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2009-08-18git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@212958 ↵v1.4.15kpfleming8-20/+19
f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29use autotagged externalsrussell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@90174 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29importing files for 1.4.15 releaserussell3-0/+13750
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@90173 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Creating tag for the release of asterisk-1.4.15russell3-13749/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@90172 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29use autotagged externalsrussell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@90169 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29importing files for 1.4.15 releaserussell3-0/+13749
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@90168 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Creating tag for the release of asterisk-1.4.15russell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@90167 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Properly escape cdr->src and cdr->dst and ensure we use thread-safe escapingtilghman1-11/+16
(Fixes AST-2007-026) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29This patch handles the case where a queue member with a negative penalty is ↵mmichelson1-1/+1
added via the manager. If a negative value is submitted for a member penalty, we set it to 0. (closes issue #11411, reported and patched by Laureano) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Properly escape input buffers (Fixes AST-2007-025)tilghman1-13/+66
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90160 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Use of "private" as a field name in a header file messes with C++ projectstilghman9-22/+22
Reported by: chewbacca Patch by: casper (Closes issue #11401) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90155 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Upgrade the core sounds release versiontilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90154 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29fix some formatting i accidentally changedrussell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90147 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29This set of changes is to make some callerID handling thread-safe.russell3-4/+17
The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90145 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Merge a change from team/russell/chan_refcount ...russell2-3/+13
This makes ast_stopstream() thread-safe. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90142 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28Fix a few memory leaks.file1-0/+1
(closes issue #11405) Reported by: eliel Patches: load_realtime.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90101 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28it is impossible to set permissions for manager accounts created by ↵kpfleming2-2/+14
users.conf (reported internally, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90098 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28Removing some seemingly pointless code. This sets a channel variable for ↵mmichelson1-6/+0
every priority executed in the dialplan if you have debug set to anything non-zero. This seems pointless due to the fact that these channel variables are not referenced anywhere else in the code and their names are esoteric enough that they would not be practical to reference in the dialplan. Plus the fact that this behavior isn't documented anywhere means that the change is not likely to cause any disruption. If anything, this may actually cause a slight performance increase if running with debug on. The motivating influence for this code change is the eventwhencalled option for queues. If set to vars, all channel variables will be output to the manager. These unnecessary channel variables make the output a lot more difficult to deal with. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90059 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28Recording greetings when using IMAP storage was causing zero-length files to ↵mmichelson1-0/+4
be stored. Since greetings are not retrieved from IMAP anyway, it is pointless to attempt storing them there. (closes issue #11359, reported by spditner, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89999 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28 - update documentation for some of the goto functions to note that theyrussell2-1/+24
handle locking the channel as needed - update ast_explicit_goto() to lock the channel as needed git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89893 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Don't do frame processing if ast_read() returned NULL.russell1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89886 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Instead of depending on the return value of ast_true(), explicitly set therussell1-1/+1
eventwhencalled variable to 1. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89844 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Don't start/stop autoservice in pbx_extension_helper() unless a channel existsrussell1-8/+16
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89839 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Two changes with regards to the 'eventwhencalled' option of queues.confmmichelson1-3/+3
1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' did exactly the same thing. Thus the sign change of the ast_true call. 2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting in bizarre output for the channel variables. This patch remedies this. (related to issue #11385, however I'm not sure if this will actually be enough to close it) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89837 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merge changes from team/russell/autoservice_1.4russell2-7/+90
This set of changes fixes an issue that was reported to me on IRC yesterday. The user, d1mas, was using chan_zap for incoming calls and was having DTMF recognition issues in some situations. Specifically, he noticed that the problem occurred when using DISA or WaitExten. He also noticed that when using Read, the problem did not occur. His system also used DUNDi for dialplan lookups. So, he theorized that if the DUNDi lookups blocked for some period of time, that audio from the zap channel could get lost. If the audio got lost, then it wouldn't be run through the DTMF detector, and digits could get lost. He was correct, and the following set of changes fixes the problem. However, the changes go a little bit further than what was necessary to fix this exact problem. 1) I updated pbx_extension_helper() to autoservice the associated channel to handle cases where extension lookups may take a long time. This would normally be a dialplan switch that does some lookup over the network, such as the DUNDi or IAX2 switches. This ensures that even while a DUNDi lookup is blocking, the channel will be continuously serviced. 2) I made a change to the autoservice code. This is actually something that has bothered me for a long time. When a channel is in autoservice, _all_ frames get thrown away. However, some frames really shouldn't be thrown away. The most notable examples are signalling (CONTROL) frames, and DTMF. So, this patch queues up important frames while a channel is in autoservice. When autoservice is stopped on the channel, the queued up frames get stuck back on the channel so that they can get processed instead of thrown away. 3) I made another change to the autoservice code to handle the case where autoservice is started on channels recursively. Previously, you could call ast_autoservice_start() multiple times on a channel, and it would stop the first time ast_autoservice_stop() gets called. Now, it will ensure that autoservice doesn't actually stop until the final call to ast_autoservice_stop(). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89790 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Changing some calls from free() to ast_free() since they were allocated withmmichelson1-3/+3
