127 Commits (a66fa4db24553d6ec6c8978c528081a94b1715a1)

Author	SHA1	Message	Date
ghjm	0b399013c6	app_followme: Add a prompt to be read when a call is connected ... This patch adds the ability to configure a prompt which will be read to the "winner" who pressed 1 (or the configured value) and received the call. ASTERISK-24372 #close Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272	7 years ago
Sean Bright	ce3d56920b	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Corey Farrell	0f49e6ee2e	Fix compiler warnings on Fedora 26 / GCC 7. ... GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88	8 years ago
Tzafrir Cohen	206d4f57dc	followme: initialize all config items on reload ... Some configuration directives were not initialized on reload, and hence were not reset to default if they were removed from followme.conf. ASTERISK-26288 #close Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150	9 years ago
Tzafrir Cohen	eec539a46e	followme: delete the right recorded name file ... FollowMe with the option a records the name of the caller and plays it to the callee. However it has failed to clean up that recorded file as it tried to delete the file name without the '.sln' extension. ASTERISK-26008 #close Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	9 years ago
Mark Michelson	4f1a8dbe92	Detect potential forwarding loops based on count. ... A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	10 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ... ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a2ce95d9d2	accountcode: Slightly change accountcode propagation. ... The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's ... Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c25c093c67	Minor tweaks with ast_moh_start() callers. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6258bbe7bd	Update Asterisk's CDRs for the new bridging framework ... This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	b6aac885be	Add dial events to app_queue and app_followme. ... Also fixes an issue in app_dial, where the channels were swapped on dial events. (closes issue ASTERISK-21551) (closes issue ASTERISK-21550) Review: https://reviewboard.asterisk.org/r/2549/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Andrew Latham	cfc6f60ca3	Doxygen Updates - Title update ... Update and extend the configuration_file group and enable linking to the application. Update title that was left behind many years ago. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	14be2a5514	Doxygen Cleanup ... Start adding configuration file linking and pages. Add module loading doxygen block. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4058ea15ab	Allow configured numbers for FollowMe to be greater than 90 characters ... When parsing a 'number' defined in followme.conf, FollowMe previously parsed the number in the configuration file into a buffer with a length of 90 characters. This can artificially limit some parallel dial scenarios. This patch allows for numbers of any length to be defined in the configuration file. Note that Clod Patry originally wrote a patch to fix this problem and received a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256 characters. Instead, the patch being committed duplicates the string in the config file on the stack before parsing it for consumption by the application. (closes issue ASTERISK-16879) Reported by: Clod Patry Tested by: mjordan patches: followme_no_limit.diff uploaded by Clod Patry (license #5138) Slightly modified for this commit. ........ Merged revisions 372390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372391 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372392 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ac35b92b62	Hangup handlers - Dialplan subroutines that run when the channel hangs up. ... Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f8746d0009	Allow non-normal execution routines to be able to run on hungup channels. ... * Make non-normal dialplan execution routines be able to run on a hung up channel. This is preparation work for hangup handler routines. * Fixed ability to support relative non-normal dialplan execution routines. (i.e., The context and exten are optional for the specified dialplan location.) Predial routines are the only non-normal routines that it makes sense to optionally omit the context and exten. Setting a hangup handler also needs this ability. * Fix Return application being able to restore a dialplan location exactly. Channels without a PBX may not have context or exten set. * Fixes non-normal execution routines like connected line interception and predial leaving the dialplan execution stack unbalanced. Errors like missing Return statements, popping too many stack frames using StackPop, or an application returning non-zero could leave the dialplan stack unbalanced. * Fixed the AGI gosub application so it cleans up the dialplan execution stack and handles the autoloop priority increments correctly. * Eliminated the need for the gosub_virtual_context return location. Review: https://reviewboard.asterisk.org/r/1984/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1ae31fd2a9	Add predial support to FollowMe. ... Like the new predial feature for Dial. This adds the same b/B options to FollowMe. Review: https://reviewboard.asterisk.org/r/1910/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b5a6de76fc	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) ... This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	108f5fafd7	Improve FollowMe accept/decline DTMF string matching. ... If you hit the wrong DTMF digit trying to accept/decline a FollowMe call, you had to wait for the prompt to repeat to try again. * Make FollowMe compare the last DTMF digits received to the accept/decline matching strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	d71d8ed995	Keep answered FollowMe calls until call accepted or last step times out. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	a689a5776e	Put winning FollowMe outgoing call on hold if the caller put it on hold. ... The FollowMe caller call leg is usually answered and listening to MOH. The caller could put the call on hold while FollowMe is looking for a winner. The winning outgoing call is now immediately placed on hold if the caller has put the call on hold before the winning call was selected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	708cadf1b1	Restructure how the FollowMe outgoing channel list is handled. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	bb5e2c48d1	Addendum to -r365766. Since it is no longer allocated. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b888b6bf23	Make FollowMe findmeexec() put the list head on the stack instead of mallocing it. ... Why this tiny struct was malloced instead of the 28k struct in the last change is beyond me. Just doing my part to help stamp out sillyness. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b1a94ddcdd	Make FollowMe app_exec() not declare a 28k struct on the stack. ... Helping to stamp out stack abuse. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	db4fb48f58	Simplify findmeexec() parameter passing. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9cd0236f61	* Fix FollowMe memory leak on error paths in app_exec(). ... * Fix FollowMe leaving recorded caller name file on error paths in app_exec(). * Use correct buffer dimension define in struct fm_args.namerecloc[]. This fixes unexpected namerecloc filename length restriction. ........ Merged revisions 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365701 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1b0428ac7d	* Fix accept/decline DTMF buffer overwrite in FollowMe. ... * Made use MAX_YN_STRING define to make all accept/decline DTMF buffers the same size. Just using 20 isn't good enough when someone didn't get the memo. * Fix stupid use of a global variable in FollowMe. (ynlongest) * Fix bit field declarations in FollowMe. ........ Merged revisions 365631 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365632 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	781f4657b9	Fix many issues from the NULL_RETURNS Coverity report ... Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Olle Johansson	a8e755700e	Use the DEFINED value for musicclass length. ... For some reason, features.c has it's own definition. Should propably be fixed too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	786f5898d1	Finalize ast_channel opaquification ... Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	a9d607a357	Opaquify ast_channel structs and lists ... Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	1fac2fba4b	Deprecated macro usage for connected line, redirecting, and CCSS ... This commit adds GoSub alternatives to connected line, redirecting, and CCSS macro hooks so that macro can finally be deprecated. This also adds deprecation warnings for those features when used and in documentation. Review: https://reviewboard.asterisk.org/r/1760/ (closes issue SWP-4256) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types ... Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields ... Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	edf466012f	Make FollowMe optionally update connected line information when the accepting endpoint is bridged. ... Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350415 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d7005bf8ad	Fix memory leaks in app_followme find_realtime(). ... (closes issue ASTERISK-19055) Reported by: Matt Jordan ........ Merged revisions 349872 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349873 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b05d4603c4	Fix crash during CDR update. ... The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348363 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	090f9d83a5	Fix FollowMe CallerID on outgoing calls. ... The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ ........ Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348102 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e218748ac1	Merged revisions 337120 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	18a6c3a415	Adds an option to FollowMe that isn't useful for the bug it was made to solve. Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields ... This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	1c60cead78	Add option to followme to delay answer until ready to bridge call. ... Followme answers an incoming call if it hasn't already been answered and starts MOH. Some poorly designed autodialers see the answer and start playing their message to the hold music. The 'N' option has been added to indicate ringing and not answer until the call is accepted. (closes issue #18479) Reported by: ianc Patches: trunk_followme.diff uploaded by ianc (license 998) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	997816819b	Merged revisions 297733 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297733 | tilghman | 2010-12-06 18:29:26 -0600 (Mon, 06 Dec 2010) | 22 lines Merged revisions 297713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8190e96fad	Merged revisions 284610 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago