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2012-08-08Allow support for early media on AMI originates and call files.Mark Michelson
This is based on the work done by Olle Johansson on review board. The idea is that the channel specified in an AMI originate or call file is typically not connected to the outgoing extension until the channel has been answered. With this change, an EarlyMedia header can be specified for AMI originates and an early_media option can be specified in call files. With this option set, once early media is received on a channel, it will be connected with the outgoing extension. (closes issue ASTERISK-18644) Reported by Olle Johansson Review: https://reviewboard.asterisk.org/r/1472 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08Add AMI_CLIENT dialplan functionTerry Wilson
Implementation of a dialplan function for checking manager accounts. Right now it only returns the number of logged in sessions for a manager account, but other attributes can be added later. Patch by: Olle Johansson Review: https://reviewboard.asterisk.org/r/421/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08Create the payload type if it does not exist when setting information based ↵Joshua Colp
on the 'm' line. An rtpmap attribute is not required for defined payload numbers. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08Do not define a cause that doesn't actually existKinsey Moore
AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. ........ Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370924 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-07Payload and RTP code are must remain separate since in non-Asterisk format ↵Joshua Colp
cases they differ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-07Add missing AST_CAUSE_* -> text translationsKinsey Moore
A few of these were missing from the list and are necessary for the Who Hung Up? functionality. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-07Fix a bug uncovered by the test suite where the RTP payload number was not ↵Joshua Colp
getting set. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-07Reduce memory consumption significantly for users of the RTP engine API by ↵Joshua Colp
storing only the payloads present and in use instead of every possible one. Review: https://reviewboard.asterisk.org/r/2052/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-07Add named callgroups/pickupgroupsMatthew Jordan
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-01Fix a possible crash due to passing NULL to ast_variables_dup()Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-01Make astobj2.h not include linkedlists.h.Richard Mudgett
Using astobj2 does not require linkedlists.h be included even though astob2 uses linked lists internally. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31Add "setvar" option to manager.conf.Mark Michelson
With this option set, channel variables can be set on every manager originate. The Variable header can still be used to set additional channel variables for individual calls if desired. This work was completed by Olle Johansson on review board. I have applied the review feedback and am committing it in order to get this into trunk before Asterisk 11 is branched. Review: https://reviewboard.asterisk.org/r/1412 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31Move event cache updates into event processing thread.Russell Bryant
Prior to this patch, updating the device state cache was done by the thread that originated the event. It would update the cache and then queue the event up for another thread to dispatch. This thread moves the cache updating part to be in the same thread as event dispatching. I was working with someone on a heavily loaded Asterisk system and while reviewing backtraces of the system while it was having problems, I noticed that there were a lot of threads contending for the lock on the event cache. By simply moving this into a single thread, this helped performance *a lot* and alleviated some deadlock-like symptoms. Review: https://reviewboard.asterisk.org/r/2066/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31Clean up and ensure proper usage of alloca()Kinsey Moore
This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30Tweak unit test warning message.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30Fix some presence-state unit test typos.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30Add a "corosync ping" CLI command.Russell Bryant
This patch adds a new CLI command to the res_corosync module. It is primarily used as a debugging tool. It lets you fire off an event which will cause res_corosync on other nodes in the cluster to place messages into the logger if everything is working ok. It verifies that the corosync communication is working as expected. I didn't put anything in the CHANGES file for this, because this module is new in Asterisk 11. There is already a generic "res_corosync new module" entry in there so I figure that covers it just fine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-25Repair editline builds using in-tree editline sources.Kevin P. Fleming
The previous change to the build system for using a system-provided editline library was missing a crucial include directory for building against the copy of the library in the Asterisk source tree. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-25Use an absolute path when referring to the embedded editline directory.Kevin P. Fleming
This patch changes the build system to refer to the embedded editline directory using an absolute path, which will resolve a problem seen on the CentOS automated build agents. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-25Enable usage of system-provided NetBSD editline library if available.Kevin P. Fleming
This patch changes the Asterisk configure script and build system to detect the presence of the NetBSD editline library (libedit) on the system. If it is found, it will be used in preference to the version included in the Asterisk source tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie Review: https://reviewboard.asterisk.org/r/1528/ Patches: 0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-25Revert a change that broke compilationTerry Wilson
1) There is no such function as ast_ref() 2) The patch was originally credited as the one uploaded by Guenther Kelleter (license 6372) via issue AST-921, but the patch committed was not the patch referenced on the issue. 3) Guenther Kelleter's patch was actually correct. It moved the ast_free above the presencechange_cleanup label. I am not committing his change as it is not technically necesary--calling ast_free(NULL) is perfectly safe and I worry that moving the ast_free outside of the label could lead to future bugs if someone ever adds another failure conditional and expects 'goto presencechange_cleanup;' to clean up after everything. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-24Don't attempt free of NULL ptr in pbx.c handle_presencechangeJonathan Rose
(closes issue AST-921) Reported by: Guenther Kelleter Patches: nullptr.patch uploaded by Guenther Kelleter (license 6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-24Rewrite a comment that didn't adequately explain the code it was documenting.Kevin P. Fleming
........ Merged revisions 370429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370430 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-24Allow permit/deny ACL lines to contain multiple items and negated entries.Kevin P. Fleming
Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple items (separated by commas), and items in the rule can be negated by prefixing them with '!'. This simplifies Asterisk Realtime configurations, since it is no longer necessray to control the order that the 'permit' and 'deny' columns are returned from queries. Review: https://reviewboard.asterisk.org/r/1592/ Initial patch contributed by Tilghman Lesher Unit tests written by Kevin P. Fleming git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-23Unit tests for the Jitter Buffer API; remove unnecessary resyncMatthew Jordan
This patch includes the following: * Unit tests for the abstract Jitter Buffer API. This includes both fixed and adaptive flavors, testing nominal creation, frame input, frame retrieval, resyncing; off nominal frame input overflow, out of order, and others. * Tweaks to the abstract_jb API to remove the unnecessary resync_threshold parameter from the create function (resync_threshold is already in the struct passed into the create function) * Ensure the fixed jitter buffer is empty before destroying it, to avoid an ASSERT * Don't "resync" the adaptive jitter buffer. The mechanism that was being used actually causes the jitter buffer to think its being overflowed by going around the jitterbuf API and attempting to 'resynch' it improperly. If a resync is needed, the jitter buffer will do it properly by itself. Note that this is only an optimization needed for trunk, as the worst that happens is the loss of three voice packets before the adaptive jitter buffer will resync anyway. Review: https://reviewboard.asterisk.org/r/2035 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-21Fix segfault introduced by conversion to ACO APITerry Wilson
The value "none" is specified in the config file as a valid value for the "video_mode" option. The code prior to the ACO conversion did not check for "none", but just ignored it and relied on the default zero value. The parsing with ACO is more strict, so without handling "none" specifically, parsing would fail. When parsing failed, but the module loaded anyway, the config info would never be stored, and one place in the code did not check for this case and would segfault. It was also possible that the aco_info struct's internals would be destroyed and used as well. This patch keeps the module from loading after parse failures, adds the "none" option to "video_mode", registers CLI functions only after parsing has completed, checks the config data for NULL before accessing it, and returns -1 on some allocation failures when initializing. (closes issue ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt uploaded by Terry Wilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-20Add hangupcause translation supportKinsey Moore
The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better facilitate access to the AST_CAUSE translations for technology-specific cause codes. The HangupCauseClear application has also been added to remove this data from the channel. (closes issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-20Add the AccountCode header to the AMI Hangup event.Richard Mudgett
It's harder to correlate the Newchannel and Hangup AMI events without specifying "AccountCode" in both. (closes issue ASTERISK-19963) Reported by: Oleg A. Arkhangelsky Patches: hangup_acctcode.diff (license #6397) patch uploaded by Oleg A. Arkhangelsky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19Convert app_confbridge to use the config options frameworkTerry Wilson
Review: https://reviewboard.asterisk.org/r/2024/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19Fix compiler warnings.Richard Mudgett
gcc (GCC) 4.2.4 has problems casting away constness. ........ Merged revisions 370275 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370277 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19Add the ability to specify technology specific documentationMatthew Jordan
A number of applications/AMI commands in Asterisk have specific behavioral differences depending on the resource or channel technology those applications are executed on. For example, the MessageSend application/ command is technology agnostic, but how the channel drivers that support that functionality behave is dependant on the protocols and channel driver implementation. Prior to this patch, those details were either documented in the application/command documentation itself, or were left undocumented. This patch adds a new element to the documentation schema, <info/>. An info node is essentially a piece of technology specific reference information that can be included by any top level XML documentation node. For example, the MessageSend application can now include XMPP/SIP specific information, where that technology specific information can be defined in chan_motif/res_xmpp/ chan_sip. Likewise, that information can also be included in the MessageSend AMI command. Review: https://reviewboard.asterisk.org/r/2049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19Fix compilation error when MALLOC_DEBUG is enabledMatthew Jordan
