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2009-03-25Fix build issues on Mac OSX.russell2-4/+9
(closes issue #14714) Reported by: ygor git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184147 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-24Merged revisions 184078 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184079 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-24Put siren7 and siren14 in ast_best_codec() just so they're in there somewhere.russell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184043 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-24Exclude slin16, siren7, and siren14 from bandwidth=low and =mediumrussell1-0/+3
The default codec configuration for chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as the codec in some test calls, but that no longer happens after this change. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184037 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-24SIP preferred codec only featuredvossel3-3/+22
Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use. (closes issue #12485) Reported by: bamby Review: http://reviewboard.digium.com/r/206/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183995 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-24Merged revisions 183913 via svnmerge from tilghman1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183914 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-23Allow browsers to cache images and other static content.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183865 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-23Removed trailing whitespace in chan_misdn files.rmudgett8-1298/+1297
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183831 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-23Merged revisions 183700 via svnmerge from mmichelson1-5/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines Fix a memory leak in res_monitor.c The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183766 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-23Fixes a documentation error introduced during the CLI cleanup at AstriDevCon ↵lmadsen1-1/+1
2008. (closes issue #14655) Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) Tested by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183701 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-22Fix a minor logic flaw with the bridge generic thread.file1-1/+1
We only want to move the channel pointers that are actually present. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183652 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-20Merged revisions 183559 via svnmerge from russell1-5/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines Fix a crash in IAX2 registration handling found during load testing with dvossel. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183560 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-20Fix chan_sip so it builds.mmichelson1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183555 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-20Remove symbols I just added to main/asterisk.exports and instead rename the ↵mmichelson3-6/+4
functions. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183554 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-20Add some missing symbols to main/asterisk.exportsmmichelson1-0/+2
Hey! chan_sip.so loads now! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183553 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-20Remove duplicate <description> inside the xml documentation.eliel1-3/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183511 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183386 via svnmerge from dvossel3-8/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183436 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Recorded merge of revisions 183342 via svnmerge from tilghman0-0/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183342 | tilghman | 2009-03-19 14:21:30 -0500 (Thu, 19 Mar 2009) | 2 lines Reordering, to change prior to unlocking ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183345 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183319 via svnmerge from tilghman1-0/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183321 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183291 via svnmerge from qwell1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183291 | qwell | 2009-03-19 13:28:16 -0500 (Thu, 19 Mar 2009) | 1 line Export some more required symbols. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183312 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Fix a memory leak associated with queues.mmichelson1-13/+14
For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183244 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183241 via svnmerge from russell4-103/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183242 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183238 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183238 | russell | 2009-03-19 12:41:39 -0500 (Thu, 19 Mar 2009) | 1 line Allow the AES API to work. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183239 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-192 symbols defined when DEBUG_THREADStilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183196 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183126 via svnmerge from dvossel3-54/+125
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183172 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183145 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183145 | russell | 2009-03-19 11:21:56 -0500 (Thu, 19 Mar 2009) | 1 line Add missing semicolon in exports script. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183148 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183123 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183123 | russell | 2009-03-19 11:13:18 -0500 (Thu, 19 Mar 2009) | 2 lines Allow the CallerID API to work again. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183124 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183115 via svnmerge from mmichelson1-5/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183117 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Improve our triggering of a T38 switchover internally when triggered by a ↵file1-184/+41
received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183108 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Fix an issue where a T38 control frame would get dropped.file1-0/+1
If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183057 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18allow this module to export everything for nowkpfleming1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183032 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Add some code removed by mistake from commit 182722 that works around a filejpeeler1-1/+10
descriptor leak in versions of PWLib prior to 1.12.0. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183028 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Blocked revisions 182965 via svnmergejpeeler0-0/+0
........ r182965 | jpeeler | 2009-03-18 15:02:40 -0500 (Wed, 18 Mar 2009) | 1 line fix typo which broke configure ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182966 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Blocked revisions 182963 via svnmergejpeeler0-0/+0
........ r182963 | jpeeler | 2009-03-18 14:57:05 -0500 (Wed, 18 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue 0011261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182964 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Fixing a lost symbol in manager.ctilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182960 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 182882 via svnmerge from kpfleming3-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182882 | kpfleming | 2009-03-18 06:31:41 -0500 (Wed, 18 Mar 2009) | 3 lines fix another symbol namespace issue (reported by Andrew on asterisk-dev) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182883 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18a few more namespace updates... res_ael_share still needs some work before ↵kpfleming7-6/+12
this can be merged to other release branches git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182848 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 182810 via svnmerge from russell17-57/+204
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182847 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 182808 via svnmerge from kpfleming19-191/+321
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar 2009) | 5 lines Improve the build system to *properly* remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix. With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182826 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Add support for the "name" option in the CHANNEL() function.russell2-2/+11
Review: http://reviewboard.digium.com/r/199/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182762 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Allow H.323 Plus library to be used in addition to the OpenH323 libraryjpeeler8-268/+156
Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182722 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Blocked revisions 182652 via svnmergeqwell0-0/+0
........ r182652 | qwell | 2009-03-17 15:13:40 -0500 (Tue, 17 Mar 2009) | 7 lines Allow dahdichanname to work as advertised. (closes issue #14056) Reported by: dsedivec Patches: load_from_zapata_conf.patch uploaded by dsedivec (license 638) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182653 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Fixing CHANGES in rev 182596.dvossel1-4/+5
Progress DTMF was added into app_dial's D() option. In CHANGES it should have been updated under 1.6.3 rather than 1.6.2. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182607 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Option to send DTMF when receiving PROGRESS statusdvossel2-8/+17
The D() option in app_dial is only able to send DTMF after the call has been answered. A progress option has been added to D() to allow DTMF to be sent upon receiving PROGRESS. This allows DTMF to be sent before the call is answered. (closes issue #12123) Reported by: VoipForces Patches: app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419) dtmf_progress.patch uploaded by dvossel (license 671) Tested by: VoipForces, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182596 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Tweak the handling of the frame list inside of ast_answer().russell1-3/+2
This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182553 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17correct logic flaw in ast_answer() changes in r182525kpfleming1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182530 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Improve behavior of ast_answer() to not lose incoming frameskpfleming3-36/+151
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182525 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Don't include a space before the optional extra text that may follow a helpseanbright1-1/+1
string. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182521 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Merged revisions 182449 via svnmerge from tilghman1-3/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182450 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17OPENR2 uses an incorrect string value if the extension delimiter is not present.rmudgett1-35/+36
* Fixed OPENR2 using an incorrect string value if the extension delimiter is not present in the Dial() function. This was fixed for SS7 and PRI in trunk -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others. * Removed trailing whitespace that appeared with OPENR2. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182408 f38db490-d61c-443f-a65b-d21fe96a405b
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