ast_calloc(). (closes issue #11390, reported and patched by Laureano) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89727 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27on second thought... revert all the other changes i've made in app options ↵kpfleming1-5/+1
parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89709 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27generate a warning when an application option that requires an argument is ↵kpfleming1-2/+5
ignored due to lack of an argument git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89701 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Add a note to the sample voicemail config noting that when using IMAP storage,russell1-1/+2
only the first format specified will be attached to the message. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89634 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Default result of STAT should be "0" not "".tilghman1-1/+1
Reported via the -users mailing list, fixed by me. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89631 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27If we get a codec offer using a well-known payload type, but using it for ↵oej3-12/+52
another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89630 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Clarify limitonpeers=yesoej1-0/+3
(closes issue #11304) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89624 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27closes issue #11379; OK, this is an attempt to make both sides happy. To the ↵murf4-3/+43
cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89622 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26After issuing a "say load new", if a caller hangs up during the middle of ↵mmichelson1-1/+5
playback of a number, app_playback will continue to try to play the remaining files. With this change, no more files will be played back upon hangup. (closes issue #11345, reported and patched by IgorG) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89618 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26After issuing a "say load new" tons of warning messages are printedmmichelson1-4/+0
out to the CLI every time do_say in app_playback is called. Removing these warnings git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89616 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Fix issues with async dialing with an application executing. The application ↵file1-5/+24
has to be terminated and control returned to the thread before hanging things up. (issue #BE-252) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89610 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Add module counting removal for error conditions.file1-0/+3
(closes issue #11333) Reported by: Laureano Patches: res_features_v2.c.patch uploaded by Laureano (license 265) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89599 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Add channel locking to a function that needed to be doing it. This is just arussell1-0/+2
little something I noticed while working on a completely unrelated issue. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89594 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Use ast_free to free memory, or else we shall implode if MALLOC_DEBUG is ↵file1-2/+2
enabled. (closes issue #11347) Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys (license 281) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89592 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Close the audio file before sending it to the post processing application.file1-3/+3
(closes issue #11357) Reported by: reformed Patches: mixmonitor.patch uploaded by reformed (license 330) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89587 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26when parsing application options that take arguments, don't indicate that ↵kpfleming1-1/+2
the option was supplied unless a non-zero-length argument was found for it git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89586 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Revert vmu->email back to an empty string if it was empty when imap_store_filemmichelson1-1/+12
was called. This prevents sending a duplicate e-mail. (closes issue #11204, reported by spditner, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89580 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26If channel allocation fails because the alert pipe could not be created also ↵file1-0/+1
free the scheduler context. (closes issue #11355) Reported by: eliel Patches: main.channel.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89577 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26When unloading app_meetme destroy any auto created contexts created by SLA.file1-0/+3
(closes issue #11367) Reported by: eliel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89571 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25We previously attempted to use the ESCAPE clause to set the escape delimiter totilghman4-6/+32
a backslash. Unfortunately, this does not universally work on all databases, since on databases which natively use the backslash as a delimiter, the backslash itself needs to be delimited, but on other databases that have no delimiter, backslashing the backslash causes an error. So the only solution that I can come up with is to create an option in res_odbc that explicitly specifies whether or not backslash is a native delimiter. If it is, we use it natively; if not, we use the ESCAPE clause to make it one. Reported by: elguero Patch by: tilghman (Closes issue #11364) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89559 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24Free some frames that would otherwise leak on error.tilghman1-0/+3
Reported by: Laureano Patch by: Laureano,tilghman (Closes issue #11351) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89545 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24Currently, zero-length voicemail messages cause a hangup in VoicemailMain.tilghman2-4/+15
This change fixes the problem, with a multi-faceted approach. First, we do our best to avoid these messages from being created in the first place, and second, if that fails, we detect when the voicemail message is zero-length and avoid exiting at that point. Reported by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89540 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-23Up until this point, the XML output of the manager has been technicallytilghman1-3/+71
invalid, due to the repetition of certain parameters in a single event. This caused various issues for XML parsers, some of which refused to parse at all, given the invalidity of the rendered XML. So this commit fixes the XML output, ensuring that each entity parameter has a unique name, thus ensuring valid XML. Reported by: msetim Patch by: tilghman (Closes issue #10220) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-23Use ESCAPE clause for the first parameter, not just 2nd-Nth parameters.tilghman1-1/+2
Reported by: apsaras Patch by: tilghman (Closes issue #11353) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89534 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22mvanbaak pointed out a spelling error in this sample configuration file. Whilerussell1-2/+2
I was at it, I went ahead and tweaked it a little bit more. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89527 f38db490-d61c-443f-a65b-d21fe96a405b
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