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2009-10-01Remove ability to control T.38 FAX error correction from udptl.conf.Kevin P. Fleming
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Use rtp properties instead of adding a callbackTerry Wilson
Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Merged revisions 221086 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Merged revisions 221200 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-29Fix channel reference leak.Mark Michelson
ast_cel_report_event would geet a reference to the bridged channel. However, certain return paths, such as if CEL was not enabled, would result in a reference leak. All return paths now properly unref the channel. (closes issue #15991) Reported by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-29Get rid of annoying and cryptic debug messages.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-25Eliminate unnecessary include of version.h in manager.c.Kevin P. Fleming
Including version.h here causes this file to get recompiled after every commit or update, which is not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-25Correct sense of logic test committed in revision 220494.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-25Don't use hash-based lookups for ast_channel_get_by_name_prefix().Kevin P. Fleming
ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based channel lookups, but this will not work properly when the channel's full name was not supplied; for name-prefix searches, there is no value in doing a hash-based lookup, and in fact doing so could result in many channels being skipped. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by ↵Tilghman Lesher
default (starting in 1.6.3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24fixes tcptls_session memory leak caused by ref count errorDavid Vossel
(closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24Add bridge related dial flags to the bridge appJeff Peeler
Most of the functionality here is gained simply by setting the feature flag on the bridge config. However, the dial limit functionality has been moved from app_dial to the features code and has been made public so both app_dial and the bridge app can use it. (closes issue #13165) Reported by: tim_ringenbach Patches: app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24Merged revisions 220288 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-20Merged revisions 219653 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-17Merged revisions 219136 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-16Merged revisions 219023 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-16Merged revisions 218867 via svnmerge from David Brooks
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines Fixes CID pattern matching behavior to mirror that of extension pattern matching. Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-14Do not attempt to add a parking extension if an error occurred while reading ↵Joshua Colp
the configuration. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-11Check the origination priority for more matches, not the current priority.Tilghman Lesher
Found by Pavel Troller on the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-09Properly terminate the response to the manager Ping action.Sean Bright
In passing, correct the formatting of the Timestamp attribute so that there is a space after the colon and before the value. (closes issue #15861) Reported by: Ivan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04Enable turning off the application delimiter warning with the 'dontwarn' option.Tilghman Lesher
Suggested on the -dev list, and implemented in an alternate way by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04Merged revisions 216435 via svnmerge from Michiel van Baak
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04Merged revisions 216430 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04make sure 'start' is always initialized.Michiel van Baak
Makes asterisk compile with --enable-dev-mode git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Document language prompt submission process.Kevin P. Fleming
This patch adds a document describing the language prompt submission process, licensing terms and other issues related to that process. In addition, it modifies the sound file searching process to support language codes with any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts can be named with gender, customer/company, etc. suffices as well. (closes issue #15771) Reported by: jtodd Patches: language-criteria.txt uploaded by jtodd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Merge code associated with AST-2009-006David Vossel
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-02- lock channel before looking for a channel variableMichiel van Baak
- Init the parkings list member of struct parkinglot. Thanks Sean for the explanation why this should be here. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-02Close up to the soft open file limit (same on Linux, but varies drastically ↵Tilghman Lesher
on OS X). Also, a Makefile fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches: 20090901__issue14542.diff.txt uploaded by tilghman (license 14) Tested by: jtodd, tilghman Change-type: bugfix git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-01Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS frames are properlyKevin P. Fleming
decoded. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-31Fix a trunk compilation warning.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-31Properly initialize the session to prevent a crash.Tilghman Lesher
(closes issue #15774) Reported by: lasko Patches: 20090831__issue15774.diff.txt uploaded by tilghman (license 14) Tested by: lasko git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-30Various patches, to enable Asterisk to once again compile on Mac OS X.Tilghman Lesher
