Another very inappropriate placement of a ')' (again introduced in r362151)
caused the various truncate operations to attempt to truncate the sound file
at a position of '0'.
(issue ASTERISK-19655)
Reported by: Matt Jordan
(issue ASTERISK-19810)
Reported by: colbec
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A very inappropriate placement of a ')' (introduced in r362151) caused the
maximum size of a file to be set as the result of a comparison operation, as
opposed to the result of the ftello operation. This resulted in seeking being
restricted to the beginning of the file, or 1 byte into the file. Thanks to
the Asterisk Test Suite for properly freaking out about this on at least one
test.
(issue ASTERISK-19655)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the formats that support seek and/or truncate operations, many of
the C library calls used to determine or set the current position indicator
in the file stream were not being checked. In some situations, if an error
occurred, a negative value would be returned from the library call. This
could then be interpreted inappropriately as positional data.
This patch checks the return values from these library calls before
using them in subsequent operations.
(issue ASTERISK-19655)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1863/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent change to app_voicemail made it such that the module now assumes that
all format modules are available while processing voicemail configuration.
However, when autoloading modules, it was possible that app_voicemail was
loaded before the format modules. Since format modules don't depend on
anything, set a module load priority on them to ensure that they get loaded
first when autoloading.
This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will
require a different approach since the module load priority functionality is
not present in the module API.
(issue #16412)
Reported by: jiddings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
