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2014-03-12app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up ↵Richard Mudgett
while MOH is playing. When MOH is playing to a user in a conference and the user is kicked or hangs up from the conference then the AMI MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event: MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported by: Benjamin Keith Ford Review: https://reviewboard.asterisk.org/r/3306/ ........ Merged revisions 410490 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410491 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-12res_pjsip_multihomed: Fix a bug where outgoing messages for TCP would go out ↵Joshua Colp
using UDP. This change fixes a bug where the code which changes the transport did not check whether the message is going out over UDP or not before changing it. For TCP and TLS transports we don't need to change the transport as the correct one is already chosen. ........ Merged revisions 410471 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-11res_pjsip_multihomed: Add module which places the correct address within ↵Joshua Colp
messages. Due to how messages are handled within PJSIP it is not until a message is actually sent that the destination is reliably known. This means that the addresses placed within the message may not be of the interface the message is being sent out on. This module determines what interface a message is being sent on and updates the message to contain the correct address if applicable. This module was tested by myself in a virtualized environment with multiple interfaces and also by Kinsey Moore in the following configuration: Networks: * 10.24.16.0/21 ** hard phone ** default gateway * 10.24.64.0/21 ** softphone with pjsip-based stack Transport details: bind address: 0.0.0.0 protocol: UDP All endpoints were tested with explicitly configured transports and unconfigured transports. This was tested with inbound and outbound calls, both of which were experiencing detrimental effects from incorrect IP addresses in SIP messages. These effects were only experienced by the soft phone on the 10.24.64.0 network since the messages to the hard phone on the 10.24.16.0 network had the correct IP address. (closes issue ASTERISK-23020) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3102/ ........ Merged revisions 410451 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10AST-2014-001: Stack overflow in HTTP processing of Cookie headers.Richard Mudgett
Sending a HTTP request that is handled by Asterisk with a large number of Cookie headers could overflow the stack. Another vulnerability along similar lines is any HTTP request with a ridiculous number of headers in the request could exhaust system memory. (closes issue ASTERISK-23340) Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10unqiueid: correct max uniqueid length testScott Griepentrog
This patch adds null string test prior to checking for a max uniqueid value that was added in r410157. ........ Merged revisions 410368 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10AST-2014-002: chan_sip: Exit early on bad session timers requestKinsey Moore
This change allows chan_sip to avoid creation of the channel and consumption of associated file descriptors altogether if the inbound request is going to be rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 410308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410311 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a ↵Joshua Colp
request will have an endpoint. This change removes the assumption that an outgoing request will always have an endpoint and makes the authenticate_qualify option work once again. (closes issue ASTERISK-23210) Reported by: Joshua Colp ........ Merged revisions 410306 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-08pjsip_cli: Create pjsip show channel and contact, and general cli code cleanup.George Joseph
Created the 'pjsip show channel' and 'pjsip show contact' commands. Refactored out the hated ast_hashtab. Replaced with ao2_container. Cleaned up function naming. Internal only, no public name changes. Cleaned up whitespace and brace formatting in cli code. Changed some NULL checking from "if"s to ast_asserts. Fixed some register/unregister ordering to reduce deadlock potential. Fixed ast_sip_location_add_contact where the 'name' buffer was too short. Fixed some self-assignment issues in res_pjsip_outbound_registration. (closes issue ASTERISK-23276) Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged revisions 410287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-08resource_channels: Check if a passed in ID is NULL before checking its lengthMatthew Jordan
Calling strlen on a NULL string is explosive. This patch checks whether or not the passed in string is NULL or zero length before checking to see if the string is too long. ........ Merged revisions 410274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07chan_sip: Fix deadlock of monlock between unload_module and do_monitorCorey Farrell
Release monlock before calling pthread_join. This ensures do_monitor cannot freeze by locking monlock during module unload. (closes issue ASTERISK-21406) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3284/ ........ Merged revisions 410224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410225 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410226 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07sorcery: correct field register argument listScott Griepentrog
This fixes mistakes I previously made in merging gtjoseph's changes with mine. ........ Merged revisions 410211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07config_options: Display the see-also information for CLI config option helpMatthew Jordan
The config option help information has always parsed the <see-also> tags in the XML documentation. Unfortunately, it just never bothered displaying them on the CLI. With this patch, when you execute 'config show help [module] [obj] [option]', it will display what other options are useful to you. (closes issue ASTERISK-22008) Reported by: Richard Mudgett ........ Merged revisions 410209 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07res_pjsip: Fix documentation for one touch recording see-also linksMatthew Jordan
