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r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines
Merged revisions 180006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines
Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank"
value for a sound file in queues.conf and have no sound played
back. The problem with this is that it would result in some ugly
CLI warnings from file.c.
This commit introduces a check when playing a file in app_queue
to see if the name of the file is zero-length and return early if
that is the case. Also, the ability to specify the blank sound
files in queues.conf is now mentioned more clearly in queues.conf.sample
(closes issue #14227)
Reported by: caspy
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r179972 | dvossel | 2009-03-03 16:01:24 -0600 (Tue, 03 Mar 2009) | 13 lines
app_meetme not setting filename and fileformat correctly for realtime
When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults.
(closes issue #14545)
Reported by: dalbaech
Patches:
app_meetme-realtime5.patch uploaded by dvossel (license 671)
Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705)
Tested by: dvossel, dalbaech
Review: http://reviewboard.digium.com/r/180/
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r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines
Add documentation for timing modules used in Asterisk
This document specifies the timing modules available in Asterisk beginning
with Asterisk 1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used for in
Asterisk. There is also a section which can be used to help customize
your setup or to troubleshoot timing issues you may have.
I also added messages to the DAHDI timing test used in res_timing_dahdi.c
that points to this new documentation if people experience problems.
Big thanks to all who contributed comments on this.
(closes issue #14490)
Reported by: mmichelson
Patches:
timing.txt uploaded by mmichelson (license 60)
Review: http://reviewboard.digium.com/r/164/
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r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar 2009) | 1 line
fix a leaked channel lock (and future deadlock) when we try to pick up our own channel
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r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines
Merged revisions 179840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines
Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing.
It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves
the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to.
We can not safely modify it afterwards because of this, so don't even try.
(closes issue #14564)
Reported by: meric
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r179745 | mmichelson | 2009-03-03 11:03:47 -0600 (Tue, 03 Mar 2009) | 8 lines
Convert pbx_spool to use string fields instead of statically-sized buffers.
In tests run after making this conversion, I noticed an approximate 85%
reduction in memory usage for call file processing.
Review: http://reviewboard.digium.com/r/168/
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r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines
Merged revisions 179741 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines
Ensure chan->fdno always gets reset to -1 after handling a channel fd event.
Since setting fdno to -1 had to be moved, a couple of other code paths that
do process an fd event return early and do not pass through the code path
where it was moved to. So, set it to -1 in a few other places, too.
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r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines
Merged revisions 179671 via svnmerge from
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r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines
Move where fdno is set to the default value to *after* the read callback of the channel driver is called.
We have to do this as the underlying channel driver may need the fdno value to determine what to read.
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r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines
Merged revisions 179608 via svnmerge from
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r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines
Make it easier to detect an improper call to ast_read().
When you call ast_waitfor() on a channel, the index into the channel fds array
that holds the file descriptor that poll() determines has input available is
stored in fdno. This patch clears out this value after a call to ast_read()
and also reports errors if ast_read() is called without an fdno set.
From a discussion on the asterisk-dev list.
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r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines
Merged revisions 179536 via svnmerge from
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r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines
Fix bridging regression from commit 176701
This fixes a bad regression where the bridge would exit after an attended
transfer was made. The problem was due to nexteventts getting set after the
masquerade which caused the bridge to return AST_BRIDGE_COMPLETE.
(closes issue #14315)
Reported by: tim_ringenbach
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r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines
Merged revisions 179532 via svnmerge from
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r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines
Move ast_waitfor() down to avoid the results of the API call becoming stale.
This call to ast_waitfor() was being done way too soon in this section of code.
Specifically, there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice. By putting the channel
in autoservice, the previous results of ast_waitfor() become meaningless,
as the autoservice thread will do it's own ast_waitfor() and ast_read()
on the channel.
So, when we came back out of autoservice and eventually hit the block of code
that calls ast_read() on the channel, there may not actually be any input on
the channel available. Even though the previous call to ast_waitfor() in
app_meetme said there was input, the autoservice thread has since serviced
the channel for some period of time.
This bug manifested itself while dvossel was doing some testing of MeetMe in
Asterisk trunk. He was using the timerfd timing module. When the code hit
ast_read() erroneously, it determined that it must have been called because of
input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was
the cause of the last legitimate call to ast_read() done by autoservice.
In this test, an IAX2 channel was calling into the MeetMe conference. It was
_much_ more likely to be seen with an IAX2 channel because of the way audio
is handled. Every audio frame that comes in results in a call to
ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify
the channel thread that a frame is waiting to be handled. So, the chances
of ast_waitfor() indicating that a channel needs servicing due to a timer
event on an IAX2 event is very high.
Finally, it is interesting to note that if a different timing interface was
being used, this bug would probably not be noticed. When ast_read() is called
and erroneously thinks that there is a timer event to handle, it calls the
ast_timer_ack() function. The pthread and dahdi timing modules handle the
ack() function being called when there is no event by simply ignoring it.
In the case of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read. This caused Asterisk
to lock up very quickly.
Thanks to dvossel and mmichelson for the fun debugging session. :-)
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r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines
Make the sip_standard_port function more granular by allowing separate
type and port arguments. This is necessary because when building our From
and Contact headers, we need to be absolutely sure that we are placing our
source port there and not the peer's source port.
(closes issue #12761)
Reported by: asbestoshead
Patches:
patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455)
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r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines
Merged revisions 179468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines
When ending a recording with silence detection, remember to reduce the duration.
The end of the recording is correspondingly trimmed, but the duration was not
trimmed by the number of seconds trimmed, so the saved duration was necessarily
longer than the actual soundfile duration.
(closes issue #14406)
Reported by: sasargen
Patches:
20090226__bug14406.diff.txt uploaded by tilghman (license 14)
Tested by: sasargen
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r179465 | russell | 2009-03-02 17:06:16 -0600 (Mon, 02 Mar 2009) | 4 lines
Fix a reference leak in timerfd_set_rate().
(found during a debugging session with dvossel and mmichelson.)
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r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines
Merged revisions 179461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines
Ensure that only one thread is calling ast_settimeout() on a channel at a time.
For example, with an IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so twice at the
same time is a bad thing.
(Found in a debugging session with dvossel and mmichelson)
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r179323 | file | 2009-03-02 10:28:09 -0400 (Mon, 02 Mar 2009) | 5 lines
Do not try to remove a registration scheduled item if the scheduler context has already been destroyed.
(closes issue #14580)
Reported by: alecdavis
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r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines
Swap reversed timevals.
This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
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This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to
this commit.
(closes issue #14338)
Reported by: fiddur
Patches:
14338.patch uploaded by mmichelson (license 60)
Tested by: fiddur
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r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines
Properly free memory and remove scheduler entries when a transmission failure occurs.
Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit
was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called,
this inevitably resulted in the reading and writing of freed memory.
XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet
at all. The proper action to take is to remove the scheduler entry we just created,
free the packet's data as well as the packet itself, and unlink it from the list of
packets on the sip_pvt structure.
(closes issue #14455)
Reported by: Nick_Lewis
Patches:
14455.patch uploaded by mmichelson (license 60)
Tested by: Nick_Lewis
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r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines
Mark res_ais as experimental, as the binary event format is subject to change.
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r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines
Add a note about the ordering of entries in sip.conf in 1.6.1.
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r179122 | mvanbaak | 2009-02-27 21:34:00 +0100 (Fri, 27 Feb 2009) | 16 lines
Add reload support to chan_skinny.
Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the way
we configure devices and lines.
(closes issue #10297)
Reported by: DEA
Patches:
skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
With mods by me based on feedback from wedhorn and Russell and seanbright
Tested by: DEA, mvanbaak, pj
Review: http://reviewboard.digium.com/r/130/
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r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines
Merged revisions 178956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In this case, it's just a matter of reducing the default timeouts from 2000
to 1000 msec, as the max def feature digit timeout is no longer halved.
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r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
As per bug 14515, a dev discussion arrived at a "mediated concensus"
of a default feature digit timeout of 1.0 sec. Some voted for 1300;
ctooley thought 1500 for distracted phone users in phone booths;
kpfleming put his foot down at 1.0 sec.
Users who found the previous default max delay of 250 msec perfect,
are welcome to override the new default. Notice that I said that
250 msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for 14515,
we found that 500 msec was actually enforcing a max of 250. The bug
fix would restore 500 msec, but we felt even that was a bit tight
for most users... 2000 msec was pushed earlier by mmichelson, so
that reduces to 1000 msec after the bug fix. Enjoy!
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r178870 | murf | 2009-02-26 10:45:22 -0700 (Thu, 26 Feb 2009) | 1 line
These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x.
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r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines
Merged revisions 178804 via svnmerge from
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r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines
This patch prevents the feature detection timeout from being cut in half.
Because the ast_channel_bridge() call will return 0 and pass
a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
field in hte config struct is getting decremented twice, which
effectively cuts the digittimeout in half. I added conditions
to the if statement to only let DTMF_END frames to flow thru,
which solved the problem. Also, when the frame pointer is null,
let control flow thru-- this usually happens on timeouts. I added
a comment to the code to explain what's going on and why.
Many thanks to sodom for reporting this problem. Personnally, it always seemed
like something was wrong with the featuredigittimeout, but I never
could quite decide what... and was too busy to investigate.
This bug forced the issue, and now we know.
Sodom had other issues in 14515, but I couldn't reproduce them. If
he still has problems, and wants to get them solved, he is welcome
to reopen 14515.
(closes issue #14515)
Reported by: sodom
Patches:
14515.patch uploaded by murf (license 17)
Tested by: murf, sodom
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r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines
Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed.
(closes issue #14541)
Reported by: grant
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r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines
IAX2 prune realtime fix
Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
(closes issue #14479)
Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/
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r178764 | file | 2009-02-26 11:40:10 -0400 (Thu, 26 Feb 2009) | 5 lines
Ensure there is a valid tone part before trying to play tones.
(closes issue #14558)
Reported by: alecdavis
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r178300 | dvossel | 2009-02-24 11:42:37 -0600 (Tue, 24 Feb 2009) | 14 lines
Allows manager command to see if IAX link is trunked and encrypted. Displays what kind of encryption is enabled as well.
Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.
(closes issue #14427)
Reported by: snuffy
Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
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r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines
Merged revisions 178205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
Skip check for extension when subscribing for MWI.
Since the remote side is not actually subscribing to a specific extension when
subscribing for MWI just skip the check to see if the extension exists. They can't use it
to specify the mailbox either since we require configuration of that in sip.conf
(closes issue #14531)
Reported by: festr
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r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines
Merged revisions 178141 via svnmerge from
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r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
(closes issue #14460)
Reported by: moliveras
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r178030 | dvossel | 2009-02-23 11:59:55 -0600 (Mon, 23 Feb 2009) | 7 lines
Changes the way keyrotation is enabled by default
Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled.
(closes issue #14523)
Reported by: mvanbaak
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r178061 | mvanbaak | 2009-02-23 19:23:38 +0100 (Mon, 23 Feb 2009) | 3 lines
update the new manager commands in chan_skinny to match
chan_sip's headers. requested by oej.
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