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2014-02-07chan_iax2: Block unnecessary control frames to/from the wire.Richard Mudgett
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later) results in an unexpected call disconnect. The problem happens because newer values in the enum ast_control_frame_type are not consistent between the branch versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) using IAX2 2) v1.8 answers and sends a connected line update control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 receives the control frame as an end-of-q (on v1.4 AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the receive queue becomes empty. Several things are done by this patch to fix the problem and attempt to prevent it from happening again in the future: * Added a warning at the definition of enum ast_control_frame_type about how to add new control frame values. * Made block sending and receiving control frames that have no reason to go over the wire. * Extended the connectedline iax.conf parameter to also include the redirecting information updates. * Updated the connectedline iax.conf parameter documentation to include a notice that the parameter must be "no" when the peer is an Asterisk v1.4 instance. (closes issue AST-1302) Review: https://reviewboard.asterisk.org/r/3174/ ........ Merged revisions 407678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407727 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407729 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07security_events: Fix error caused by DTD validation errorMatthew Jordan
The appdocsxml.dtd specifies that a "required" attribute in a parameter may have a value of yes, no, true, or false. On some systems, specifying "False" instead of "false" would cause a validation error. This patch fixes the casing to explicitly match the DTD. ........ Merged revisions 407676 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07indications.conf: add stutter tone; end properlyTzafrir Cohen
* If the "stutter" (voicemail indication) tone is indeed a stutter tone, and it ends with a constant tone, make sure that it is the dial tone. This was done for India (in), Mexico (mx) and the Philippines (ph). * If no "stutter" tone exists for a country, provide one. This was done for Spain (es), Malaysia (my) and Venezuela (ve). Review: https://reviewboard.asterisk.org/r/3158/ ........ Merged revisions 407622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407623 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-06security_events: Add AMI documentation; output optional fieldsMatthew Jordan
This patch adds documentation for the Security Events that are emited over AMI. It also notes these events in the UPGRADE/CHANGES file. ........ Merged revisions 407589 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-06configs/pjsip.conf.sample: Configuration section naming in pjsip.conf.sample ↵Rusty Newton
needs a little clarification There is a bit of nuance to how you name things in pjsip.conf. This is a documentation patch to at least clear it up a little for users. Review: https://reviewboard.asterisk.org/r/3180/ ........ Merged revisions 407587 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-06pjsip realtime: already created enum failure for postgresqlKevin Harwell
If an enum had been previously created the alembic script would attempt to re-create it and an error would be generated while running migrations for a postgresql server. The work around for this is to use the ENUM object type for postgres as opposed to the generic enum type used by sqlalchemy. Using this type in the script seems to work properly for both postgres and mysql. ........ Merged revisions 407572 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-06res_pjsip: Updates and adds more PJSIP CLI commands.Richard Mudgett
* Adds identify, transport, and registration support to the PJSIP CLI. * Creates three additional callbacks, one for an iterator, one for a comparator, and one for a container. This eliminates the link dependency from higher level modules to lower level ones. * Eliminates duplicate sorting in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. * Pushes CLI command registration down to the implementing source file. * Adds several ast_sip_destroy_sorcery functions to complement existing ast_sip_sorcery_initialize functions. The destroy functions unregister PJSIP CLI commands and PJSIP CLI formatters. Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3104/ ........ Merged revisions 407568 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05formats/format_wav: enhancing log message "Not a wav file" to be clear on ↵Rusty Newton
what is supported Modifying the log message to be more specific as to what is supported. Specifically it seems format_wav supports only PCM encoded versions with a lower-case '.wav' extension. (closes issues ASTERISK-22310) Reported by: Jim Credland Review: https://reviewboard.asterisk.org/r/3188/ ........ Merged revisions 407511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407512 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407513 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05CHANGES: Improved description of Name/Creator changes to bridge ARI, adds AMIJonathan Rose
The changes log was written with language that was a little too internal Asterisk specific, so it's been changed to be more in the frame of reference of an ARI user. Also, previously the AMI event changes were omitted from the change log as well as the ability to include a bridge name in the ARI post bridges command. ........ Merged revisions 407461 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05Logger: Fix handling of absolute pathsKinsey Moore
