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2007-03-18 Don't return a non-zero return code if the profile doesn't exist, to match ↵bweschke1-1/+1
what the documentation says it already does. (#9307 Reported by kkiely) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59035 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-16Wait for the async thread to exit when hanging up all of the paged phones ↵file1-3/+2
under all circumstances. (issue #9181 reported by PhilSmith) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58992 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-16Making these documentation changes in the 1.4 branch upset various people, sorussell1-7/+7
these chanes will only be done in the trunk. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58955 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15Expand deprecation warnings from simply warning on use to the builtin ↵tilghman3-4/+9
documentation. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58939 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15Merge changes from svn/asterisk/team/russell/LaTeX_docs.russell1-7/+7
* Convert most of the doc directory into a single LaTeX formatted document so that we can generate a PDF, HTML, or other formats from this information. * Add a CLI command to dump the application documentation into LaTeX format which will only be include if the configure script is run with --enable-dev-mode. * The PDF turned out to be close to 1 MB, so it is not included. However, you can simply run "make asterisk.pdf" to generate it yourself. We may include it in release tarballs or have automatically generated ones on the web site, but that has yet to be decided. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58931 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-14By default, don't attempt to do any CallerID handling at all with SLA becauserussell1-4/+11
it is known to not work properly in some situations. However, add an option to enable it for those that would like to use it anyway. The short story behind this is that to properly handle CallerID with SLA, we need the ability to change the CallerID on an existing call, and we are not ready to handle that. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58894 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-13Ensure that the blinky lights show that the trunk stopped ringing when therussell1-0/+3
trunk hangs up before a station has answered it. (issue #9234, reported by francesco_r) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58872 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-10Make the compiler happy and initialize a variable.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58669 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-09Fix spelling of unavailable in voicemail documentation. (issue #9248 ↵file1-1/+1
reported by tensai) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58604 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08Hang up the channel that put the call on hold in the event processing thread torussell1-6/+5
avoid a race condition. Also, if the station originated the call that it is putting on hold, don't hang up the trunk if it was the only station on the call and it is hanging up due to hold and not a normal hangup. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08Refactor hold handling a bit so that it does not require keeping the call uprussell1-38/+41
when a call is put on hold. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58474 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08fix a compiler warning, and overwriting 'res' valuekpfleming1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58352 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Don't create a listen channel and record the conference unless the option is ↵file1-17/+16
turned on. (issue #9204 reported by francesco_r) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57872 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Merged revisions 57869 via svnmerge from file1-8/+8
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57870 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-03Merged revisions 57648 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines Memory leak of a list, if call recording was abandoned ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57649 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-01Merge changes from svn/asterisk/team/russell/sla_updatesrussell1-25/+16
* Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57364 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge more changes from svn/asterisk/team/russell/sla_updatesrussell1-13/+99
* Add support for private hold. By setting "hold=private" for a trunk, only the station that put the call on hold will be able to retrieve it from hold. Also, by setting "hold=private" for a station, any call that station puts on hold can only be retrieved by that station. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57203 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Minor formatting changerussell1-7/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57146 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge changes from svn/asterisk/team/russell/sla_updatesrussell1-9/+26
* Add support for the "barge=no" option for trunks. If this option is set, then stations will not be able to join in on a call that is on progress on this trunk. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57144 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge current set of changes from svn/asterisk/team/russell/sla_updatesrussell1-149/+457
* Add support for station ring delays. Ring delays can be set globally for a station or for specific trunks on the station. * Fix a few bugs in existing code. * Restructure and Reorganize code to improve readability and maintainability. * Improve formatting of the "sla show (trunks|stations)" CLI commands. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57089 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Picky compiler...file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57055 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Better handle timeouts when the individual speaks after everything has been ↵file1-3/+4
played but before the timeout ends. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57053 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-27Fix voicemail email attachments. I missed the conversion of one of the linerussell1-2/+2
endings and there was an extra one where it should not have been. (issue #9128) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56975 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Picky, picky... show deprecation warning in application help, too (reported ↵tilghman2-2/+6
via list) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56922 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Update app_record documentation to use new CLI command, core show file ↵file1-1/+1
formats. (issue #9151 reported by junky) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56839 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Move a comment to be in the correct struct.russell1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56740 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-23The IMAP storage code uses the same code to build the email that is used whenrussell1-41/+47
voicemail is sent via email using something like sendmail. In the patch from bug 8033 to fix various IMAP storage problems, the line endings in the email file were changed in the code from "\n" to "\r\n". However, this breaks sending regular voicemail to email. So, this change conditionally sets line endings to "\r\n" only if IMAP_STORAGE is enabled. (issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56341 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22Merge changes from team/russell/sla_updates.russell1-285/+664
This batch of changes to the SLA code does a few different things. * I made the SLA code event driven instead of having to act in a lot of busy loops while dialing things to wait for state changes. This makes the code more efficient and readable at the same time. * I have implemented a couple of new features. The first is inbound trunk ringing timeouts. This is an option that defines how long to let an incoming call on a trunk to ring. * I have also implemented ring timeouts for stations. They may be specified for the entire station, meaning it is how long to let the station ring before giving up. You can also specify a ring timeout for a specific trunk on a station. So, you can say that you only want a specific station to ring 5 seconds if it is line1 ringing, but otherwise, there is no timeout. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56277 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Merged revisions 55956 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 lines Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55957 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Simplify the last change to app_meetme, and move the call to dispose_conf()russell1-2/+0
up into the block where we know a conf exists. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55951 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Only dispose of the conference if one was created.file1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55949 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Only start playing the next file if we have not been quieted.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55947 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Improve the reference counting to fix bugs where people report seeingrussell1-43/+53
conferences listed that have no members. (issue #9073) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55758 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Better handle dropped IMAP connections. (issue #9054 reported by bsmithurst)file1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55741 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-19Merged revisions 55434 via svnmerge from tilghman1-9/+16
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) | 2 lines forcename and forcegreetings options should check to see if the recording already exists ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55435 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-18Merged revisions 55277 via svnmerge from oej1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 lines Documentation update (#9053, jsmith) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55278 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-17Add missing membername option to AddQueueMember documentation. (issue #9088 ↵file1-1/+1
reported by seanbright) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55219 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-17Merged revisions 55153 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines Answer the channel before recording privacy information. (issue #8926 reported by lmamane) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55154 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-17Make the 'i' option of Queue actually work. (issue #8986 reported by utis)file1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55129 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Merged revisions 55005 via svnmerge from russell1-56/+37
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a conference is created from meetme.conf, it is acceptable behavior that the pin can not be changed until the conference goes away. I also added a note in meetme.conf to describe this behavior. We still have another issue in 1.4 and trunk where some conferences with no users don't go away. That is the real bug that needs to be addressed here. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55006 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Merged revisions 54955 via svnmerge from russell1-37/+56
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines For conferences that are configured in meetme.conf, check the configuration file every time someone joins the conference instead of only when the conference is first created. This is to ensure that changes to the pin numbers in the config file are always honored. (issue #9073) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54969 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Need to check macro extension as well as macro context for directed pickup.file1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54924 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Allow directed pickup to pick up the real context instead of the macro ↵file1-1/+4
context if a Macro is used. (issue #8984 reported by jamesb63) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54884 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Don't let dtmf leak over into the engine and let it skew the results... also ↵file1-26/+30
give DTMF results priority. (issue #9014 reported by surftek) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54714 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-15Merged revisions 54622 via svnmerge from file1-9/+13
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 lines Use a separate variable to indicate execution should continue instead of the return value. (issue #8842 reported by pluto70) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54623 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman)file1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54481 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-12- Add the ability to register a callback to monitor state changes in anrussell2-3/+3
asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54066 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-10Much simpler than previous one ;-)pcadach1-15/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53880 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-10Add some output for "show application SLAStation/SLATrunk"russell1-2/+13
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53821 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-10Merge team/russell/sla_rewriterussell1-514/+1329
This is a completely new implementation of the SLA functionality introduced in Asterisk 1.4. It is now functional and ready for testing. However, I will be adding some additional features over the next week, as well. For information on how to set this up, see configs/sla.conf.sample and doc/sla.txt. In addition to the changes in app_meetme.c for the SLA implementation itself, this merge brings in various other changes: chan_sip: - Add the ability to indicate HOLD state in NOTIFY messages. - Queue HOLD and UNHOLD control frames even if the channel is not bridged to another channel. linkedlists.h: - Add support for rwlock based linked lists. dial.c: - Add the ability to run ast_dial_start() without a reference channel to inherit information from. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53810 f38db490-d61c-443f-a65b-d21fe96a405b
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