asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	f248f6eb71	Fix error that caused truncate operations to fail 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	13 years ago
Matthew Jordan	8cbe9f9fa7	Fix error that caused seek format operations to set max file size to '1' or '0' 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	13 years ago
Matthew Jordan	b3a38a51c7	Check for IO stream failures in various format's truncate/seek operations 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	13 years ago
Leif Madsen	d4938a111e	Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	6fba8e4fff	Formats need to load before apps, because some apps call ast_format_str_reduce() at load time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	8ab22f5dd0	Set a module load priority for format modules. 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	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	39e730b4e2	Remove useless debugging message. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2a53f2ec98	Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) 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	16 years ago

10 Commits (1.8)