allow you to initiate an ENUM query using ENUMQUERY, and then access the
details of all of the results using ENUMRESULT. Previously, if you wanted
to access multiple results, Asterisk would have to do a new DNS lookup every
time. (patch by bbryant)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This caused a problem with the buildinfo.o file not being able to find asterisk/build.h
This was affecting DESTDIR, but I *think* that if asterisk had never been installed before, it would've failed also.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r63982 | qwell | 2007-05-11 15:16:17 -0500 (Fri, 11 May 2007) | 7 lines
Hide manager password from "manager show user foo".
I realize that there are other ways to get this,
but we really don't need to just show it in plain text so easily.
Issue 9273, patch by junky
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When enabled, it will set the systemname to be the hostname of the system
Issue 9713, patch by Juggie - slightly modified by me, to "failover" to localhost
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r63886 | russell | 2007-05-11 11:05:43 -0500 (Fri, 11 May 2007) | 6 lines
When MD5 authentication is not possible because there is no challenge present,
either because the Challenge action was never issued, or some other reason,
give a proper error message and return an error instead of claiming that the
user wasn't found.
(reported by jsmith on IRC)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) | 5 lines
Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the
code in that if a channel does not have a send_digit_begin() callback, it only
cares about DTMF END events. (pointed out by Michael Neuhauser on the
asterisk-dev list)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) | 5 lines
Only call ast_senddigit_begin() in ast_senddigit() if the channel has a
send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the
wrong thing to do, because that flag indicates that a *bridged* channel
only wants DTMF END events coming from this channel.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Add missing option to options.h
- Add missing variables to asterisk.h
- Move manager version to manager.h include file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62986 | kpfleming | 2007-05-03 11:38:56 -0500 (Thu, 03 May 2007) | 2 lines
improve loader a bit, by avoiding trying to initialize embedded modules twice and avoiding trying to load modules from disk when they have been loaded already during the 'preload' pass (reported by blitzrage on IRC, patch by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | 20 lines
Merge changes from team/russell/inband_dtmf ...
Fix some issues related to generating inband DTMF. There are two changes here:
1) The list of DTMF tones in the senddigit_begin() function explicitly
specified 100ms of the tone followed by 100ms of silence. This really
broke things with the way that Asterisk now wants complete control
over when the digit begins and ends. So, regardless of what Asterisk
really wanted to do, this was going to play out the tone at the length it
wanted to. This caused various problems like DTMF translation to inband to
be extremely unreliable.
The list of tones has been changed so that the correct DTMF tone is played
indefinitely until Asterisk tells it to stop.
2) ast_write() had to be modified to let a DTMF_END frame get processed even
when a generator is present. This is how the tone will finally get stopped.
(issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for
the testing and feedback!)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | 6 lines
If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a
channel. Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) | 5 lines
Add a few more state changes in handle_frame_ownerless() so that the SLA code
will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) | 5 lines
Some dialplan functions, such as CUT(), expect to operate on variables on a
channel. So, this little hack lets them work in places where a channel doesn't
exist, such as within DUNDi configuration.
(issue #9465, reported and patched by Corydon76, testing by blitzrage)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines
Avoid invalid seqno cycling detection.
Per comment from Dave Troy:
This adds back in some simple typecasting I had in an earlier version
which I realize now may be breaking things.
Issue #9554.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61708 65c4cc65-6c06-0410-ace0-fbb531ad65f3