To fix a memory leak in CEL, a channel datastore was introduced whose destruction function pointer was pointed to the ast_free macro. Without MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free. With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a different place then utils.h, and became undefined. This patch resolves this by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be ast_free, which is defined to be free. (issue AST-916) Reported by: Thomas Arimont ........ Merged revisions 370273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370274 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19named_acl: Remove systemname option from acl.conf, use asterisk.conf valueJonathan Rose
Review: https://reviewboard.asterisk.org/r/2057/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19CallID Logging: Remove new line/carriage return from callID change test eventJonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18callid logging: Issue test events when the callid is changed for a channelJonathan Rose
Review: https://reviewboard.asterisk.org/r/2054/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Resolve severe memory leak in CEL logging modules.Kevin P. Fleming
A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2053/ ........ Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370206 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Ensure that all ast_datastore_info structures are 'const'.Kevin P. Fleming
While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13Add support for parsing SDP attributes, generating SDP attributes, and ↵Joshua Colp
passing it through. This support includes codecs such as H.263, H.264, SILK, and CELT. You are able to set up a call and have attribute information pass. This should help considerably with video calls. Review: https://reviewboard.asterisk.org/r/2005/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Handle deprecated (aliased) option names with the config options apiTerry Wilson
Add a simple way to register "deprecated" option names that alias to a different "current" name. Review: https://reviewboard.asterisk.org/r/2026/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Named ACLs: Introduces a system for creating and sharing ACLsJonathan Rose
This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Allow the REALTIME() function to report errors back to the caller.Tilghman Lesher
Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ ........ Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Don't perform an XInclude to a document node that may not always be presentMatthew Jordan
Because some of the manager events are defined in the top of the source, due to the macro calls not containing all necessary information to have the documentation colocated with the call itself, several include statements were failing when built with 'make'. While this did not cause any problems in compilation or validation, it did result in a number of warnings being dumped to stderr. This patch changes those references such that they always resolve, regardless of the documentation build options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Fix validation errors when producing documentation using default build scriptMatthew Jordan
The awk script parses out the first instance of the DOCUMENTATION tag that it finds within a file. If a file did not previously have a DOCUMENTATION tag but received one due to it having an AMI event, then the XML fragment associated with the AMI event was erroneously placed in the resulting XML file. Without the python scripts, these XML fragments will not validate. This patch adds DOCUMENTATION tags at the top of those files that did not previously have them to prevent the awk script from pulling AMI event documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Add some additional documentation for core AMI eventsMatthew Jordan
This patch adds some basic documentation for a number of modules. This includes core source files in Asterisk (those in main), as well as chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD has also been updated to allow referencing of AMI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Improve Goto and GotoIf related documentationKinsey Moore
Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369871 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Fix initial loading problem with res_curlMatthew Jordan
When the OpenSSL duplicate initialization issues were resolved in r351447, res_curl could fail to load if it checked SSL_library_init after SSL initialization completed. This is due to the SSL_library_init stub returning a value of 0 for success, as opposed to a value of 1. OpenSSL uses a value of 1 to indicate success - in fact, SSL_library_init is documented to always return 1. Interestingly, the CURL libraries actually checked the return value - the fact that nothing else that depends on OpenSSL was having problems loading probably means they don't check the return value. (closes issue AST-924) Reported by: Guenther Kelleter patches: (AST-924.patch license #6372 uploaded by Guenther Kelleter) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09When receiving a STUN binding request send one out as the Google Talk client ↵Joshua Colp
uses this as a method to determine if the remote party is still reachable or not. Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis. (closes issue ASTERISK-20107) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06Remove a superfluous and dangerous freeing of an SSL_CTX.Mark Michelson
The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the SSL_CTX structures for servers will be properly freed when ast_ssl_teardown is called. (closes issue ASTERISK-20074) Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license #5049) Testers: Trevor Helmsley ........ Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369732 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06Fix bridging thread leak.Mark Michelson
The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 ........ Merged revisions 369708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369709 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-01Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.Joshua Colp
Review: https://reviewboard.asterisk.org/r/1891/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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