Russell Bryant [Thu, 24 May 2007 15:04:51 +0000 (15:04 +0000)]
Merged revisions 65853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65853 | russell | 2007-05-24 10:04:14 -0500 (Thu, 24 May 2007) | 4 lines
Ensure that frames are fully initialized. This will probably fix getting
weird timestamp log messages in logs when using the Festival app.
(issue #9781, patch by me)
Russell Bryant [Thu, 24 May 2007 14:50:25 +0000 (14:50 +0000)]
Merged revisions 65842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines
Fix the calculation of the RTT for RTCP. The previous code would result in
oscillating and incorrect data. Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)
Merged revisions 65767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line
we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
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Russell Bryant [Thu, 24 May 2007 03:28:39 +0000 (03:28 +0000)]
- Remove debug variable that was only used in one place
- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
Russell Bryant [Wed, 23 May 2007 17:17:45 +0000 (17:17 +0000)]
Don't check for MWI event subscribers before creating the MWI event in voicemail.
MWI events get cached, so go ahead and always generate them so the cache gets
populated.
Russell Bryant [Tue, 22 May 2007 18:52:59 +0000 (18:52 +0000)]
Add a new feature for Music on Hold. If you set the "digit" option for a
class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
Russell Bryant [Tue, 22 May 2007 13:12:15 +0000 (13:12 +0000)]
Blocked revisions 65394 via svnmerge
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r65394 | russell | 2007-05-22 08:09:34 -0500 (Tue, 22 May 2007) | 12 lines
Merged revisions 65389 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | 4 lines
Fix a memory leak that I just noticed in the device state handling in app_queue.
On most device state changes, it would leak roughly 8 to 64 bytes (the length of
the name of the device).
Merged revisions 65328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line
we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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Russell Bryant [Mon, 21 May 2007 06:56:21 +0000 (06:56 +0000)]
I know we have talked about rewriting app_queue for Asterisk 1.6, but once I
saw this, I couldn't help myself from changing it. Previously, for *every*
device state change, app_queue would spawn a thread to handle it. Now, the
device state callback just puts the state change in a queue and it gets
handled by a single state change processing thread.
Joshua Colp [Sun, 20 May 2007 14:48:31 +0000 (14:48 +0000)]
Add the adsistub file to the Asterisk makefile, fix a stub definition, and no longer make the symbols from res_adsi global since they don't need to be.
Steve Murphy [Fri, 18 May 2007 22:35:48 +0000 (22:35 +0000)]
Merged revisions 65201 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65201 | murf | 2007-05-18 16:26:51 -0600 (Fri, 18 May 2007) | 1 line
Ugh. The svnmerge didn't catch the shift from cdr.c to main/cdr.c, and neither did I. This is the remainder of the 9717 patch, the fix for the run-away FAIL status for a call
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Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.
A special Thank You to WeBRainstorm that gave me access to his system.
Merged revisions 64513 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line
in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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Merged revisions 62945,63402,63519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
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we can now make 30 channels on a PRI (before we forgot chan 31..)
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r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line
don't be verbose if no need
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r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line
added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour
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added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers
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r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line
fixed the crypt_keys stuff
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r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one.
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r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line
Russell Bryant [Fri, 18 May 2007 02:51:07 +0000 (02:51 +0000)]
Merged revisions 64868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64868 | russell | 2007-05-17 21:48:51 -0500 (Thu, 17 May 2007) | 5 lines
Fix a small bug I noticed while working on something else. app_queue did not
unregister its device state monitoring callback in unload_module(). So, this
would make Asterisk crash on the first device state change after you
unload the module.
Russell Bryant [Thu, 17 May 2007 17:12:23 +0000 (17:12 +0000)]
Add an option that lets you only allow one connection at a time for each
manager user. (issue #8664, reported and original patch by ssokol, patch
updated by bkruse, and further updated by me)
Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)
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.
projects / thirdparty / asterisk.git / log
