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2007-06-05When shutting down "gracefully", go through and run the unload() callbacks forrussell1-0/+7
all of the modules. "stop now" is considered a non-graceful shutdown and will not go through this process. (issue #9804, reported by chrisost, patch by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@67308 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-04Fix some compiler warnings in C++ modules.russell1-2/+2
(issue #9866, reported by osk, patch by Corydon76) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@67018 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-31Change a couple of header files to not use "new", which is a reserved keywordrussell2-2/+2
in C++. (issue #9830, reported by osk) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@66775 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Checking for the strip application needs to be done with AC_PATH_TOOLrussell1-0/+3
instead of AC_PATH_PROG to properly handle cross compilation environments. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@66026 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Fix handling of zero-length frames when a codec is capable of native PLC.qwell1-0/+1
Issue 9183, patch by Mihai. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65877 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18Merged revisions 65172 via svnmerge from murf1-2/+10
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-17Merged revisions 64819 via svnmerge from tilghman1-6/+6
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) | 2 lines How is it that we never caught that this is returning the opposite of our documentation, until now? ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@64820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Merged revisions 63285 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@63286 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30When serving dynamic content, include a Cache-Control header to instruct therussell1-1/+3
browsers to not store the resulting content. (issue #9621, reported by Pari, patch by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@62414 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Merged revisions 61804 via svnmerge from file1-1/+18
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61805 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24Improve DTMF handling in ast_read() even more in response to a discussion onrussell1-1/+1
the asterisk-dev mailing list. I changed the enforced minimum length of a digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in between digits. These values are not configurable in a configuration file right now, but they can be easily changed near the top of main/channel.c. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61781 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Fix the UpdateConfig manager action to properly treat "variables" and "objects"russell1-1/+2
differently (a=b versus a=>b). (issue #9568, reported by pari, patch by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61690 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09This is a big improvement over the current CDR fixes. It may still need ↵murf1-1/+1
refinement, but this won't have as many folks bothered. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60989 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09Merged revisions 60849 via svnmerge from tilghman1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60850 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06To be able to achieve the things that we would like to achieve with therussell3-0/+30
Asterisk GUI project, we need a fully functional HTTP interface with access to the Asterisk manager interface. One of the things that was intended to be a part of this system, but was never actually implemented, was the ability for the GUI to be able to upload files to Asterisk. So, this commit adds this in the most minimally invasive way that we could come up with. A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in the parser, and updated it to be thread-safe. The ability to check permissions of active manager sessions was added by Dwayne Hubbard. Then, hacking this all together and do doing the modifications necessary to the HTTP interface was done by me. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60603 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06Add support for returning different types of results (ie: NBest).file1-0/+13
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60361 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-30several changes via kpflemings reviewmurf1-7/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59522 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-30These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. ↵murf1-0/+12
Mainly with CDRs generated from transfer situations. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59486 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in somerussell1-1/+13
because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, ↵nadi1-0/+3
make chan_misdn use it. * add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in' (the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59202 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-20The fix for the AEL <<security hole>> (bug 9316) is here...murf1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59069 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15Add configure script checking for GTK2 and some additional Makefile targetsrussell1-0/+3
to support gmenuselect git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58947 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-01Merge changes from svn/asterisk/team/russell/sla_updatesrussell1-9/+10
* 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
2007-02-20Increase the maximum number of manager headers to 128, at the request of Pari.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55590 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-17If the pg_config application is found, but there is probably executing it,russell1-3/+0
then consider postgres unavailable. (issue #8637) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55052 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-13Fix the documentation on the return values from device state providerrussell1-2/+3
registration and deletion. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54218 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-12Change ast_set_state_callback() to ast_dial_set_state_callback()russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54103 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-12- Add the ability to register a callback to monitor state changes in anrussell1-3/+12
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
2007-02-10Merge team/russell/sla_rewriterussell4-1/+211
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
2007-01-30When we are checking for a system installed version of libgsm, we need to checkrussell1-0/+6
for gsm.h as well. Furthermore, when checking for this header, it may be located in a gsm/ sub directory, so check for that, as well. (issue #8773) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52997 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29Clean up a few things in the last commit to the adaptive jitterbuffer code.russell1-2/+1
- Specifically indicate to the compiler that the "dropem" variable only needs one but. - Change formatting to conform to coding guidelines. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52506 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29Fixed problem with jitterbuf, whereas it would not complain about, andjdixon1-0/+1
would allow itself to be overfilled (per the max_jitterbuf parameter). Now it rejects any data over and above that size, and complains about it. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52494 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-24Fix the formatting of doxygen comments to properly indicate that the commentrussell1-13/+13
documents the previous entity, as opposed to the next one. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52107 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-24Merge in dialing API and the app_page that uses it. (issue #BE-118)file1-0/+142
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52049 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Merge the changes from the /team/group/vldtmf_fixup branch.russell1-16/+20
The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51311 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-15use the ACX_PTHREAD macro from the Autoconf macro archive for setting up ↵kpfleming1-3/+7
compiler pthreads support... should improve portability to platforms with unusual pthreads requirements git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@50867 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-11Add support to see whether NAT was detected (yay symmetric RTP) and also add ↵file1-0/+2
a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@50466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-05reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most ↵kpfleming2-18/+13
cases git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49676 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-04add support for tracking thread-local-storage objects that exist via ↵kpfleming2-0/+67
'threadstorage' CLI commands git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49553 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-28Backport support for read/write locks.file2-0/+72
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49022 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-28removed <err.h> as in trunk from the ael stuff. Also, threw in a minor fix ↵murf1-1/+1
to frame.c to avoid build-killing compiler warnings. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49020 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27move extern declaration for this option to a header file where it belongskpfleming1-0/+2
provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48998 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27allow 'show memory' and 'show memory summary' to distinguish memory ↵kpfleming2-0/+17
allocations that were done for caching purposes, so they don't look like memory leaks git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48987 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-26Add an API call that initializes an RTP structure. We need this because ↵file1-0/+2
chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48964 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-18unbreak the macro used for incrementing the frame counters.rizzo1-1/+1
I don't know when the bug was introduced, but with the typical usage c->fin = FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects trunk as well (fix coming). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48566 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-16use m4 quoting for AC_MSG_NOTICE calls, to keep these calls from thinking ↵kpfleming1-0/+28
they have multiple arguments git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48528 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-16since we really, really have to have autoconfig.h included before all other ↵kpfleming1-0/+7
headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48521 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-14Payload values on the RTP structure can change AFTER a bridge has started. ↵file1-1/+1
This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48472 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-02- Disable RTP hold timers while T.38 fax transmission happensoej1-0/+15
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send something that video phones support in the RTP stream. I now this is a big architectual change at this stage for 1.4, but decided it was needed to avoid future bug reports. - Document the RTP NAT keepalive option in sip.conf.sample Issue 7679 in the bug tracker. Please test. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48199 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-02Backport the comment containing the warning regarding the limitations on therussell1-0/+10
usage of this function. It is thread safe, but not technically reentrant. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48195 f38db490-d61c-443f-a65b-d21fe96a405b
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