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${ noResults }
1853 Commits (dea550a29252aa6ef4f67f323d90eb29c3e8c794)
| Author | SHA1 | Message | Date |
|---|---|---|---|
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|
2f24689b49 |
Merged revisions 180372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180373 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
4c9ab0df8c |
Merge phase 1 support for the new bridging architecture.
This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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979eb709ae |
app_read does not break from prompt loop with user terminated empty string
In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4ac2fd4cde |
Use notification when timezone files change and re-scan then.
(closes issue #14300) Reported by: jamessan Patches: 20090127__bug14300.diff.txt uploaded by tilghman (license 14) 20090224__bug14300.diff uploaded by jamessan (license 246) Tested by: jamessan Review: http://reviewboard.digium.com/r/136/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178605 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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935185ce8a |
Trailing whitespace, minor coding guideline fixes, and start beefing up the
hashtab documentation a bit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177884 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7dc56a0c27 |
Fix build issues on Solaris and OpenBSD.
(closes issue #14512) Reported by: snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9967ef01df |
Merged revisions 177701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177732 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3af1c558df |
Allow semicolons to be escaped, when passing arguments to the System command.
(closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177664 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ef84acf002 |
Fix another merge error from 176708
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a1f583177e |
ODBC transaction support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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39d573920f |
Merged revisions 177096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines Document the return value of the update method (as requested on -dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b5410fad00 |
Add example code for a heap traversal.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f40edf2793 |
Merged revisions 176701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6d60de7efa |
Clear up documentation of AST_FRIENDLY_OFFSET in frame.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c461d29b0b |
Update the timing API to have better support for multiple timing interfaces.
1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bb03ef8d47 |
Add an implementation of the heap data structure.
A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176632 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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47913cab6d |
Typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4ec301360c |
Merge a large set of updates to the Asterisk indications API.
This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0381d94d14 |
Merged revisions 176216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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115c6abef4 |
Merged revisions 175921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8c75380f52 |
Make ast_sched_report() and ast_sched_dump() thread safe.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ca9d3b8ac9 |
Fix a number of problems with ast_sched_report().
1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2a53f2ec98 |
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3a9d79f056 |
Make lock information for ao2_trylock be more useful and gnarly
Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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47ebea6a8d |
Fix 'd' option for app_dial and add new option to Answer application
The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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23939e54f3 |
improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0cbada4f96 |
Add a common implementation of a scheduler context with a dedicated thread.
This commit expands the Asterisk scheduler API to include a common implementation of a scheduler context being processed by a dedicated thread. chan_iax2 has been updated to use this new code. Also, as a result, this resolves some race conditions related to the previous chan_iax2 scheduler handling. Related to rev 171452 which resolved the same issues in 1.4. Code from team/russell/sched_thread2 Review: http://reviewboard.digium.com/r/129/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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39ec5d1576 |
Merged revisions 173211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e179e613f7 |
1. Make OS X compile cleanly with app_stack.
2. Use curl to download sound files, as curl is installed natively on OS X, whereas wget and fetch are not. (closes issue #14332) Reported by: oej Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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53d9b77898 |
This reverts the changes I made for 11583; will
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17 years ago |
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c61e8a7865 |
This change allows the disconnect feature (as in "one-touch" in features.c)
to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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458dfe4748 |
Fix redefinition of flag in channel.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8d782f96b8 |
Merged revisions 172517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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14cebb2c38 |
Fixed some doxygen comments
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7ecda45482 |
Fix "cancel answered elsewhere" through app_queue with members in chan_local.
Also, implement a private cause code (as suggested by Tilghman). This works with chan_sip, but doesn't propagate through chan_local. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b79a12e929 |
- Make sure we set setvar= variables on outbound calls too, not only inbound calls.
- Also, change a function in app.c to return a userful value instead of always returning 0. Patch by fnordian, changed by Corydon76 and myself. This does not close the bug report, as fnordian had an additional change we're still discussing. (related to issue #14059) Reported by: fnordian Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) Tested by: fnordian, Corydon76, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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268ac221a2 |
Merged revisions 172030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6101eccc9f |
Change ARRAY_LEN() to be more C++ safe.
When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170943 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9a28a07739 |
change VMWI to use new DAHDI_VMWI ioctl call.
Change configure script to detect the new ioctl call data structure. (issue #14104) Reported by: alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded by dbailey (license ) Tested by: alecdavis, dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e3b431ebcb |
Merged revisions 169943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is whether autoconf detected a static initializer value. This fixes rwlocks on all such platforms (mainly, Mac OS X). (closes issue #13767) Reported by: jcovert Patches: 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) Tested by: jcovert, Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1c2911f5a1 |
ast_str_SQLGetData is *not* part of the ast_str API, it's part of the ast_odbc API and just happens to use an ast_str as the buffer; move all of it to res_odbc.c and res_odbc.h, renaming appropriately
along the way fix some minor coding style issues in strings.h and add some attribute_pure annotations to functions in the ast_str API git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e0b40036e1 |
Fix qualify for TCP peer
(closes issue #14192) Reported by: pabelanger Patches: asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169080 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3728c3aa92 |
Merged revisions 168828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c6cb67b941 |
Resolve issue with negative vs non-negative length parameters.
(closes issue #14245) Reported by: dveiga git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7c6d9c7235 |
Add option to hide console connect messages
(closes issue #14222) Reported by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176) Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ef6ad2b53c |
Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fd3cb90841 |
Some platforms (notably, the BSDs) have a more efficient implementation called
closefrom(3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4a9e8078b9 |
When using ast_str with a non-ast_str-enabled API, we need to update the buffer
or otherwise, we cannot use ast_str_strlen(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e1f941d7f6 |
Mostly just whitespace, but also convert 'CVS' to 'SVN' in a couple
places and fix a few typos I found in the CODING_GUIDELINES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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aa905e347e |
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |