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2008-07-01another minor ast_channel memory size decrease... for nearly all channels, ↵kpfleming1-6/+3
'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126960 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-26Improve consistency between app_dial and app_queue with regardsmmichelson1-2/+0
to how language is handled between two channels whose native language is different. Prior to this patch, app_dial would have the callee inherit the caller's language, and app_queue would not. After this patch, app_dial no longer has the language inheritance capability. This seems to make the most sense since it seems more natural for a person to hear files played back in his/her native language instead of the language of the person on the far end of the call. See the CHANGES file for hints on how to keep the previous behavior of app_dial if desired. (closes issue #12489) Reported by: bcnit git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-18Channel lock janitor -- add locks around retrieval of channel variablestilghman1-13/+43
(closes issue #12840) Reported by: pputman Patches: app_dial_threadsafe3.patch uploaded by pputman (license 81) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123648 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-16(closes issue #12689)murf1-1/+1
Reported by: ys Many thanks to ys for doing the research on this problem. I didn't think it would be best to unlock the contexts and then relock them after the remove_extension2() call, so I added an extra arg to remove_extension2() and set it appropriately in each call. There were not that many. I considered forcing the code to lock the contexts before the call to remove_extension2(), but that would require a slightly greater degree of changes, especially since the find_context_locked is local to pbx.c I did a simple sanity test to make sure the code doesn't mess things up in general. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123165 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵jpeeler1-5/+5
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-02Merged revisions 119530 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines Fix another typo in documentation ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119531 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-01Merged revisions 119478 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines small typo fix 'retires' => 'retries' ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119479 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-30Add native AGI command GOSUB, as invoking Gosub with EXEC does not worktilghman1-4/+4
properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119296 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causemvanbaak1-5/+6
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Modify TIMEOUT() to be accurate down to the millisecond.tilghman1-3/+4
(closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-28Add incomplete matching to PBX code and app_dialtilghman1-3/+10
(closes issue #12351) Reported by: Corydon76 Patches: 20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76 (license 14) pbx_incomplete_with_timeout.diff uploaded by fabled (license 448) Tested by: Corydon76, fabled git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114773 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-25Whitespace changes onlytilghman1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114667 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-24Pass the hangup cause all the way to the calling app/channel.mvanbaak1-1/+4
(closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114637 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-14Merged revisions 114112 via svnmerge from mmichelson1-2/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines If the datastore has been moved to another channel due to a masquerade, then freeing the datastore here causes an eventual double free when the new channel hangs up. We should only free the datastore if we were able to successfully remove it from the channel we are referencing (i.e. the datastore was not moved). (closes issue #12359) Reported by: pguido ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114113 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-09Permit callee to continue in the dialplan, after caller has hung up.tilghman1-5/+16
(closes issue #11954) Reported by: johan Patches: app_dial_rev104031.patch uploaded by johan (license 334) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@113649 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Remove some redundant logic from wait_for_answer. This also let's us get rid ↵mmichelson1-13/+15
of one of those XXX comments from the code. The redundancy occurs because the 'single' flag implies that the 'r' and 'm' flags are not set, so there's no need to explicitly check them again. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107530 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10Merged revisions 107016 via svnmerge from file1-17/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107017 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07(closes issue #6002)murf1-3/+1
Reported by: rizzo Tested by: murf Proposal of the changes to be made, and then an announcement of how they were accomplished: http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html and: http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html Here is a recap, file by file, of what I have done: pbx/pbx_config.c pbx/pbx_ael.c All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set. Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it is just as necessary to have the TABLE available. This is because the list/table in question might not be the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global position when things are ready. We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing "find" and "create", as all existing usages used both in tandem anyway. pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and then call merge_contexts_and_delete, which will merge (now) existing contexts and priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will lock down the contexts, swap the lists and tables, and unlock (real quick), and then destroy the old dialplan. chan_sip.c chan_iax.c chan_skinny.c All the channel drivers that would add regcontexts now use the ast_context_find_or_create now. chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered. apps/app_meetme.c apps/app_dial.c apps/app_queue.c All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead. include/asterisk/pbx.h ast_context_create() is removed. Find_or_create_ is the new method. ast_context_find_or_create() interface gets the hashtab added. ast_merge_contexts_and_delete() gets the local hashtab arg added. ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking. ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael ast_hashtab_hash_contexts was in like fashion make public. include/asterisk/pval.h ast_compile_ael2() interface changed to include the local hashtab table ptr. main/features.c For the sake of the parking context, we use ast_context_find_or_create(). main/pbx.c I changed all the "tree" names to "table" instead. That's because the original implementation was based on binary trees. (had a free library). Then I moved to hashtabs. Now, the names move forward too. refcount field added to contexts, so you can keep track of how many modules wanted this context to exist. Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING. Added some calls to ast_verb(3,...) for debug messages Lots of little mods to ast_context_remove_extension2, which is now excersized in ways it was not previously; one definite bug fixed. find_or_create was upgraded to handle both local lists/tables as well as the globals. context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables ast_merge_contexts_and_delete() was heavily modified. ast_add_extension2() was also upgraded to handle changes. the context_destroy() code was re-engineered to handle the new way of doing things, by exten/prio instead of by context. res/ael/pval.c res/ael/ael.tab.c res/ael/ael.tab.h res/ael/ael.y res/ael/ael_lex.c res/ael/ael.flex utils/ael_main.c utils/extconf.c utils/conf2ael.c utils/Makefile Had to change the interface to ast_compile_ael2(), to include the hashtab ptr. This ended up involving several external apps. The main gotcha was I had to include lock.h and hashtab.h in several places. As a side note, I tested this stuff pretty thoroughly, I replicated the problems originally reported by Luigi, and made triply sure that reloads worked, and everything worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into trunk, that did not appear in my tests of bug6002. How's this for verbose commit messages? git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106757 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Merged revisions 106235 via svnmerge from file1-1/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106239 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Create a centralized configuration option for silencethresholdtilghman1-1/+4
