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.
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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)
Russell Bryant [Tue, 1 May 2007 22:24:51 +0000 (22:24 +0000)]
In addition to making it so attended transfers don't fail unnecessarily,
add some new options to control what happens when you hangup on an attended
transfer before the target extension answers the transferred channel. You
can now have it send the transferee back to the transferer.
(issue #8413, patch from sergee with very minor modifications by me)
Russell Bryant [Tue, 1 May 2007 21:57:58 +0000 (21:57 +0000)]
Merged revisions 62548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62548 | russell | 2007-05-01 16:57:10 -0500 (Tue, 01 May 2007) | 12 lines
Merged revisions 62547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines
Remove an unnecessary check that makes it so if you hang up after doing an
attended transfer before the target extension answers the channel, the transfer
is not successful. (issue #9338, patch by svanlund)
Russell Bryant [Mon, 30 Apr 2007 16:46:49 +0000 (16:46 +0000)]
This patch adds additional information to the EXITWITHKEY and EXITWITHTIMEOUT
entries in the queue log.
(issue #7561, reported and originally patched by fkasumovic, patch slightly
modified and updated to trunk by me)
Russell Bryant [Mon, 30 Apr 2007 16:16:26 +0000 (16:16 +0000)]
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)
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588, reported and patched by softins)
Russell Bryant [Mon, 30 Apr 2007 15:37:23 +0000 (15:37 +0000)]
Don't crash when invalid arguments are provided to the CHANNEL() function
for a SIP channel.
(issue #9619, reported by jtodd, original patch by Corydon76, committed patch
slightly modified by me)
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)
Russell Bryant [Sat, 28 Apr 2007 21:01:44 +0000 (21:01 +0000)]
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.
Russell Bryant [Sat, 28 Apr 2007 19:53:12 +0000 (19:53 +0000)]
Update the DUNDi section of the documentation with example usage of DUNDIQUERY
and DUNDIRESULT. Also, update the automatically generated application docs.
Russell Bryant [Sat, 28 Apr 2007 19:52:37 +0000 (19:52 +0000)]
Merge changes from team/russell/dundi_results
This introduces two new dialplan functions: DUNDIQUERY and DUNDIRESULT.
DUNDIQUERY lets you intitiate a DUNDi query from the dialplan. Then,
DUNDIRESULT will let you find out how many results there are, and access each
one without having to the query again.
Russell Bryant [Fri, 27 Apr 2007 22:08:54 +0000 (22:08 +0000)]
Add a min-announce-frequency option to queues.conf which allows you to control the
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)
Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)
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)
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
projects / thirdparty / asterisk.git / log