One note on defining _POSIX_C_SOURCE: while this feature test macro works to require certain behaviors on Linux, it works differently on *BSD platforms to REMOVE certain API calls that are not in the POSIX specification, such as vasprintf(3). Thus, defining it while depending upon vasprintf (and other extensions to the POSIX standard) to be defined is a recipe to ensure that Asterisk is only buildable on Linux. Hence, this define which was meant to INCREASE portability, effectively ensures the opposite. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-28Merged revisions 214701 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines Modify comment to be a bit more accurate. We have kept this comment around long enough, that it's pretty clear that we're keeping the code, because changing the code would require a pretty fundamental architectural shift. We've also taken criticism in some quarters, because it was believed that it was referring to the code being nasty. No, the code isn't nasty, just the operation itself is rather odd. Fixed for eternity (probably not). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-27Ensure that we check for the special value CONFIG_STATUS_FILEINVALID.Tilghman Lesher
(closes issue #15786) Reported by: a_villacis Patches: asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch uploaded by a villacis (license 660) (Plus a few of my own, to catch the remaining places within manager.c where it could have been a problem) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-26Add two new dialplan variables when using featuresJeff Peeler
Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the unique channel name on the other side and is set when a dynamic feature is triggered. (closes issue #14663) Reported by: tamiel Patches: 20090313_features.diff uploaded by tamiel (license 712) Tested by: tamiel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-26Merged revisions 214194 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) | 19 lines ast_write() ignores ast_audiohook_write() results In ast_write(), if a channel has a list of audiohooks, those lists are written to and the resulting frame is what ast_write() should continue with. The problem was the returned audiohook frame was not being handled at all, and the original frame passed into it did not contain the mixed audio, so essentially audio was being lost. One result of this was chan_spy's whisper mode no longer worked. To complicate the issue, frames passed into ast_write may either be a single frame, or a list of frames. So, as the list of frames is processed in the audiohook_write, the returned frames had to be added to a new list. (closes issue #15660) Reported by: corruptor Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-25Merged revisions 214068-214069 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) | 6 lines Fix pronunciation of German dates. (closes issue #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded by Benjamin Kluck (license 803) ........ r214069 | tilghman | 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should always compile before committing... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-25Merged revisions 213970 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) | 7 lines Improve error message by informing user exactly which function is missing a parethesis. (closes issue #15242) Reported by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-21Make LOAD_ORDER actually workTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-20Fix a crash by checking the proper pointer for validity before deferencing it.Matthew Nicholson
(closes issue #15751) Reported by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license 242) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-19Fix compile when certain G711 menuselect options are enabled.Jason Parker
(closes issue #15697) Reported by: slavon git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-19Don't blow up on a NULL cdr.Russell Bryant
Reported in #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-18Merged revisions 212763 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug 2009) | 11 lines Delay the creation of temporary files until we have a valid manager command to handle. Without this patch, asterisk creates a temporary file before determining if the specified command is valid. If invalid, we weren't properly cleaning up the file. (closes issue #15730) Reported by: zmehmood Patches: M15730.diff uploaded by junky (license 177) Tested by: zmehmood ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-17Correct the return value check for ast_safe_system.Sean Bright
The logic here was reversed as ast_safe_system returns -1 on error and not on success. Fix suggested by reporter. (closes issue #15667) Reported by: loic git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-17Define our desires for POSIX and X/OPEN API features properly.Kevin P. Fleming
Based on a post on the gcc-help mailing list and some subsequent reading, we can increase our portability to various platforms by directly defining the POSIX and X/OPEN API feature sets we wish to have available. This patch does that, and also includes a double-check to ensure that the system we are compiling on can actually provide the requested feature sets. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-16Add two more API calls for getting the current glue and channel in bridging ↵Joshua Colp
code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-15Add an empty line after each option when printing theMichiel van Baak
documentation of a function/application. This will make reading the docs on the CLI way more easy. (closes issue #15694) Reported by: mvanbaak Patches: 2009081100-extralinesoptionlist.diff.txt uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-13Add an API call for retrieving the engine in use by an RTP instance.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10AST-2009-005Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10Fix up some issues with getting a channel by "name".Russell Bryant
Even though the get_channel_by_name() API advertised that you could search by name or uniqueid (just as the old API did), searching by uniqueid was not actually implemented. This patch fixes that problem. The ast_channel_get_full() function now makes a second search attempt by uniqueid if the parameter was a name. The channel comparison function also now knows how to compare by unqieueid. Finally, a bug was fixed in passing where OBJ_POINTER was being passed in some scenarios where it should not have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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