The one touch recording options have several see-also links between the various configuration options. These were 'broken' by the snake casing of those options. This patch corrects the see-also links such that they reference the correct option names. ........ Merged revisions 410194 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07Make res_sorcery_realtime filter unknown retrieved results.Mark Michelson
When retrieving data from a database or other realtime backend, it's quite possible to retrieve variables that Asterisk does not care about but that are legitimate to exist. Asterisk does not need to throw a hissy fit when these variables are encountered but rather just filter them out. Review: https://reviewboard.asterisk.org/r/3305 ........ Merged revisions 410187 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07pjsip: allow and disallow show same codecsScott Griepentrog
In order to prevent confusion over the allow and disallow list of codecs being the same an option for registering a field as an alias is added. The alias field will be read from the configuration file, but afterwards is not listed as a known field. With disallow set as an alias, the CLI command pjsip show endpoint # will list the allow= field, but not the disallow field. (closes issue ASTERISK-23092) Review: https://reviewboard.asterisk.org/r/3193/ ........ Merged revisions 410190 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07stasis cache: Enhance to keep track of an item from different entities.Richard Mudgett
A stasis cache entry now contains more than a single message/snapshot. It contains messages/snapshots for the local entity as well as any remote entities that post to the cached item. In addition callbacks can be supplied when the cache is created to compute and post the aggregate message/snapshot representing all entities stored in the cache entry. * All stasis messages now have an eid to indicate what entity posted it. * The stasis cache enhancements allow device state to cache and aggregate the device states from local and remote entities in a single operation. The cached aggregate device state is available immediately after it is posted to the stasis bus. This improves performance by eliminating a cache dump and associated ao2 container traversals to calculate the aggregate state. (closes issue ASTERISK-23204) Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3281/ ........ Merged revisions 410184 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07uniqueid: Fix chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler ↵Richard Mudgett
errors. (issue ASTERISK-23120) ........ Merged revisions 410171 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07uniqueid: channel linkedid, ami, ari object creation with id'sScott Griepentrog
Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07chan_sip: Allow static realtime members to be qualified during module load.Matthew Jordan
When a static realtime peer with qualify=yes is loaded, Asterisk will fail to send an OPTIONS request due to the lastms being equal to 0. This results in the peer being unable to receive calls from Asterisk because the status is permanently UNKNOWN. This patch allows an OPTIONS request to be sent during module load by ignoring the lastms value on startup only. Review: https://reviewboard.asterisk.org/r/3294/ (closes issue ASTERISK-17523) Reported by: Maciej Krajewski Tested by: wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112) ........ Merged revisions 410105 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410106 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410107 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06sorcery.c: Fix off-nominal path ref and memory leak in ↵Richard Mudgett
ast_sorcery_objectset_json_create(). * Made exit a loop early on error in ast_sorcery_objectset_json_create(). * Removed some dead code in ast_sorcery_objectset_create2(). ........ Merged revisions 410089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06moh: fix a refcount error with realtime MOHRussell Bryant
I observed a crash in res_musiconhold on an Asterisk 11 system using realtime MOH. Investigation of the backtrace showed a corrupt mohclass, implying that it got destroyed before the code expected it to. I went looking for reference counting errors that could have caused this crash and this patch this result. It contains 2 changes. 1) Remove a usless block of code that was impossible to reach. There was even a comment indicating that it was impossible to reach. The conditional includes "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if block with the opposite check "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no good reason to keep it around. 2) A similar block to #1 contained a reference counting error. It stores state->class in the local variable mohclass without increasing its reference count. The reference count on mohclass is decremented at the end of the function. This block of code probably very rarely runs, which would help explain why this system was working fine for many months before experiencing a crash. Review: https://reviewboard.asterisk.org/r/3282/ ........ Merged revisions 410043 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410044 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410090 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06sorcery: Create AST_SORCERY dialplan function.George Joseph
This patch creates the AST_SORCERY dialplan function which allows someone to retrieve any value from a sorcery-based config file. It's similar to AST_CONFIG. The creation of the function itself was fairly straightforward but it required changes to the underlying sorcery infrastructure that rippled into individual sorcery objects. The changes stemmed from inconsistencies in how sorcery created ast_variable objectsets from sorcery objects and the inconsistency in how individual objects used that feature especially when it came to parameters that can be specified multiple times like contact in aor and match in identify. You can read more here... http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html So, what this patch does, besides actually creating the AST_SORCERY function, is the following... * Creates ast_variable_list_append which is a helper to append one ast_variable list to another. * Modifies the ast_sorcery_object_field_register functions to accept the already-defined sorcery_fields_handler callback. * Modifies ast_sorcery_objectset_create to accept a parameter indicating return type preference...a single ast_variable with all values concatenated or an ast_variable list with multiple entries. Also fixed a few bugs. * Modifies individual sorcery object implementations to use the new function definition of the ast_sorcery_object_field_register functions. * Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement sorcery_fields_handler handlers so they return multiple occurrences as an ast_variable_list. * Added a whole bunch of tests to test_sorcery. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06pjsip configuration: Make transport TOS values consistent with endpointsJonathan Rose
Transport TOS values were interpreted as DSCP values without being documented as such. Endpoint TOS values (tos_audio/tos_video) behaved normally as TOS values have historically. This patch makes the transport TOS values behave as TOS values and makes all TOS values readable as string values (e.g. AF11). In addition, alembic scripts have been updated to use the proper field types for all TOS/COS values. (issue ASTERISK-23235) Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3304/ ........ Merged revisions 410028 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06res_stasis_recording: Add a "target_uri" field to recording events.Joshua Colp
This change adds a target_uri field to the live recording object. It contains the URI of what is being recorded. (closes issue ASTERISK-23258) Reported by: Ben Merrills Review: https://reviewboard.asterisk.org/r/3299/ ........ Merged revisions 410025 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06Don't attempt to link in an aggregate MWI subscription if an endpoint does ↵Mark Michelson
not aggregate MWI. Attempting to link a NULL object into an ao2 container had been benign previously, but since enabling DO_CRASH in the testsuite, this is now causing a crash. It's better to be right here anyway. ........ Merged revisions 410011 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06Blocked revisions 410006George Joseph
........ sorcery: Create AST_SORCERY dialplan function. This patch creates the AST_SORCERY dialplan function which allows someone to retrieve any value from a sorcery-based config file. It's similar to AST_CONFIG. The creation of the function itself was fairly straightforward but it required changes to the underlying sorcery infrastructure that rippled into individual sorcery objects. The changes stemmed from inconsistencies in how sorcery created ast_variable objectsets from sorcery objects and the inconsistency in how individual objects used that feature especially when it came to parameters that can be specified multiple times like contact in aor and match in identify. You can read more here... http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html So, what this patch does, besides actually creating the AST_SORCERY function, is the following... * Creates ast_variable_list_append which is a helper to append one ast_variable list to another. * Modifies the ast_sorcery_object_field_register functions to accept the already-defined sorcery_fields_handler callback. * Modifies ast_sorcery_objectset_create to accept a parameter indicating return type preference...a single ast_variable with all values concatenated or an ast_variable list with multiple entries. Also fixed a few bugs. * Modifies individual sorcery object implementations to use the new function definition of the ast_sorcery_object_field_register functions. * Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement sorcery_fields_handler handlers so they return multiple occurrences as an ast_variable_list. * Added a whole bunch of tests to test_sorcery. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06res_fax_spandsp: Fix crash when passing ulaw/alaw data to spandspMatthew Jordan
When acting as a T.38 fax gateway, res_fax_spandsp would at times cause a crash in libspandsp. This would occur when, during fax tone detection, a ulaw/alaw frame would be passed to modem_connect_tones_rx. That particular routine expects the data to be in slin format. This patch looks at the frame type and, if the data is ulaw/alaw, converts the format to slin before passing it to modem_connect_tones_rx. Review: https://reviewboard.asterisk.org/r/3296 (closes issue ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal Rybarik patches: spandsp_g711decode.diff uploaded by Michal Rybarik (license 6578) ........ Merged revisions 409990 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409991 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06app_confbridge: Remove some noop code.Richard Mudgett
........ Merged revisions 409976 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06res_musiconhold.c: Remove some unnecessary RAII_VAR() usage.Richard Mudgett
* Made the moh_register() define use useful parameter names. ........ Merged revisions 409967 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05config: Fix inverted testKinsey Moore
The test of the result of the stat() call was inverted such that its output was only used if the call failed. This inverts the test so that the output of stat() is used correctly. This was causing full reloads on unchanged files. (closes issue ASTERISK-23383) Reported by: David Woolley ........ Merged revisions 409916 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409918 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05bridge_native_rtp: Fix crash involving masqueradeKinsey Moore
It is possible for a channel to be masqueraded out of a bridge which means it may no longer have RTP glue to check upon leaving said bridge. If this situation occurred (it's possible at least during dial and call pickup) then Asterisk would crash. This change makes sure the glue is checked before use. (closes issue AST-1290) Reported by: John Bigelow ........ Merged revisions 409900 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05alembic: Add missing queue and CDR table creation scripts.Richard Mudgett
* Added the queues and queue_members tables to the config alembic scripts. * Added the CDR table alembic creation script. The CDR table is more of an example for new setups since the actual table can be fully customized in cdr_adaptive_odbc.conf. (closes issue ASTERISK-23233) Reported by: jmls Review: https://reviewboard.asterisk.org/r/3227/ ........ Merged revisions 409885 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Fix documentation for PRESENCE_STATE to properly illustrate how to create a ↵Mark Michelson
presence hint. There was a missing comma. This was discovered by Dan Kaplan. ........ Merged revisions 409886 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409887 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Corrected cross-platform stat nanosecond codeDavid M. Lee
When nanosecond time resolution was added for identifying config file changes, it didn't cover all of the myriad of ways that one might obtain nanosecond time resolution off of struct stat. Rather than complicate the #if even further figuring out one system from the next, this patch directly tests for the three struct members I know about today, and #ifdef's accordingly. Review: https://reviewboard.asterisk.org/r/3273/ ........ Merged revisions 409833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409834 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409835 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Fix res/res_http_websocket.c build failure in 32bit due to incorrect print ↵Moises Silva
format for uint64_t git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Fix WebRTC over WSS not workingMoises Silva
Several fixes for the WebSockets implementation in res/res_http_websocket.c * Flush the websocket session FILE* as fwrite() may not actually guarantee sending the data to the network. If we do not flush, it seems that buffering on the SSL socket for outbound messages causes issues * Refactored ast_websocket_read to take into account that SSL file descriptors may be ready to read via fread() but poll() will not actually say so because the data was already read from the network buffers and is now in the libc buffers (closes issue ASTERISK-23099) (closes issue ASTERISK-21930) Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged revisions 409681 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409697 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Fix references to 'keys' CLI commands in astgenkeySean Bright
........ Merged revisions 409777 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409778 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409779 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Blocked revisions 409762Igor Goncharovskiy
........ Correct RTP handling in chan_unistim and fix transfer process broken in previous fix: - Fixed too early RTP setup with phone, that cause no ringback tone on caller side - Handle call transfer cancel only in STATE_CALL case (related to ASTERISK-23073) (Reported by: Németh Tamás, niurkin sil) ........ Merged revisions 409761 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue ↵Igor Goncharovskiy
functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload' ........ Merged revisions 409705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409745 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05stasis: Made internal_stasis_subscribe() prototype and definition match exactly.Richard Mudgett
........ Merged revisions 409682 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-04func_audiohookinheritance: Check If A Channel Was SpecifiedMichael L. Young
This patch prevents a crash when using the function audiohookinheritance without setting the channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal Tested by: Joel Vandal Patches: asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3272/ ........ Merged revisions 409623 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409625 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409626 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-04res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICEJonathan Rose
ICE sessions will now be restarted if sessions are changed to use new sets of remote candidates. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/ ........ Merged revisions 409565 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409570 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-04AO2: Add an assert for bad objectsKinsey Moore
This adds an assert that will only be active if Asterisk is compiled with DO_CRASH and allows the testsuite to fail tests that would otherwise require log file parsing. ........ Merged revisions 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409567 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409568 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-04Minor whitespace change to 'sip show peers' output.Sean Bright
(closes issue ASTERISK-23406) Reported by: ibercom Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom ........ Merged revisions 409472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409473 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409474 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-03res_stasis_recording: Fix memory leak of the absolute name.Joshua Colp
........ Merged revisions 409422 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-03doxygen: Tweak the link back to ye olde Digium websiteMatthew Jordan
........ Merged revisions 409361 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409362 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409363 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-02Makefile: replace -O6 with -O3Tzafrir Cohen
-O6 is not a legal option of gcc. Unofficially gcc considers it to be equivalent of -O3. clang chalks on it, though. This commit sets the default optimization flag to be -O3, like gcc actually considered it. Review: https://reviewboard.asterisk.org/r/3280/ ........ Merged revisions 409308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409344 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409346 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-01res_pjsip_session: Set options (100rel, timers) on incoming sessions.Joshua Colp
This change passes options to the UAS creation function. This in turn sets up 100rel and session timer properties on the incoming session. Reported by Julian Russell on asterisk-users mailing list. ........ Merged revisions 409287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-01devicestate.c: Simplified some logic in _ast_device_state().Richard Mudgett
........ Merged revisions 409274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-01stasis_cache.c: Remove some unnecessary RAII_VAR() usage.Richard Mudgett
........ Merged revisions 409272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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