did unconditional seeking on files that broke playback of > formats that don't support that functionality. This patch avoids the > seek that was causing the problem. > (closes issue ASTERISK-18994) Patched by: Timo Teras > > * Add pjmedia probation concepts to res_rtp_asterisk's learning mode. In > order to better handle RTP sources with strictrtp enabled (which is the > default setting in 10) using the learning mode to figure out new sources > when they change is handled by checking for a number of consecutive (by > sequence number) packets received to an rtp struct based on a new > configurable value called 'probation'. Also, during learning mode instead > of liberally accepting all packets received, we now reject packets until a > clear source has been determined. > > * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing > to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop > causes the loop to exit prematurely. This causes a variety of negative side > effects, depending on when the loop exits. This patch handles the frame by > essentially swallowing the frame in the local loop, as the current channel > drivers expect the RTP bridge to handle the frame, and, in the case of the > local bridge loop, no additional action is necessary. > (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested > by: Matt Jordan > > * Fix timing source dependency issues with MOH. Prior to this patch, > res_musiconhold existed at the same module priority level as the timing > sources that it depends on. This would cause a problem when music on > hold was reloaded, as the timing source could be changed after > res_musiconhold was processed. This patch adds a new module priority > level, AST_MODPRI_TIMING, that the various timing modules are now loaded > at. This now occurs before loading other resource modules, such > that the timing source is guaranteed to be set prior to resolving > the timing source dependencies. > (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, > Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont > Patched by elguero > > * Fix RTP reference leak. If a blind transfer were initiated using a > REFER without a prior reINVITE to place the call on hold, AND if Asterisk > were sending RTCP reports, then there was a reference leak for the > RTP instance of the transferrer. > (closes issue ASTERISK-19192) Reported by: Tyuta Vitali > > * Fix blind transfers from failing if an 'h' extension > is present. This prevents the 'h' extension from being run on the > transferee channel when it is transferred via a native transfer > mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported > by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by > Mark Michelson (license 5049) > > * Restore call progress code for analog ports. Extracting sig_analog > from chan_dahdi lost call progress detection functionality. Fix > analog ports from considering a call answered immediately after > dialing has completed if the callprogress option is enabled. > (closes issue ASTERISK-18841) > Reported by: Richard Miller Patched by Richard Miller > > * Fix regression that 'rtp/rtcp set debup ip' only works when a port > was also specified. > (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: > Walter Doekes > > For a full list of changes in this release candidate, please see the > ChangeLog: > > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0 > > Thank you for your continued support of Asterisk! > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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