asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	a737df8603	Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	eb449d514e	Minor improvements to app_fax. This patch makes some small changes to handle watchdog timeouts in a better way, and also uses a 'cleaner' method of including the spandsp header files. (closes issue #14769) Reported by: andrew Patches: app_fax-20090406.diff uploaded by andrew (license 240) v1-14769.patch uploaded by dimas (license 88) Tested by: freh, deti, caspy, dimas, sgimeno, Dovid git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	28ad5ced1a	Initial minimum ast_party_caller support. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	48363c16e1	Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ba020fc390	Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	d81d6d3415	Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	17e2d9fdbc	Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0a6e06c7ff	Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum IFP size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending our MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d223e3636f	Merged revisions 208083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208083 \| tilghman \| 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009) \| 4 lines Export symbols for functions included in our compatibility headers. (closes issue #15556) Reported by: smw1218 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	299a9ff3fa	Remove trailing whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	4cf8a968fd	Add an API for reporting security events, and a security event logging module. This commit introduces the security events API. This API is to be used by Asterisk components to report events that have security implications. A simple example is when a connection is made but fails authentication. These events can be used by external tools manipulate firewall rules or something similar after detecting unusual activity based on security events. Inside of Asterisk, the events go through the ast_event API. This means that they have a binary encoding, and it is easy to write code to subscribe to these events and do something with them. One module is provided that is a subscriber to these events - res_security_log. This module turns security events into a parseable text format and sends them to the "security" logger level. Using logger.conf, these log entries may be sent to a file, or to syslog. One service, AMI, has been fully updated for reporting security events. AMI was chosen as it was a fairly straight forward service to convert. The next target will be chan_sip. That will be more complicated and will be done as its own project as the next phase of security events work. For more information on the security events framework, see the documentation generated from doc/tex/. "make asterisk.pdf" Review: https://reviewboard.asterisk.org/r/273/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	67d1957e60	Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f3560397be	Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 \| dvossel \| 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) \| 2 lines Changing ast_samp2tv to not use floating point. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ba2a8457b8	Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 \| dvossel \| 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) \| 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e39a252b1e	Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 \| dvossel \| 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) \| 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	15b94d1182	Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 \| dvossel \| 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) \| 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	e75ae63ac2	Fix a few compilation problems found when building Asterisk against uClibc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0e8c630224	Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	ee0cd5a32c	Add a configure check for Reverse Charging Indication support in LibPRI. Also go back and wrap all of the places that use the specific reverse charge APIs with preprocessor conditionals. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	c381cf82e7	Wrap rtp_engine.h header comments to 80 characters. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	48c9a85d91	Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 \| dvossel \| 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) \| 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c511a26749	Move Asterisk-addons modules into the main Asterisk source tree. Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b5f6eac49e	Allow trunk to once again compile under MALLOC_DEBUG git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cce4fad522	Make invalid hints report Unavailable instead of Idle. (closes issue #14413) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	59c1998d67	Improve T.38 negotiation by exchanging session parameters between application and channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	27e1708eed	Note a new API call, and one that changed in doxygen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	e06c6f97c4	Add functions to map syslog facilities and priorities constants to strings. Also change the default casing of the string contants to lowercase. This really just saves us from have to lowercase them later when displaying them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	2f88262abb	Add checks in configure for non-POSIX syslog facilities. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	4535305772	Move syslog utility functions into a separate file so they can be re-used. This has the pleasant side effect of cleaning up the header inclusion process in logger.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	87c8658912	attempting to load running modules Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	e85296e244	Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dce6a54a4a	Trunk implementation of setting an alternate RTP source. This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	962bd7ab26	Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 \| kpfleming \| 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) \| 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is not, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f1dc620467	Enable applications to enable/disable digit and tone detection. Some applications (notably app_fax) do not need digit detection nor FAX tone detection while they are running, and if Asterisk is using software DSPs to provide the detection, this consumes extra CPU cycles that could be better spent on the actual application. This patch allows applications to query and control the state of digit and tone detection on a channel, and modifies app_fax to disable them while the FAX operations are occurring (and re-enable digit detection afterwards). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4c0265664e	Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 \| kpfleming \| 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) \| 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	aaeec3b40f	Last batch of 'static' qualifiers for module-level global variables. Fix up modules in the 'apps' directory, and also correct the bad example of enum definitions in include/asterisk/app.h, which many developers followed (thanks for reading the documentation!). In addition, add some basic usage examples of the 'pahole' and 'pglobal' tools to the coding guidelines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	82fb56886e	More 'static' qualifiers on module global variables. The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	6c5987811c	Redesigned 'optional API' support. This patch provides a new implementation of the optional API support defined in asterisk/optional_api.h; this new version provides solves compatibility issues with the use of linker version scripts for suppressing global symbols. In addition, there is now a functional (and tested!) implementation for Mac OS/X, so module writers no longer need to use special tests before calling optional API functions. All future implementations must provide these same semantics, so that module writers can rely on them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	2cec55f038	Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 \| seanbright \| 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) \| 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d532cbcd8a	module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	8ceb6c5d20	Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 \| seanbright \| 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) \| 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 \| seanbright \| 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) \| 2 lines Fix a typo in the stack size calculation just introduced. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d1d9beadc9	Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 \| dvossel \| 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) \| 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	befad10893	Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 \| seanbright \| 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) \| 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	3830c415c7	Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	298d745fb4	Add the ability to execute connected line interception macros. When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8da5e991ee	Minor whitespace fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8580871fd4	Constify the ast_frame arg to ast_queue_frame(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	c8b0c41ed8	Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 \| mnicholson \| 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) \| 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	c772d5a0e6	Update references to downloads.digium.com to its new URL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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