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2011-07-14Merged revisions 328247 via svnmerge from lmadsen1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-03Merged revisions 316265 via svnmerge from russell1-2/+0
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: http://svn.digium.com/svn/asterisk/trunk@316293 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-18Adds an option to FollowMe that isn't useful for the bug it was made to ↵jrose1-3/+10
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: http://svn.digium.com/svn/asterisk/trunk@311427 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Asterisk media architecture conversion - no more format bitfieldsdvossel1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Add option to followme to delay answer until ready to bridge call.jpeeler1-21/+40
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: http://svn.digium.com/svn/asterisk/trunk@304384 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Merged revisions 297733 via svnmerge from tilghman1-2/+8
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: http://svn.digium.com/svn/asterisk/trunk@297734 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284610 via svnmerge from tilghman1-1/+0
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: http://svn.digium.com/svn/asterisk/trunk@284628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett1-1/+4
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve compiler warnings on FreeBSD.russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Fix app_followme playing wrong sound files.jpeeler1-7/+7
Fixes regression introduced in 140167 that uses the wrong variable names. (closes issue #16930) Reported by: ianc Patches: fix_reload_followme.diff uploaded by ianc (license 998) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250979 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Add an option to app_followme to disable the "please hold" announcement.mnicholson1-6/+12
(closes issue #14155) Reported by: junky Patches: M14555-trunk.diff uploaded by junky (license 177) (modified) Tested by: junky git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230964 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman1-4/+4
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218577 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines Ensure FollowMe sets language in channels it creates. Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@218579 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-4/+5
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-05Merged revisions 192429 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 lines Fix a bug where the followme application would continue trying numbers after the caller hung up. (closes issue #13624) Reported by: sgenyuk ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192430 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-29Merged revisions 184842 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184843 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-07 Answer the channel if it has not already been answered and we've already ↵bweschke1-0/+5
found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167478 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-13Merge ast_str_opaque branch (discontinue usage of ast_str internals)tilghman1-3/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163991 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-25Make the options for the general and profiles more consistentmmichelson1-1/+1
for the "pls_hold_prompt" option. This does not affect any released version of Asterisk, so there is no need to update the CHANGES file for this. (closes issue #13893) Reported by: eliel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159250 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-18Merged revisions 157305 via svnmerge from mmichelson1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157306 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-09Merged revisions 155553 via svnmerge from seanbright1-21/+22
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155554 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05- Add FollowMe() application XML documentation.eliel1-15/+36
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154469 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31Recent CDR fixes moved execution of the 'h' exten into the bridging code, so ↵twilson1-9/+22
variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153181 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-1/+5
when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-26OpenBSD compat fix (reminded by mvanbaak on #asterisk-dev)tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140201 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-26Standardize the option names for consistency (but continue to work with thetilghman1-20/+33
existing names for backwards compatibility). (closes issue #13370) Reported by: jsturtevant git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140167 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-25Realtime capabilities for the Find-Me-Follow-Me application.tilghman1-6/+78
(closes issue #13295) Reported by: Corydon76 Patches: 20080813__followme_realtime_enabled.diff.txt uploaded by Corydon76 (license 14) Tested by: dferrer git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139775 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causemvanbaak1-2/+2
- 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
2008-04-25Whitespace changes onlytilghman1-1/+1
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.mvanbaak1-0/+3
(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
2008-03-13Merged revisions 108469 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines Fix a couple uses of sprintf. The second one could actually cause an overflow of a stack buffer. It's not a security issue though, it only depends on your configuration. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@108472 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Merged revisions 106235 via svnmerge from file1-0/+3
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 silencethresholdtilghman1-72/+64
(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
2008-02-05Get 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 99594 via svnmerge from oej1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 lines Add more dependencies on chan_local and add a note to the description of chan_local so that people don't disable it in menuselect just to clean up. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99596 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Merged revisions 98219 via svnmerge from file1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution. (closes issue #10327) Reported by: kkiely ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98220 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 theyrizzo1-1/+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
2007-11-08improve linked-list macros in two ways:kpfleming1-7/+3
- the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06"show application <foo>" changes for clarity.mmichelson1-1/+1
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-04Merged revisions 81455 via svnmerge from qwell1-5/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10634) ........ r81455 | qwell | 2007-09-04 15:54:51 -0500 (Tue, 04 Sep 2007) | 4 lines Rather than attempt to play a file, we can just check whether it exists. Issue 10634, patch by me, testing by pabelanger, sanity checked by bweschke ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81456 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Don't reload a configuration file if nothing has changed.tilghman1-5/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵tilghman1-1/+1
few other formatting cleanups, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-30Minor clean up of app_followme.file1-387/+396
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77773 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-26Do a massive conversion for using the ast_verb() macrorussell1-44/+23
(closes issue #10277, patches by mvanbaak) Basically, this changes ... if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, "Something\n"); git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77299 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-5/+0
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-23 22:45:45 +0000