This fixes path handling for log files so that an extra / is not appended to the file path when the path is absolute (begins with /). This would previously result in different but functionally equivalent paths in the output of 'logger show channels'. ........ Merged revisions 407455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407456 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05res_pjsip: When no global type the debug option defaults to "yes"Kevin Harwell
If the global section was not specified in pjsip.conf then the configuration object does not exist in sorcery so when retrieving "debug" option it would return NULL. Then the NULL result was passed to ast_false utils function which would return false because it wasn't set to some representation of false, thus enabling sip debug logging. Made it so if the global config object does not exist then it will return a default of "no" for sip debugging. (issue ASTERISK-23038) Reported by: Rusty Newton ........ Merged revisions 407442 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05CHANGES: Update changes log to include r403414 entryJonathan Rose
Adds note of additional 0 for operator option on app_record git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05CHANGES: Update changes log to include new bridge fields added in r404042Jonathan Rose
........ Merged revisions 407419 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05ARI/AMI: Update versions; update UPGRADE/CHANGES notes for 12.1.0 changesMatthew Jordan
Due to backwards compatible changes made to AMI/ARI, the version needs to be bumped to 1.1.0/2.1.0, respectively. ........ Merged revisions 407402 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-04devicestate: Make ast_devstate_changed_literal() return value and doxygen ↵Richard Mudgett
consistent. Nothing actually cares about the value anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose ........ Merged revisions 407337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407338 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407339 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-04res_pjsip: Fix assertion for pjsip.conf authorization list options.Richard Mudgett
(closes issue ASTERISK-23168) Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3143/ ........ Merged revisions 407324 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-04tcptls.c: Made TLS handle a certificate chain file.Richard Mudgett
Thanks to Guillaume Martres for doing the necessary research to validate the change. (closes issue ASTERISK-17727) Reported by: LN Patches: use_certificate_chain.patch (license #5864) patch uploaded by st documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres ........ Merged revisions 407272 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407273 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-04funcs/func_cdr: Fix non-epoch timestamps broken by improper char array derefMatthew Jordan
Thanks to snuffy for pointing this issue out and fixing it. (closes issue ASTERISK-23250) Reported by: snuffy patches: func_cdr-fix.diff uploaded by snuffy (License 5024) ........ Merged revisions 407259 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-04res_clialiases: Fix crash when reloading and re-aliasing an alias that is in ↵Joshua Colp
use. The code assumed that unregistering the alias would always succeed while in practice this is not actually true. A common case is the "reload" command itself. If the cli_aliases.conf configuration file was changed and reload executed the command would fail to unregister and ultimately point to freed memory. The reload process now checks whether unregistering succeeded or not and if not the old CLI alias is retained. (closes issue ASTERISK-19773) Reported by: Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth Blades ........ Merged revisions 407205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407210 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407213 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-04Skinny - Fix deadlock when pickup of no call.Damien Wedhorn
Locking issues in skinny when picking up a call that doesn't exist. Cleaned up sub locking by fully removing and using the chan lock instead. Also changed ast_call_pickup to check whether chan was masq'd. (closes issue ASTERISK-23249) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-locking01.diff uploaded by wedhorn (license 5019) ........ Merged revisions 407197 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-03cdrs: Check for applications to lock onto during dial begin handlingMatthew Jordan
This patch brings CDR processing further in line with r407085. During some dial operations, the application would not be locked to the Dial application and would instead continue to show the previously known application. In particular, this would occur when a Parked call would time out. This was due to a previous snapshot already locking the application to Park - processing this in a Dial Begin allows the Dial application to reassert its rightful place. (CDRs. Ugh.) But hooray for the Parked Call tests for catching this in the Asterisk Test Suite. ........ Merged revisions 407166 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-01res_stasis: Enable transfers and provide events when they occur.Joshua Colp
This change enables transfers within ARI created bridges and adds events for when they occur. Unlike other events these will be received if *any* subscribed object is involved in the transfer. (closes issue ASTERISK-22984) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/3120/ ........ Merged revisions 407153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-01app_stack: protect against missing parameters to STACK_PEEK and LOCAL_PEEKCorey Farrell
STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter. This protects against situations where those parameters are blank or missing by logging an error and returning. (closes issue ASTERISK-23220) Reported by: James Sharp ........ Merged revisions 407100 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407103 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407104 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31CDRs: fix a variety of dial status problems, h/hangup handler creating CDRsMatthew Jordan
This patch fixes a number of small-ish problems that were noticed when witnessing the records that the FreePBX dialplan produces: (1) Mid-call events (as well as privacy options) have the ability to change the overall state of the Dial operation after the called party answers. This means that publishing the DialEnd event when the called party is premature; we have to wait for the execution of these subroutines to complete before we can signal the overall status of the DialEnd. This patch moves that publication and adds handlers for the mid-call events. (2) The AST_FLAG_OUTGOING channel flag is cleared if an after bridge goto datastore is detected. This flag was preventing CDRs from being recorded for all outbound channels that had a 'continue' option enabled on them by the Dial application. (3) The CDR engine now locks the 'Dial' application as being the CDR application if it detects that the current CDR has entered that app. This is similar to the logic that is done for Parking. In general, if we entered into Dial, then we want that CDR to record the application as such - this prevents pre-dial handlers, mid-call handlers, and other shenaniganry from changing the application value. (4) The CDR engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more places to determine if the channel is in hangup logic or dead. In either case, we don't want to record changes in the channel. (5) The default option for "endbeforehexten" has been changed to "yes". In general, you don't want to see CDRs in the 'h' exten or in hangup logic. Since the semantics of that option changed in 12, it made sense to update the default value as well. (6) Finally, because we now have the ability to synchronize on the messages published to the CDR topic, on shutdown the CDR engine will now synchronize to the messages currently in flight. This helps to ensure that all in-flight CDRs are written before shutting down. (closes issue ASTERISK-23164) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3154 ........ Merged revisions 407084 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31app_dial: Allow macro/gosub pre-bridge execution to occur on prioritiesMatthew Jordan
The parsing for the destination of the macro/gosub uses the '^' character to separate out context, extension, and priority. However, the logic for the macro/gosub execution was written such that it would only do the actual macro/gosub jump if a '^' character existed. This doesn't apply when the macro/gosub jump occurs in a priority/priority label. This patch changes the logic so that the parsing still occurs, but the jump will occur even for priorities/priority labels. (issue ASTERISK-23164) Review: https://reviewboard.asterisk.org/r/3154 ........ Merged revisions 407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407074 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407082 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31res_pjsip: Config option to enable PJSIP logger at load time.Kevin Harwell
Added a "debug" configuration option for res_pjsip that when set to "yes" enables SIP messages to be logged. It is specified under the "system" type. Also added an alembic script to add the option to realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged revisions 407036 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31res_pjsip_exten_state: Exporting global symbols caused load order issuesKevin Harwell
Removed the exportation of global symbols from the module as it is no longer needed and it could potentially cause load problems as on some systems it would try to load before res_pjsip_pubsub ........ Merged revisions 407034 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31ChanSpy: Add ability to specify channel uniqueids as well as channel names.Richard Mudgett
* Made ChanSpy accept a channel uniqueid or a fully specified channel name as the chanprefix parameter if the 'u' option is specified. (closes issue AFS-42) Review: https://reviewboard.asterisk.org/r/3160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31Add file that apparently got missed in the merge.Mark Michelson
........ Merged revisions 407031 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31Decouple subscription handling from NOTIFY/PUBLISH body generation.Mark Michelson
When the PJSIP pubsub framework was created, subscription handlers were required to state what event they handled along with what body types they knew how to generate. While this serves well when implementing a base RFC, it has problems when trying to extend the body to support non-standard or proprietary body elements. The code also was NOTIFY-specific, meaning that when the time comes that we start writing code to send out PUBLISH requests with MWI or presence bodies, we would likely find ourselves duplicating code that had previously been written. This changeset introduces the concept of body generators and body supplements. A body generator is responsible for allocating a native structure for a given body type, providing the primary body content, converting the native structure to a string, and deallocating resources. A body supplement takes the primary body content (the native structure, not a string) generated by the body generator and adds nonstandard elements to the body. With these elements living in their own module, it becomes easy to extend our support for body types and to re-use resources when sending a PUBLISH request. Body generators and body supplements register themselves with the pubsub core, similar to how subscription and publish handlers had done. Now, subscription handlers do not need to know what type of body content they generate, but they still need to inform the pubsub core about what the default body type for a given event package is. The pubsub core keeps track of what body generators and body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core will check that there is a subscription handler for the event in the SUBSCRIBE, then it will check that there is a body generator that can provide the content specified in the Accept header(s). Because of the nature of body generators and supplements, it means res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no longer worry about body types, instead calling ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body. Review: https://reviewboard.asterisk.org/r/3150 ........ Merged revisions 407016 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31alembic: script modifications due to errorsKevin Harwell
A couple of the scripts had errors that would not allow a full migration to take place. The extensions table needed to make its 'id' column a primary key in order to work with mysql. The other script ...add_endpoints... was missing tables that it was trying to add columns to. Added the primary key on id for extensions and added the tables in for the missing pjsip configuration options. While it is not ideal to modify already released scripts this was a case where it had to be done due to errors in the script and lacking a better alternative. Review: https://reviewboard.asterisk.org/r/3167/ ........ Merged revisions 407019 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31res_pjsip_mwi: Subscribe fails when missing aor nameKevin Harwell
When subscribing to MWI (res_pjsip_mwi) and the sip uri did not contain a name (ex: sip:<ip address>) then the subscription would fail since it would be unable to locate an associated aor. This patch makes it so that when a subscribe comes with no aor name then it will subscribe to all aors on the located endpoint. (closes issue ASTERISK-23072) Reported by: Bob M Review: https://reviewboard.asterisk.org/r/3164/ ........ Merged revisions 407014 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31PJSIP: Fix address for ACK in NAT situationsKinsey Moore
In NAT scenarios where a call is placed to a Grandstream phone, res_pjsip will sometimes send the ACK to a 200 OK to the private address of the device behind the NAT instead of the address of the NAT device. This corrects that behavior by rewriting the address in the Contact header in the incoming 200 OK and the dialog's target address if necessary (since it has already been rewritten to the incorrect private address). (closes issue ASTERISK-23106) Review: https://reviewboard.asterisk.org/r/3168/ Reported by: Matt Jordan ........ Merged revisions 407000 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31Skinny: fix up possible double unlock of chan.Damien Wedhorn
Return before chan is possibly unlocked a second time when hanging up a channel in SUBSTATE_OFFHOOK. ........ Merged revisions 406987 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-30res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictionsCorey Farrell
ast_bind to a port reserved for another program by SELinux causes errno == EACCES. This caused random failures when binding rtp or udptl sockets. Treat EACCES as a non-fatal error, try next port. (closes issue ASTERISK-23134) Reported by: Corey Farrell ........ Merged revisions 406933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406934 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406935 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-30Make a NOTICE about an invalid channel name more useful.Sean Bright
........ Merged revisions 406918 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406919 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-29queues.conf.sample Fix documented default for persistentmembersRussell Bryant
Closes issue ASTERISK-22662 ........ Merged revisions 406860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406861 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406862 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28res_pjsip_pubsub: potential crash on timeoutKevin Harwell
What seems to be happening is if a subscription has been terminated and the subscription timeout/expires is less than the time it takes for all pending transactions (currently on the subscription) to end then the subscription timer will not have been canceled yet and sub will be null. Since the subscription has already been canceled nothing needs to be done so a null check in the asterisk code is sufficient in working around this problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins ........ Merged revisions 406847 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28cdr_radius, cel_radius: build agains libfreeradius-clientKevin Harwell
