Russell Bryant [Tue, 15 May 2007 23:05:20 +0000 (23:05 +0000)]
Add two new dialplan functions: ENUMQUERY and ENUMRESULT. These functions
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)
Jason Parker [Mon, 14 May 2007 21:51:03 +0000 (21:51 +0000)]
With libmmime.a as a .PHONY target, asterisk gets rebuilt every time, but without proper ASTCFLAGS.
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.
Russell Bryant [Mon, 14 May 2007 21:17:52 +0000 (21:17 +0000)]
Merged revisions 64353 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64353 | russell | 2007-05-14 16:16:39 -0500 (Mon, 14 May 2007) | 4 lines
When someone requests a specific parking space using the PARKINGEXTEN variable,
ensure that no other caller is already there.
(issue #9723, reported by mdu113, patch by me)
Russell Bryant [Mon, 14 May 2007 19:21:31 +0000 (19:21 +0000)]
Merged revisions 64306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
will trigger an error and cause sounds to stop, which in this case, is ringing.
Only perform stripping of - strings from the channel name for Zap channels. Anywhere else we might remove a legitimate part of a device name. (issue #9668 reported by stevedavies)
Olle Johansson [Sun, 13 May 2007 19:20:36 +0000 (19:20 +0000)]
Improve handling network errors on transmission to hosts that don't reply or are unreachable
With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.
Russell Bryant [Fri, 11 May 2007 16:21:45 +0000 (16:21 +0000)]
Merged revisions 63886 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
Russell Bryant [Thu, 10 May 2007 22:25:54 +0000 (22:25 +0000)]
Merged revisions 63804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r63804 | russell | 2007-05-10 17:23:42 -0500 (Thu, 10 May 2007) | 4 lines
Strip terminal escape sequences from CLI command output that is going to be
sent out over the manager interface.
(issue #9659, reported by pari, fixed by me)
Russell Bryant [Wed, 9 May 2007 19:21:35 +0000 (19:21 +0000)]
Merged revisions 63612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
Russell Bryant [Wed, 9 May 2007 16:44:33 +0000 (16:44 +0000)]
Merged revisions 63608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
Russell Bryant [Wed, 9 May 2007 13:24:38 +0000 (13:24 +0000)]
Merged revisions 63535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r63535 | russell | 2007-05-09 08:24:03 -0500 (Wed, 09 May 2007) | 6 lines
I have seen multiple people post questions trying to figure out what the
message "The configure script must be executed before running 'make'" means.
So, add another like that says to specifically run ./configure. If this isn't
obvious enough, then they should be using something like AsteriskNOW and not
installing from source.
Russell Bryant [Mon, 7 May 2007 22:14:09 +0000 (22:14 +0000)]
Add a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
except it lets you operate on a channel by name instead of conference member
number. It is very useful in combination with the 'X' option to ChanSpy.
(issue #9671, patch by mnicholson, with some small modifications by me)
Olle Johansson [Sat, 5 May 2007 08:05:38 +0000 (08:05 +0000)]
- Adding some missing spaces
- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily
Steve Murphy [Fri, 4 May 2007 16:37:23 +0000 (16:37 +0000)]
Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
Olle Johansson [Fri, 4 May 2007 13:56:25 +0000 (13:56 +0000)]
Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
is needed to follow the call through the PBX.
Olle Johansson [Fri, 4 May 2007 13:44:50 +0000 (13:44 +0000)]
- Add manager command CoreSettings
- Add missing option to options.h
- Add missing variables to asterisk.h
- Move manager version to manager.h include file
When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)
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)
Russell Bryant [Thu, 3 May 2007 15:23:44 +0000 (15:23 +0000)]
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
improve static Realtime config loading from PostgreSQL:
don't request sorting on fields that are pointless to sort on
use ast_build_string() instead of snprintf()
don't request the list of fieldnames that resulted from the query when we both knew what they were before we ran the query _AND_ we aren't going to do anything with them anyway
(patch by me, inspired by blitzrage's bug report about res_config_odbc)
increase reliability and efficiency of static Realtime config loading via ODBC:
don't request fields we aren't going to use
don't request sorting on fields that are pointless to sort on
explicitly request the fields we want, because we can't expect the database to always return them in the order they were created
(reported by blitzrage in person (!), patch by me)
Russell Bryant [Wed, 2 May 2007 23:50:07 +0000 (23:50 +0000)]
When a conference is created, the UNIQUEID of the channel that caused it to be
created will now be stored. Then, every channel that joins the conference will
have the MEETMEUNIQUEID channel variable set with this ID. This can be used to
relate callers that come and go from long standing conferences.
(issue #7295, patch by softins)
Russell Bryant [Wed, 2 May 2007 23:00:07 +0000 (23:00 +0000)]
Merged revisions 62789 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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!)
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.
Steve Murphy [Wed, 2 May 2007 17:24:03 +0000 (17:24 +0000)]
Merged revisions 62689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
........
Russell Bryant [Wed, 2 May 2007 15:46:49 +0000 (15:46 +0000)]
Update the device state functionality of chan_local such that it will return
NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048, patch from tim_ringenbach)
projects / thirdparty / asterisk.git / log
