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2012-05-14Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)Kinsey Moore
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10Run predial routine on local;2 channel where you would expect.Richard Mudgett
Before this patch, the predial routine executes on the ;1 channel of a local channel pair. Executing predial on the ;1 channel of a local channel pair is of limited utility. Any channel variables set by the predial routine executing on the ;1 channel will not be available when the local channel executes dialplan on the ;2 channel. * Create ast_pre_call() and an associated pre_call() technology callback to handle running the predial routine. If a channel technology does not provide the callback, the predial routine is simply run on the channel. Review: https://reviewboard.asterisk.org/r/1903/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28res_corosync: Fix build against corosync 2.0.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-27Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machinesMatthew Jordan
The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ ........ Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364285 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Make it possible to change the minimum DTMF duration in asterisk.confOlle Johansson
Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Fix connected-line/redirecting interception gosubs executing more than intended.Richard Mudgett
* Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so execution will stop after the routine returns there. (s@gosub_virtual_context:1) * Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and gosub application respectively with the parameter string already created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Add original party id and reason support.Richard Mudgett
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who was the original redirecting party of a call. * Added support for the original redirecting party and reason to the REDIRECTING function and the system core as well as to the stubbed locations in sig_pri.c. Review: https://reviewboard.asterisk.org/r/1829/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16Add IPv6 address support to security events framework.Michael L. Young
The current Security Events Framework API only supports IPv4 when it comes to generating security events. This patch does the following: * Changes the Security Events Framework API to support IPV6 and updates the components that use this API. * Eliminates an error message that was being generated since the current implementation was treating an IPv6 socket address as if it was IPv4. * Some copyright dates were updated on files touched by this patch. (closes issue ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael L. Young Patches: security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12Add option to invoke the extensions.conf stdexten using the legacy macro method.Richard Mudgett
ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in favor of the Gosub method without a means of backwards compatibility. (issue ASTERISK-18809) (closes issue ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1855/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03Fix dev-mode compiler warning about gnu_printfMark Murawki
(related to ASTERISK-19575) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03Allow the Hangup manager action to match channels by regexMark Murawki
* Hangup now can take a regular expression as the Channel option. If you want to hangup multiple channels, use /regex/ as the Channel option. Existing behavior to hanging up a single channel is unchanged, but if you pass a regex, the manager will send you a list of channels back that were hung up. (closes issue ASTERISK-19575) Reported by: Mark Murawski Tested by: Mark Murawski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Misc changes to make astobj2 enhancement diffs easier to follow.Richard Mudgett
* Rename astobj2 API parameter funcname to func. * Rename astobj2 API iterator parameter to iter. * Update some documentation for OBJ_MULTIPLE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Introducing the log message unique call identifiers featureJonathan Rose
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29undoing 360785 due to merging mistakeJonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Introducing the log message unique call identifiers featureJonathan Rose
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-27Add global ao2 array container.Richard Mudgett
Global ao2 objects must always exist after initialization because there is no access control to obtain another reference to the global object. It is expected that module configuration could use these new API calls to replace an active configuration parameter object with an updated configuration parameter object. With these new API calls, the global object could be replaced, removed, or referenced without the risk of someone using a stale global object pointer. Review: https://reviewboard.asterisk.org/r/1824/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-22Adds F option to Bridge applicationJonathan Rose
Similar to dial and queue F option. (Closes issue ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff uploaded by To (license 6347) Review: https://reviewboard.asterisk.org/r/1825/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-20chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus AMI EventsSean Bright
The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. Post was removed and the AMI version has been updated to 1.3. ........ Merged revisions 359982 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-20Allow AMI action callback to be reentrant.Richard Mudgett
Fix AMI module reload deadlock regression from ASTERISK-18479 when it tried to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The patch stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ Review: https://reviewboard.asterisk.org/r/1820/ ........ Merged revisions 359979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359980 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-16Simplify some code in ast_app_run_sub().Richard Mudgett
* Remove unnnecessary const from const char * const var declaration in the ast_app_run_macro() and ast_app_run_sub() prototypes. The second const is unnecessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-16Revert the pre-dial addition.Mark Michelson
The code may be just fine, but it had not received a "ship it!" on review board yet. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-15Add options PreDial options 'b' and 'B' to app_dialMark Murawki
* Added 'b' and 'B' options to Dial. These options will allow you to run last-minute dialplan on the caller and callee channels while the Dial application is executing, but before the call is started. For example you can use the 'b' option to run dialplan on the callee channel to get the name of the newly created channel right away. Review: https://reviewboard.asterisk.org/r/1229/ (closes issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark Murawski, Stefan Schmidt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14app.h: Always initialize AST_DECLARE_APP_ARGS().Russell Bryant