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106072 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-01Asterisk, when parking can drop rights a caller when a parking timeout ↵twilson1-0/+64
occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue. (closes issue #11520) Reported by: pliew Tested by: otherwiseguy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105477 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-09whitespace fixes only.mvanbaak1-179/+170
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103249 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-05Get rid of any remaining ast_verbose calls in the code in favor of mmichelson1-4/+2
ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@102525 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Merged revisions 99592 via svnmerge from oej1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5 lines Add dependency on chan_local to app_dial. Dial still runs without chan_local, but will be missing forwarding functionality. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99595 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Convert ast_verbose to ast_verb.tilghman1-14/+9
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92913 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-12Conversions of free to ast_free, where applicable, and several other ↵tilghman1-81/+83
formatting fixes. Reported by: eliel Patch by: eliel,tilghman (Closes issue #11209) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92594 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Merged revisions 91783 via svnmerge from russell1-26/+43
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | 6 lines * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Remove the dialed variable as it isn't needed. * Restructure some code for clarity and coding guidelines stuff ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91784 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Merged revisions 91693 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) | 2 lines Don't unlock the dialed_interfaces list until we're done messing with the iterator. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91700 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Merged revisions 91677 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) | 4 lines Allow dialing local channels from Queue() and Dial() again. There was a slight flaw in the code to prevent call forwards from looping that caused this problem. (related to issue #11486) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91678 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-06- Dial eventoej1-8/+10
- Event Dial has new headers, to comply with other events - Source -> Channel Channel name (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring Dialstring in app data (moremanager) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91407 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Merged revisions 91273 via svnmerge from mmichelson1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec 2007) | 11 lines The 'G' option for Dial() did not properly handle the case where only a label was provided. This was due to the fact that the answering channel did not have an extension set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto on the answering channel since it is a wasteful call. The answering channel and the calling channel are both directed to the same extension and context, just different priorities, so we can just copy the values from the calling channel to the answering channel and increment the answering channel's priority. (closes issue #11382, reported by jon, patch by me with correction by jon) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91291 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this ↵qwell1-1/+1
hadn't been merged yet. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90877 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Merged revisions 90735 via svnmerge from mmichelson1-69/+75
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90873 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03Remove the file descriptors from the main poll channel when the channel is ↵file1-2/+7
hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end. (closes issue #11441) Reported by: blitzrage git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90508 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Adding support for the "automixmonitor" dial and queue options.mmichelson1-4/+19
This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merged revisions 89622 via svnmerge from murf1-1/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89623 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21closes issue #11285, where an unload of a module that creates a dialplan ↵murf1-1/+4
context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89513 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20move asterisk/paths.h outside asterisk.h and into those filesrizzo1-0/+1
who really need it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-2/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+0
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-14Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor ↵russell1-2/+2
happy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89264 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01This commits the performance mods that give the priority processing engine ↵murf1-7/+7
in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11Make sure we propogate ANI2 to the outbound channelmattf1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85499 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-09Remove redundant includes (patch by snuffy) (Closes issue #10922)tilghman1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85140 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-01Merged revisions 84166 via svnmerge from russell1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) | 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84167 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-01Merged revisions 84158 via svnmerge from file1-5/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4 lines Only attempt early bridging if the options given to Dial() permit it. (closes issue #10861) Reported by: peekyb ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84159 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-17Make the MALLOC_DEBUG output for free() useful again. After changing calls torussell1-1/+1
free to be ast_free, astmm said all calls to free were coming from utils.h git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82628 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-31Merged revisions 81412 via svnmerge from qwell1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500 (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line with the existing alpha order. Issue 10621, initial patch by junky ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81413 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Add support for using epoll instead of poll. This should increase ↵file1-1/+13
scalability and is done in such a way that we should be able to add support for other poll() replacements. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78683 f38db490-d61c-443f-a65b-d21fe96a405b