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2008-12-13Merge ast_str_opaque branch (discontinue usage of ast_str internals)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-13Only detach and destroy the whisper audiohooks if they are actually in use.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-12When using realtime queues, app_queue wasn't updating the strategy if it was ↵Terry Wilson
changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered. (closes issue #14034) Reported by: cristiandimache Tested by: otherwiseguy, cristiandimache git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-11Add an option to voicemail.conf to allow urgent messages to beMark Michelson
forwarded as not urgent. (closes issue #14063) Reported by: jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-11Merged revisions 163084 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines Revert this cast to long. Using time_t here causes build failures on a FreeBSD 32-bit build. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-11Merged revisions 163080 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines Fix a potential crash due to unsafe datastore handling. This patch also contains a conversion from using long to time_t for representing times for a queue, as well as some whitespace fixes. (closes issue #14060) Reported by: nivek Patches: datastore_fixup.patch.corrected uploaded by nivek (license 636) with slight modification from me Tested by: nivek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-10Finish conversion to using ARRAY_LEN and remove it as a janitor project.Joshua Colp
(closes issue #14032) Reported by: bkruse Patches: 14032.patch uploaded by bkruse (license 132) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-09Merged revisions 162463 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-09Merged revisions 162348 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines We appear to have documented tz= in the [general] section of voicemail.conf, without actually having implemented it. Oops. (Reported by Olivier on the -users list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-09Merged revisions 162341 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing. (closes issue #14005) Reported by: ddl ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-09Merged revisions 162286 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback. We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-09Merged revisions 162273 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines Fix double declaration of 'x' on the PPC platform. (closes issue #14038) Reported by: ffloimair ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-09Merged revisions 162014 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines Allow DISA to handle extensions that start with #. (closes issue #13330) Reported by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-08Add voicemail related applications and functions XML documentation:Eliel C. Sardanons
applications: - VoiceMail() - VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: - MAILBOX_EXISTS() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-07Introduce SMS() application XML documentation.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-06Move Speech* applications and functions documentation to XML.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05If the autoloop flag is set on a channel, then we need to Mark Michelson
add 1 to the priority when checking if the extension exists. Otherwise, gosubs will fail. This was discovered when investigating an asterisk-users mailing list post made by Gary Hawkins. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05Use ast_free() instead of free(), pointed out by eliel on IRC.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05When using IMAP_STORAGE, it's important to convert bare newlines (\n) inSean Bright
emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed by Mark M. on IRC. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05Resolve a compiler warning from buildbot about a NULL format string.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05Janitor, use ARRAY_LEN() when possible.Eliel C. Sardanons
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05Check the return value of fread/fwrite so the compiler doesn't complain. Only aSean Bright
problem when IMAP_STORAGE is enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03Merged revisions 160770 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines Some compilers warn on null format strings; some don't (caught by buildbot) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03Add some safety measures when using gosub, especially when using the optionsMark Michelson
for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03- Add <variable /> tags when naming a channel variable.Eliel C. Sardanons
- Add <filename /> tags when naming a filename. - Simplify the xml formatting putting some enters. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03When investigating issue #13548, I found that gosubMark Michelson
handling in app_queue was just completely wrong, mostly because the channel operations being performed were being done on the incorrect channel. With this set of changes, a gosub will correctly run on the answering queue member's channel. There are still crash issues which occur if there are dialplan syntax errors, so I cannot yet close the referenced issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03- Avoid setting .synopsis and .syntax if we are using XML documentation (or theEliel C. Sardanons
xml documentation wont be loaded). - Use <variable></variable> to refer to a dialplan variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02Add LOCAL_PEEK function, as requested by lmadsen.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02Merged revisions 160207 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-29Allow the '#' sign to exist within an extension (inspired by issue #13330)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-26improve handling of API calls provided by loaded modules through use of some ↵Kevin P. Fleming
GCC features; this makes app_stack's usage of AGI APIs even cleaner, and will allow it to work 'as expected' either with or without res_agi being loaded reviewed at http://reviewboard.digium.com/r/62 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-26Add some necessary hangup commands in the case that forwardingMark Michelson
a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25Make the options for the general and profiles more consistentMark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@159250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25Add missing variable declaration for PPC codeTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25Copyright clarification; also, have variable set to "t" or "i" on timeout orTilghman Lesher
invalid extension, respectively. (closes issue #13944) Reported by: chappell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25Merged revisions 159025 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25This is basically a complete rollback of r155401, as it was determined thatSean Bright
it would be best to maintain API compatibility. Instead, this commit introduces ao2_callback_data() which is functionally identical to ao2_callback() except that it allows you to pass arbitrary data to the callback. Reviewed by Mark Michelson via ReviewBoard: http://reviewboard.digium.com/r/64 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25Make the Join event from app_queue use CallerIDNum insead of CallerID forMatthew Nicholson
indicating the callerid number just like the rest of asterisk. (closes issue #13883) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-24This patch adds a new application for sending MWI to phones via Asterisk's ↵Terry Wilson
event subsystem. Also, the minivm documentation is all converted to use xmldocs. (closes issue #13946) Reported by: Marquis Patches: minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32) Tested by: otherwiseguy, Marquis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-20Merged revisions 158053 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19Add a space to the outputMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19Add a RES_NOT_DYNAMIC case for the CLI command Mark Michelson
'queue remove member' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19make some corrections to the ast_agi_register_multiple(), ↵Kevin P. Fleming
ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-18Fix the logic for when delete=yes when IMAP storageMark Michelson
is in use so that the message is deleted from both local and IMAP storage. (closes issue #13642) Reported by: jaroth Patches: deleteyes.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-18Merged revisions 157365 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines (closes issue #13899) Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-18Merged revisions 157305 via svnmerge from Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@157306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-17Can't use items duplicated off the stack frame in an element returned fromTilghman Lesher
a function: in these cases, we have to use the heap, or garbage will result. (closes issue #13898) Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-14Fix some refcounting in app_queue.c and change theMark Michelson
hashing used by app_queue.c to be case-insensitive. This is accomplished by adding a new case-insensitive hashing function. This was necessary to prevent bad refcount errors (and potential crashes) which would occur due to the fact that queues were initially read from the config file in a case-sensitive manner. Then, when a user issued a CLI command or manager action, we allowed for case-insensitive input and used that input to directly try to find the queue in the hash table. The result was either that we could not find a queue that was input or worse, we would end up hashing to a completely bogus value based on the input. This commit resolves the problem presented in issue #13703. However, that issue was reported against 1.6.0. Since this fix introduces a behavior change, I am electing to not place this same fix in to the 1.6.0 or 1.6.1 branches, and instead will opt for a change which does not change behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-14Merged revisions 156816 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines If the prompt to reenter a voicemail password timed out, it resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-14Merged revisions 156755 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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