This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always fully initialized. I'm not sure if this fixes any real bugs, but it silences a bunch of warnings from coverity, and is generally a good thing to do anyway. ........ Merged revisions 359452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359454 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14Fix deadlock potential with some ast_indicate/ast_indicate_data calls.Richard Mudgett
Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. ........ Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359453 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14Three copies of the file contents in channel_internal.h are a bit excessive.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14Force non-inlining of ao2_iterator_destroy when TEST_FRAMEWORK is enabledMatthew Jordan
In r357272, astobj2 was changed to automatically enable REF_DEBUG when the TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers (gcc 4.5.1 at least) will attempt to inline ao2_iterator_destroy in handle_astobj2_test. This by itself is not a problem; unfortunately, the compiler believes that there is a code path wherein an object allocated on the stack will be free'd. As warnings are treated as errors, this prevents compilation of astobj2. This patch works around that by adding the noinline attribue to ao2_iterator_destroy, but only if the TEST_FRAMEWORK flag is enabled. Preventing inlining is only needed for the test method defined in astobj2, which is also only enabled if TEST_FRAMEWORK is enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14Fix bogus reads/writes of console log levels in asterisk.c Russell Bryant
This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. ........ Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359260 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Remove chan_usbradio and app_rpt.Russell Bryant
These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged revisions 359050 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359051 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Add missing channel_internal.hTerry Wilson
...again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Make hints for invalid SIP devices return Unavail, not idleTerry Wilson
This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged revisions 358943 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358944 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Finalize ast_channel opaquificationTerry Wilson
Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-10Transition app_page to using app_confbridge internally for the conference ↵Joshua Colp
bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles. Review: https://reviewboard.asterisk.org/r/1754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-08Add some underscores in a few of our llist macros to reduce name collisions.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-05Make usage of DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-02Fix case-sensitivity for device-specific event subscriptions and CCSSKinsey Moore
This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ ........ Merged revisions 357940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357941 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-01Opaquify ast_channel typedefs, fd arrays, and softhangup flagTerry Wilson
Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29Update stringfield documentation for removed second va_list in favor of va_copy.Walter Doekes
In r320946, the second va_list that was passed to ast_string_field_build_va and friends, was removed. This patch updates the documentation to reflect that. ........ Merged revisions 357620 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29Opaquify ast_channel structs and listsTerry Wilson
Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28Documentation update. There is no AST_SOCKADDR_UNSPEC.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28Convert struct ast_tcptls_session_instance to finally use the ao2 object lock.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28Astobj2 locking enhancement.Richard Mudgett
Add the ability to specify what kind of locking an ao2 object has when it is allocated. The locking could be one of: MUTEX, RWLOCK, or none. New API: ao2_t_alloc_options() ao2_alloc_options() ao2_t_container_alloc_options() ao2_container_alloc_options() ao2_rdlock() ao2_wrlock() ao2_tryrdlock() ao2_trywrlock() The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning change. They no longer mean that the object is protected by an external mechanism. They mean the lock associated with the object has already been manually obtained by one of the ao2_lock calls. This change is necessary for RWLOCK support since they are not reentrant. Also an operation on an ao2 container may require promoting a read lock to a write lock by releasing the already held read lock to re-acquire as a write lock. Replaced API calls: ao2_t_link_nolock() ao2_link_nolock() ao2_t_unlink_nolock() ao2_unlink_nolock() with the respective ao2_t_link_flags() ao2_link_flags() ao2_t_unlink_flags() ao2_unlink_flags() API calls to be more flexible and to allow an anticipated enhancement to control linking duplicate objects into a container. The changes to format.c and format_cap.c are taking advantange of the new ao2 locking options to simplify the use of the format capabilities containers. Review: https://reviewboard.asterisk.org/r/1554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28Add ability to clone ao2 containers.Richard Mudgett
Occasionally there is a need to put all objects in one container also into another container. Some reasons you might need to do this: 1) You need to reconfigure a container. You would do this by creating a new container with the new configuration and ao2_container_dup the old container into it. Then replace the old container with the new. Then destroy the old container. 2) You need the contents of a container to remain stable while operating on all of the objects. You would do this by creating a cloned container of the original with ao2_container_clone. The cloned container is a snapshot of the objects at the time of the cloning. When done, just destroy the cloned container. Review: https://reviewboard.asterisk.org/r/1746/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Converts locking for odbc containers from ast_mutex_lock to ao2_locks.Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Deprecated macro usage for connected line, redirecting, and CCSSKinsey Moore
This commit adds GoSub alternatives to connected line, redirecting, and CCSS macro hooks so that macro can finally be deprecated. This also adds deprecation warnings for those features when used and in documentation. Review: https://reviewboard.asterisk.org/r/1760/ (closes issue SWP-4256) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Convert netsock.h over to use ast_sockaddrs rather than sockaddr_in and updateSean Bright
chan_iax2 to pass in the correct types. chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk at this point, so this feels like a safe change to make. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Make ast_netsock_set_qos() delegate to ast_set_qos().Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Correct typo in deprecation comment.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Prefer ast_set_qos() over ast_netsock_set_qos()Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24astobj2.h comment tweaks.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24astobj2.h documentation updates.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-04 12:35:28 +0000