Asterisk's RADIUS module currently build against libradiusclient-ng, but this project has been superseeded by libfreeradius-client. The API is 99% compatible except that the header name has changed, the library name has changed, and the configuration file location has changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé Patches: freeradius-client.patch uploaded by sharky (license 6561) ........ Merged revisions 406801 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406802 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406803 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28res_pjsip,compat: INFINITY and NAN undefinedKevin Harwell
On some systems the values for INFINITY and NAN are not defined thus causing a build error on those systems. Added definitions for those if they had not previously been defined. (closes issue ASTERISK-23056) Reported by: capouch Patches: inf-nan-patch.txt uploaded by capouch (license 6564) ........ Merged revisions 406788 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28ARI: Make double subscribe respond with successKinsey Moore
Currently, attempting to subscribe an application to a device state that it has already subscribed to will generate a 500 error response. This will now be treated as a subscription refresh even though ARI subscriptions don't currently support lifetimes and will respond with the normal response for a successful subscription (200 OK). (closes issue ASTERISK-23143) Reported by: Matt Jordan ........ Merged revisions 406775 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28rtp_engine: improved handling of get_rtp_info failureScott Griepentrog
In ast_rtp_instance_make_compatible(), after a failure of channel tech call get_rtp_info() to return peer_instance, the null pointer would be passed to ao2_ref, producing an error that looked like a refernce counting problem but is not. This patch corrects that and adds helpful LOG_ERROR messages to indicate which failure path occurred. (issue AST-1276) Review: https://reviewboard.asterisk.org/r/3156/ ........ Merged revisions 406721 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406722 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406723 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28test_cdr.c, test_cel.c: Correctly destroy created bridges.Richard Mudgett
* Fixed the test_cel_attended_transfer_bridges_link unit test to also account for the local channel link being destroyed now that the bridges are actually destroyed. * Made CDR unit test use its own version of do_sleep() from the CEL unit tests. ........ Merged revisions 406707 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-27manager: ExtensionStatus event status human readableKevin Harwell
Added a note in the changes file about the new 'StatusText' field that was added to the 'ExtensionStatus' event. (issue ASTERISK-23154) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-27manager: ExtensionStatus event status human readableKevin Harwell
When an 'ExtensionStatus' event was raised it included the status as a numerical value, but did not include a text description of the status. Added a 'StatusText' field to the event which is a string representation of the extension status. Also added this to the 'Extension State' command response. (closes issue ASTERISK-23154) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-27Allow nested #includes in extconfig.confRussell Bryant
extconfig.conf was hard-coded to not allow nested includes for some reason. The code has been this way since a patch was merged for ASTERISK-3333 (revision 4889), which was a significant update to this code ("Merge config updates"). I can't figure out any good reason why this should be limited. This patch just removes the limit and uses the default nesting depth limit. Closes issue ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/ ........ Merged revisions 406643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406644 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406645 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-27manager: The eventfilter= option now takes an extended regex.Walter Doekes
In pre-trunk versions (...12) it accepts a basic regex, which is confusing because all other regexes in asterisk are of the extended kind. Review: https://reviewboard.asterisk.org/r/3147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-27Protect ast_filestream object when on a channelRussell Bryant
The ast_filestream object gets tacked on to a channel via chan->timingdata. It's a reference counted object, but the reference count isn't used when putting it on a channel. It's theoretically possible for another thread to interfere with the channel while it's unlocked and cause the filestream to get destroyed. Use the astobj2 reference count to make sure that as long as this code path is holding on the ast_filestream and passing it into the file.c playback code, that it knows it's valid. Bug reported by Leif Madsen. Review: https://reviewboard.asterisk.org/r/3135/ ........ Merged revisions 406566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406567 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406574 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-26tcptls.c: Add missing cleanup on off nominal path.Richard Mudgett
........ Merged revisions 406514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406515 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406516 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-26live_ast: run wrapped programs with execTzafrir Cohen
live_ast can be used as a wrapper script to run asterisk, gdb or valgrind. In those cases it runs them and returns the result. It is more useful to use 'exec' to avoid having another odd process in the chain. Review: https://reviewboard.asterisk.org/r/3110/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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