Il giorno 27 gennaio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 18.104.22.168
Dal post originale:
The release of Asterisk 22.214.171.124 resolves several issues reported by the
community and would have not been possible without your participation.
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
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:
For a full list of changes in this release candidate, please see the ChangeLog: