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Author	SHA1	Message	Date
Mark Michelson	5819278c46	Revert Makefile change to remove embedding res_adsi.so ... The change has resulted in a linking error for certain versions of GCC. This is much worse than the original issue, so for now, temporarily revert the change. A more thorough change will be sought out. ........ Merged revisions 368927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368928 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	cfa0826c49	Add a post_apply callback to the Config Options API ... This adds a callback that only fires when changes have been successfully applied via the Config Options API. Review: https://reviewboard.asterisk.org/r/1980/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	01307e4b7b	Add filename alias support to the Config Options API ... This adds the ability to handle a single filename alias for a config file. This is useful if a config filename has changed, but the old filename should be supported for backwards compatibility. Review: https://reviewboard.asterisk.org/r/1981/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	21997aa7bb	Fix a deadlock that occurs when func_volume is used on a local channel. ... This was discovered by trying to perform a call forward to an extension that makes use of func_volume. When the local channel is optimized away, the datastore on the local;2 channel would have its audiohook destroyed rather than detaching the audiohook from the channel and then destroying it. With this patch, func_volume's datastore destructor takes the proper route of detaching the audiohook and then destroying it. (closes issue ASTERISK-19611) reported by Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368899 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	ff0b561045	Mark res_smdi/res_adsi as 'core' supported modules ... Recently, various issues surrounding weak symbols have caused problems with modules that rely on that feature to be enabled in menuselect. This includes app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi, which, in certain circumstances, may not be enabled by default in menuselect. Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this patch marks both as 'core' supported modules. This will allow both app_voicemail and chan_dahdi to be enabled as well, regardless of whether or not that system supports weak symbols. (issue AST-900) Reported by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto Martinez ........ Merged revisions 368894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368895 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	b445e8a7c8	Remove forced linking of res_adsi.o ... In GCC 4.5+ the result is that Asterisk has a phantom module loaded at startup, claiming to be res_adsi. (closes issue ASTERISK-19920) reported by Leif Madsen ........ Merged revisions 368873 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368885 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a8e895c1a0	Replace MODULES_DIR with ASTMODDIR in Makefile's INSTALLDIRS ... Post Asterisk 10, the MODULES_DIR variable no longer exists, and was replaced with ASTMODDIR. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	2362ee5738	Do not install empty directories; add ASTLIBDIR ... r368830 modified the installation script to only create a directory if that directory does not exist. If some directory variable was empty, it would attempt to create the empty location. It also failed to create the ASTLIBDIR directory. This patch fixes it such that the correct directories are made and only created if a value specifying them actually exists. ........ Merged revisions 368852 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368853 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	019010e611	Do not perform install on existing directories ... If a directory already exists, performing a 'make install' will remove the permissions associated with the current directory and replace them with the permissions of the user executing the install. This patch changes this behavior to only perform an install on the directory if the directory does not exist. Thus, if a user later changes the permissions on that directory, those permissions will be preserved in subsequent installs. Review: https://reviewboard.asterisk.org/r/1986 Review: https://reviewboard.asterisk.org/r/1864 (closes issue ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger, Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded by mjordan) ........ Merged revisions 368830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368831 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	6bd3eb4995	Set the Caller ID "tag" on peers even if remote party information is present. ... On incoming calls, we were setting the cid_tag on the dialog only if there was no remote party information (Remote-Party-ID or P-Asserted-Identity) present. The Caller ID tag is an invented parameter, though, and should be set no matter the circumstance. (closes issue ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884) Reported by Trey Blancher ........ Merged revisions 368807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368808 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	def53014ec	Update merge property information ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	8bc3c1e20f	Fix deadlock in SIP transfers that involve a REFER request ... In r367163, "send to voicemail" functionality was added to the SIP channel driver. This required updating the party redirecting information for the channel based on the headers provided in the REFER request. When the redirecting party information is updated on the channel, a call to ast_indicate_data occurs. Because handle_request_refer still had the sip_pvt locked, a deadlock could occur between the pbx_thread and the do_monitor thread servicing the REFER request. This patch preserves the proper locking order between the channel and the sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party redirecting information on the channel. (closes issue AST-903) Reported by: Matt Jordan patches: jira_ast_903_trunk.patch by rmudgett (license 5621) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	afa03bd310	Parse ANI2 information from SIP From header parameters ... ANI2 information is now parsed out of SIP From headers when present in the oli, isup-oli, and ss7-oli parameters and is available via the CALLERID(ani2) dialplan function. (closes issue ASTERISK-19912) Patch-by: Rob Gagnon Review: https://reviewboard.asterisk.org/r/1947/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	72eb8eb1e7	Fix deadlock potential with ast_set_hangupsource() calls. ... Calling ast_set_hangupsource() with the channel lock held can result in a deadlock because the function also locks the bridged channel. (issue ASTERISK-19537) (closes issue AST-891) Reported by: Guenther Kelleter Tested by: Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec Davis ........ Merged revisions 368759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368760 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c6142cf2cc	Fix coverity UNUSED_VALUE findings in core support level files ... Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) ........ Merged revisions 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368739 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	566ea22e18	Recorded merge of revisions 368721 from http://svn.asterisk.org/svn/asterisk/branches/10 ... ........ Fix compilation in dev-mode Backport a compilation fix in md5.c from trunk that only showed up in dev-mode under certain compiler versions. ........ Merged revisions 368719 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	745484e1b3	Fix error paths in action_hangup() for AMI Hangup action. ... * Check allocation function return values for failure. Crashing is bad. * Tweak ast_regex_string_to_regex_pattern() parameters for proper ast_str usage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8b2412db28	Tweak ast_channel_softhangup_withcause_locked() to take a typed parameter. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	4ca35e0907	Fix MWI update so LED display correct voicemail state after phone usage. Also fixes few warnings. ...	14 years ago
Damien Wedhorn	d979399071	Skinny cleanup (mwi_event_cb). ... Original was testing for d->session, setting and testing again (all nested). Removed duplicate testing and restructured function to test/return and then the main code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	0271734f2e	Skinny cleanup. ... Removed d->registered which was mirroring d->session. Changed relevant references to use d->session instead. Moved setting and unsetting of l->device from session register to device configuration. As such, l->device will always be valid unless it is has not been configured to a device. Revised various test where checking if a device is registered to use l->device->session. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3f59ad990c	Fix app_queue debug message use of args.options after the string has been parsed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9ecd6c9ab4	Fix inverted test in app_queue for ringinuse. ... Regression from -r367080 ringinuse commit. (issue ASTERISK-19536) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9f704b5d59	Fix reloading an unchanged file with the Config Options API ... Adding multiple file support broke reloading an unchanged file. This adds an enum for return values for the aco_process_* functions and ensures that the config is not applied if res is not ACO_PROCESS_OK. Review: https://reviewboard.asterisk.org/r/1979/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	ac6ec71fd2	Fix a typo in format_ogg_vorbis.c: suport ... Review: https://reviewboard.asterisk.org/r/1970/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	aeeff8cfa2	Add default handler documentation and standardize acl handler ... Added documentation describing what flags and arguments to pass to aco_option_register for default option types. Also changed the ACL handler to use the flags parameter to differentiate between "permit" and "deny" instead of adding an additional vararg parameter. Review: https://reviewboard.asterisk.org/r/1969/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0f71b29e2f	Fix POTS flash hook to orignate a second call deadlock. ... A deadlock can occur when a POTS phone tries to flash hook to originate a second call for 3-way or transfer. If another process is scanning the channels container when the POTS line flash hooks then a deadlock will occur. * Release the channel and private locks when creating a new channel as a result of a flash hook. (closes issue ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368644 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368645 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	ea8cf8b5f3	Fix a specific scenario where ACKs are not matched. ... If a dialog-starting INVITE contains a to-tag, then Asterisk will respond with a 481. In this case, the resulting incoming ACK would not be matched, so Asterisk would continue retransmitting the 481 until the transaction times out. There were two issues. Asterisk, upon creating a sip_pvt would generate a local tag. However, when the time came to transmit the 481, since there was a to-tag in the INVITE, Asterisk would place this original to-tag in the 481 response. When the ACK came in, Asterisk would attempt to match the to-tag in the ACK to the generated local tag. Unfortunately, Asterisk never actually transmitted a response with the generated local tag, so the to-tag in the ACK would not match. The other problem was that when the 481 was sent, nothing was set on the sip_pvt to indicate what CSeq is expected in the ACK. To fix the first problem, we zero out the to-tag seen in the incoming INVITE. This way, Asterisk, when time to send a response, will send its generated local tag instead. To fix the second problem, we set the sip_pvt's pendinginvite to the CSeq of the INVITE when we send a 481. (closes issue ASTERISK-19892) Reported by Mark Michelson ........ Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368629 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	d197f69107	Add feature modifier to versions produced from branches ... Certain branches, such as Certified Asterisk, may have a modifier added to them that specifies the features available in that branch. For branches, this modifier is expected to be reflected in the location of the branch in subversion. For example, a subversion of URL of /certified/branches/1.8.11 would have a feature modifier of 'certified'. This is slightly different then how features are determined for tags, where the feature is part of the actual tag name, e.g., "10.5.0-digiumphones". In keeping with the nomenclature used for tags, the feature specifier for branches is translated and placed after the revision numbers. For the example given previously, this would result in a branch version of "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged revisions 368604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	1492177b7b	Ensure overlapping hold flags do not conflict ... When changing between different modes of hold, the flags were not being cleared out properly causing a failure to change hold states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368587 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a2402dbe25	Fix parked call performing a DTMF blind transfer after being retrieved. ... When a parked call was retrieved from the parking lot, it could not do a blind transfer because it caused the involved calls to be hung up unconditionally. * Made the ParkedCall application return the ast_bridge_call() return value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc ........ Merged revisions 368567 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368568 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	faacb8ba52	Make builtin_blindtransfer() fully use ast_async_goto() abilities. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	37677a8cc2	Merge 'core' and 'core changes' sections in CHANGES file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	f6b5fd5411	Recorded merge of revisions 368536 from http://svn.asterisk.org/svn/asterisk/branches/10 ... ........ Resolve some build warnings My newly upgraded compiler caught these usages of uninitialized values. They weren't actually used. ........ Merged revisions 368533 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	bd958c037f	Ensure that pages and emails are sent using RFC822-compliant date format ... When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) ........ Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368524 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	571445ab9c	Convert AST_FLAG_ANSWERED_ELSEWHERE usage to AST_CAUSE_ANSWERED_ELSEWHERE ... This was essentially duplicated functionality where normal channels used AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review: https://reviewboard.asterisk.org/r/1944 (closes issue ASTERISK-19865) Patch-by: Birger Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	d210685a20	Relay proper SIP responses on calling side. ... Revision 351130 broke corect HANGUPCAUSE setting for the 404 case in chan_sip. Other cases were also potentially broken. This patch fixes the relaying of causes to be what they used to be. (closes issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter Doekes (via a reviewboard test to be committed later) Patches: chan_sip.diff uploaded by Pavel Troller (license #6302) ........ Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368499 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cc69a0deaf	Document BLINDTRANSFER behavior change. ... (issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call ........ Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	f4218dc4e6	Also have vim syntax-highlight type=network. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	005661bfdf	Add vim syntax highlighting for type=line, type=phone, and type=application. ... (closes issue ASTERISK-19800) Reported by: Billy Chia Patches: asterisk.vim.patch uploaded by Billy Chia (license #6381) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	c6a2cbab19	Remove some extra debugging I forgot to remove in the merge of Digium phone support. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	458f6c4bc0	Remove automerge properties. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	14a985560e	Merge changes dealing with support for Digium phones. ... Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c1bbe79748	Fix potential deadlock between masquerade and chan_local. ... * Restructure ast_do_masquerade() to not hold channel locks while it calls ast_indicate(). * Simplify many calls to ast_do_masquerade() since it will never return a failure now. If it does fail internally because a channel driver callback operation failed, the only thing ast_do_masquerade() can do is generate a warning message about strange things may happen and press on. * Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This change fixes half of the deadlock reported in ASTERISK-19801 between masquerades and chan_iax. (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1915/ ........ Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368407 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	380c7c5c39	Add res_http_websocket module which implements the WebSocket protocol according to RFC 6455. ... Review: https://reviewboard.asterisk.org/r/1952/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	91a20ee2f9	Fix deadlock when Gosub used with alternate dialplan switches. ... Attempting to remove a channel from autoservice with the channel lock held will result in deadlock. * Restructured gosub_exec() to not call ast_parseable_goto() and ast_exists_extension() with the channel lock held. (closes issue ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	dd02d976f5	Improve SDP offer/answer RFC compliance ... Asterisk should not accept SDP offers that contain unknown RTP profiles (for audio/video streams) or unknown top-level media types. When it does, it answers with an SDP that does not match the offer properly, and this will nearly always result in a broken call. This patch causes such offers to be rejected. Review: https://reviewboard.asterisk.org/r/1811/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	66e5c30716	Improve SDP parsing warning messages ... * 'Unsupported media type' is only reported when that is in fact the case, not when a supported media type is included in an 'm' line that has an invalid format. * All warning messages related to parsing 'm' lines now include the 'm' line contents. * (minor bugfix) newline added to port-number-zero warning messages. * Warning messages improved to use RFC-specified terminology for various items. * Warnings for offers that include more than one port for a single media type now include the media type. Review: https://reviewboard.asterisk.org/r/1811/ ........ Merged revisions 368218 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368267 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6016094db7	Add missing config for config API test ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	d54717c39e	Add new config-parsing framework ... This framework adds a way to register the various options in a config file with Asterisk and to handle loading and reloading of that config in a consistent and atomic manner. Review: https://reviewboard.asterisk.org/r/1873/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago