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${ noResults }
1161 Commits (65cf5ba448c1d225b51e16702f09b1f5985e3348)
Author | SHA1 | Message | Date |
---|---|---|---|
|
b419fc1134 |
Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
3181a91148 |
Merged revisions 62414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) | 4 lines When serving dynamic content, include a Cache-Control header to instruct the browsers to not store the resulting content. (issue #9621, reported by Pari, patch by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
b6b1bf3213 |
Merge changes from team/russell/events
This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
3f414d2aea |
Remove a message that goes to LOG_ERROR that's not really an error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
8b2b3e172b |
Merged revisions 61805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61804 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@61806 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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94459660a3 |
Merged revisions 61781 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines Improve DTMF handling in ast_read() even more in response to a discussion on 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: https://origsvn.digium.com/svn/asterisk/trunk@61782 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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97d0661327 |
Merged revisions 61690 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61690 | russell | 2007-04-20 13:19:18 -0500 (Fri, 20 Apr 2007) | 4 lines Fix the UpdateConfig manager action to properly treat "variables" and "objects" differently (a=b versus a=>b). (issue #9568, reported by pari, patch by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e1d30288c0 |
Doxygen changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61667 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2151e532fe |
changed #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6a5f3599bb |
added HAVE_SYSINFO preprocessor directives for portability and general happiness
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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97d93d091b |
Add a configure script check for sysinfo support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61557 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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62256ee410 |
added option_minmemfree for use in asterisk.conf to specify the amount of minimum free memory prior to accepting calls. added CLI 'core show sysinfo' to display system information
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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69bb679e14 |
via 8118, a RealTime upgrade to make RT a complete storage abstraction. The store/destroy mechanisms needed these missing peices.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61374 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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47dd5a15af |
Issue 6082 - New DTMF event for manager
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61324 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ab31bec392 |
Add an option to the dial API for playing music instead of ringing to the caller.
I started this for use with SLA but ended up deciding not to use it. However, there is no reason not to put this part in, anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ecaf781933 |
Merged revisions 60989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5ecaea3d0a |
Merged revisions 60850 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60850 | tilghman | 2007-04-08 22:01:12 -0500 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60849 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@60851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0a9750ef9f |
Merged revisions 60603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines To be able to achieve the things that we would like to achieve with the 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: https://origsvn.digium.com/svn/asterisk/trunk@60604 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1c2e3e1ffc |
Major res_speech cleanup. It looks much better now!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4b618442a0 |
Merged revisions 60361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2 lines Add support for returning different types of results (ie: NBest). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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09c0d56c5c |
Merged revisions 59522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1 line several changes via kpflemings review ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0f11d3c8c3 |
Merged revisions 59486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1 line These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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be6013b206 |
Enhancement via 8118: Realtime API extension: add methods store_func and destroy_func, to make Realtime a complete database abstraction
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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08e3a9bdc8 |
Merged revisions 59207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some 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: https://origsvn.digium.com/svn/asterisk/trunk@59208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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24d8595d00 |
Merged revisions 59202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines * mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, 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: https://origsvn.digium.com/svn/asterisk/trunk@59203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e6d32d3132 |
The fix for the AEL <<security hole>> (bug 9316) is here...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59073 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7f31b58b21 |
Merged revisions 58947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58947 | russell | 2007-03-15 18:53:26 -0500 (Thu, 15 Mar 2007) | 3 lines Add configure script checking for GTK2 and some additional Makefile targets to support gmenuselect ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58948 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5bea998a55 |
Merge changes from team/russell/sqlite:
* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a SQLite3 database. (issue #7149, alerios) * Add new module, res_config_sqlite, which adds realtime database configuration support for SQLite version 2. I decided that this was ok since we didn't have any realtime support for version 3. If someone ports this to version 3, then version 2 support can be removed or marked deprecated. (issue #7790, rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom. Also, note that there were other modules on the bug tracker that did not make the cut because they provided some duplicated functionality. Those are: * cdr_sqlite3 (issue #6754, moy) * cdr_sqlite3 (issue #8694, bsd) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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39c92fe449 |
Add some documentation on the arguments to the base64 encode/decode functions.
(inspired by issue #9215) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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590cb3a6fa |
Expand datastores to add the notion of inheritance. This will be needed for
the conversion of IAX2 variables from the current custom method to ast_storage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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3d6e6e07ef |
Merged revisions 57364 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines Merge changes from svn/asterisk/team/russell/sla_updates * 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: https://origsvn.digium.com/svn/asterisk/trunk@57365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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e7b03a991e |
Convert the PBX core to use read/write locks. This yields a nifty performance improvement when it comes to simultaneous calls going through the dialplan. Using murf's test the old mutex based core took an average of 57.3 seconds while the rwlock based core took 31.1 seconds. That's a nifty 26.2 seconds performance improvement. The other good part is that if we do need to switch back then we just have to change the lock/unlock API calls. I converted everywhere that used to touch the mutex locks directly to use them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57184 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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75d387acbc |
Doxygen additions, corrections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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e916cf45da |
Doxygen updates and corrections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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bc01e39174 |
Creating new doxygen macro "\extref" to create page that lists
external libraries and URLs to these. Please help me add these references. We might want to create a similar macro "\linuxpackage" to list the needed Linux packages in popular distributions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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c683aacfc6 |
Add some external references
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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e930d1e88e |
Doxygen updates for AJI - The Asterisk Jabber API
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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ad7d044c75 |
move the ast_module_info structure into the special section as well, otherwise when restore_globals() is called it will lose its pointer to the ast_module structure that the loader put there
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56209 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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1bec2f5bfa |
give embedded modules a helping hand by backing up and restoring their global variables when they are loaded and unloaded
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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8088fcd312 |
Merged revisions 55590 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55590 | russell | 2007-02-20 13:57:07 -0600 (Tue, 20 Feb 2007) | 2 lines Increase the maximum number of manager headers to 128, at the request of Pari. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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a347807103 |
Merged revisions 55052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55052 | russell | 2007-02-16 18:40:34 -0600 (Fri, 16 Feb 2007) | 3 lines If the pg_config application is found, but there is probably executing it, then consider postgres unavailable. (issue #8637) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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ba32ee49d0 |
Adding Realtime Text support (T.140) to Asterisk
T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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8ac0fb2bc3 |
New CLI command "Core show settings" to list some core settings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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f60efe347a |
This introduces a new dialplan function, DEVSTATE, which allows you to do some
pretty cool things. First, you can get the device state of anything in the dialplan: NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)}) NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)}) Most importantly, this allows you to create custom device states so you can control phone lamps directly from the dialplan. Set(DEVSTATE(Custom:mycustomlamp)=BUSY) ... exten => mycustomlamp,hint,Custom:mycustomlamp git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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3b7cfbb80d |
Merged revisions 54218 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54218 | russell | 2007-02-13 14:56:50 -0600 (Tue, 13 Feb 2007) | 3 lines Fix the documentation on the return values from device state provider registration and deletion. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54219 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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4300a7a6cd |
- Constify the format string passed to ast_cli()
- Simplify printing out the warranty and license git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54147 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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8f8df3e5a9 |
Merged revisions 54103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) | 2 lines Change ast_set_state_callback() to ast_dial_set_state_callback() ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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2a5477b35e |
Merged revisions 54066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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5715b49c30 |
Merged revisions 53810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines Merge team/russell/sla_rewrite 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: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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f722121ca0 |
Merged revisions 52997 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52997 | russell | 2007-01-30 17:23:24 -0600 (Tue, 30 Jan 2007) | 5 lines When we are checking for a system installed version of libgsm, we need to check 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: https://origsvn.digium.com/svn/asterisk/trunk@52998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |