Reverse the fix from issue #6176 and add proper handling for that issue.
(Closes issue #13962, closes issue #13363)
Fixed by myself (license 14)
........
................
Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING.
........
Use the create_vm_state_from_user function in a place where
it was not being used before. Also, I've moved the urgent
folder check in messagecount() up a bit so that the flow is
a bit better.
This was something I noticed while taking a look at issue
#13973, although I don't think this is the underlying cause
of the issue.
A possibly "horrible fix" for a "horribly broken"
situation.
As stuff shifts around in the asterisk code, the
miscellaneous inclusions from the standalone stuff
gets broken. There's no easy fix for this situation.
I made sure that everything in utils builds without
problem ***AND*** that aelparse runs the regressions
correctly with the following make menuselect options
both on and off:
I think from now on, I'm going to #undef
all these features in the various utils native
files; I guess I could do the same for the
copied-in files, surrounded by STANDALONE ifdef.
A standalone isn't going to care about threads,
mutexes, etc.
After looking through SIP registration code most of the day, this
is one of the few things I could find that was just plain wrong.
Even though it probably isn't possible for it to happen, it seems weird
to have code that checks if a pointer is NULL and then immediately dereferences
that pointer if it was NULL.
Russell Bryant [Tue, 16 Dec 2008 21:13:18 +0000 (21:13 +0000)]
Merged revisions 164877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines
Merged revisions 164876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines
Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.
This is a bug I noticed while looking at the code for app_macro. This return code
means that another thread has assumed ownership of the channel and it can no longer
be touched. (I hate this return code with a passion, by the way.)
Russell Bryant [Tue, 16 Dec 2008 20:43:16 +0000 (20:43 +0000)]
Merged revisions 164807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008) | 17 lines
Merged revisions 164806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines
Add "restart gracefully" to the AMI blacklist of CLI commands.
"module unload" was already identified as a command that can not be used
from the AMI. "restart gracefully" effectively unloads all modules, and will
run in to the same problems.
Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time.
(closes issue #13217)
Reported by: cervajs
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introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl.
(closes issue #13433)
Reported by: mvanbaak
Patches:
2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7)
with two free calls replaced with ast_free based on feedback on reviewboard
Review:
http://reviewboard.digium.com/r/91/
........
OK, Well this issue has had its share of flip-flopping.
I found the following:
1. the code in question, in ext_cmp1 in pbx.c, would not
allow two extensions that vary only by any dashes contained
within them, to be defined in the same context.
2. for input dialstrings, dashes are NOT ignored.
So, skipping them when sorting patterns seemed a bit silly.
Thus, you might declare ext 891 in a context, but
if you try dialing 8-9-1, it will NOT match 891.
So, I proposed to remove the code from ext_cmp1 to
skip the spaces and dashes. Just kept us from
declaring 891 and 8-9-1 in the same context,
forcing users to generate otherwise uselessly
obfuscated dialplan code to get the same effect.
Then, I tried out 1.4, and found that:
1. you can declare 891 and 8-9-1 in the
same context!
2. You can't define 891, and have 8-9-1 match
it! Nor can you define 8-9-1, and have 891
match it!
So, it appears that my proposal simply restores
the pbx to behaving as it did in 1.4.
Russell Bryant [Tue, 16 Dec 2008 17:15:43 +0000 (17:15 +0000)]
Merged revisions 164737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) | 22 lines
Merged revisions 164736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines
Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
One issue was that the ast_mutex_* API was being used within the context of the
thread local data destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all. This led to a memory
leak.
Another issue was an invalid argument being provided to the the object_add
API call.
(closes issue #13678)
Reported by: ys
Tested by: Russell
Be more detailed about why the include did not get included.
(closes issue #14071)
Reported by: kshumard
Patches:
pbx_config.patch.improvederroroutput.txt uploaded by kshumard (license 92)
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Russell Bryant [Tue, 16 Dec 2008 16:02:22 +0000 (16:02 +0000)]
Merged revisions 164675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines
Merged revisions 164672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines
Fix a memory leak related to the use of the "setvar" configuration option.
The problem was that these variables were being appended to the list of vars
on the sip_pvt every time a re-registration or re-subscription came in.
Since it's just a waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying the vars.
(closes issue #14037)
Reported by: marvinek
Tested by: russell
When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself.
(closes issue #13634)
Reported by: performer
........
I added a sentence to clarify why - and ' ' are ignored in patterns
as per bug 14076. Leif says he'll put some stuff about it in the
extensions.conf sample, etc.
I was getting this warning during a compile
on a 64-bit machine running ubuntu server 8.10,
and gcc-4.3.2:
[CXXi] chan_vpb.ii -> chan_vpb.oo
cc1plus: warnings being treated as errors
In file included from /home/murf/asterisk/trunk/include/asterisk/utils.h:671,
from chan_vpb.cc:46:
/home/murf/asterisk/trunk/include/asterisk/strings.h: In function ‘char* ast_str_truncate(ast_str*, ssize_t)’:
/home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: comparison between signed and unsigned integer expressions
make[1]: *** [chan_vpb.oo] Error 1
make: *** [channels] Error 2
Do not try to unlock a non-existant channel if the transfer fails.
(closes issue #13800)
Reported by: dwagner
Patches:
asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608)
........
................
Use autoconf logic to determine whether the system has timersub or not. Do not blindly assume Solaris does not.
(closes issue #13838)
Reported by: ano
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Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module.
(closes issue #14079)
Reported by: elguero
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Russell Bryant [Mon, 15 Dec 2008 16:26:09 +0000 (16:26 +0000)]
Merged revisions 164203 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r164203 | russell | 2008-12-15 08:40:24 -0600 (Mon, 15 Dec 2008) | 39 lines
Merged revisions 164201 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines
Handle a case where a call can be bridged to a channel that is still ringing.
The issue that was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once the parked
call got taken out of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing indication. This patch
fixes this case so that the caller receives ringback once it comes out of
parking until the other side answers.
The fixes are:
- Make sure we remember that a channel was an outgoing channel when doing
a masquerade. This prevents an erroneous ast_answer() call on the channel,
which causes a bogus 200 OK to be sent in the case of SIP.
- Add some additional comments to explain related parts of code.
- Update the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that are events
or commands should be ignored.
- When a bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still ringing.
- Rework ast_indicate_data() a bit for the sake of readability.
Russell Bryant [Mon, 15 Dec 2008 16:19:00 +0000 (16:19 +0000)]
Merged revisions 164272 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r164272 | russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines
When a reload is issued, always process the configuration for dundi.conf.
The reason is that a reload can be used to refresh DNS lookups for defined peers.
Even if the config file hasn't changed, we want to process it for that purpose.
* Fix reference counting used in the __queues_show function
* Add code to be sure that the "queue show" command does not
print information for a realtime queue which has been deleted
from the backend
* Add a missing unref to the realtime queue loading function for
the case where a queue is in the module's container but has been
deleted from the realtime backend
Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret.
(closes issue #14073)
Reported by: seandarcy
........
When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered.
Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a
pointer inside editline to look back to asterisk.c, so others don't spend
as much time as I did looking (in the wrong place) for the appropriate
function.
Reported by: ZX81, via the #asterisk-users channel
Fixed by: me (license 14)
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Russell Bryant [Fri, 12 Dec 2008 18:52:14 +0000 (18:52 +0000)]
Merged revisions 163670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines
Rename a number of tcptls_session variables. There are no functional changes here.
The name "ser" was used in a lot of places. However, it is a relic from when
the struct was a server_instance, not a session_instance. It was renamed since
it represents both a server or client connection.
When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found.
(closes issue #13811)
Reported by: pj
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Behaviour now is that general codec config flows to default_line and default_device. [devices] stuff amends default_device and similar for [lines]. These are copied to individual device and line as they are created.
Added confcapability and confprefs for the configured stuff which doesn't change as device and so on are connected. prefs are based on line prefs if they exist, else the device prefs are used (prefs identifies codec order).
(closes issue #13806)
Reported by: pj
Patches:
codecs.diff uploaded by wedhorn (license 30)
Tested by: pj and me
........
Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven.
(closes issue #13525)
Reported by: pj
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Russell Bryant [Fri, 12 Dec 2008 14:05:01 +0000 (14:05 +0000)]
Merged revisions 163449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines
Merged revisions 163448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines
Resolve issues that could cause DTMF to be processed out of order.
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal
is messed up. By intercepting those events with a signal handler in the remote
console, we can avoid those issues.
(closes issue #13464)
Reported by: tzafrir
Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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Russell Bryant [Thu, 11 Dec 2008 20:09:10 +0000 (20:09 +0000)]
Merged revisions 163171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r163171 | russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines
Fix the "failed" extension for outgoing calls.
The conversion to use ast_check_hangup() everywhere instead of checking the softhangup
flag directly introduced this problem. The issue is that ast_check_hangup() checked
for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances,
such as with a dummy channel.
The fix is simple. Don't check tech_pvt. It's pointless, because the code path that
sets this to NULL is when the channel hangup callback gets called. This happens inside
of ast_hangup(), which is the same function responsible for freeing the channel. Any
code calling ast_check_hangup() better not be calling it after that point, and if so,
we have a bigger problem at hand.
Sometimes even Linux needs -lm to link libtonezone, such as when libtonezone
is compiled statically.
(closes issue #13887)
Reported by: tzafrir
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Russell Bryant [Thu, 11 Dec 2008 17:11:50 +0000 (17:11 +0000)]
Merged revisions 163094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r163094 | russell | 2008-12-11 11:06:16 -0600 (Thu, 11 Dec 2008) | 19 lines
Merged revisions 163092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines
Fix an issue that made it so you could only have a single caller executing
a custom feature at a time. This was especially problematic when custom
features ran for any appreciable amount of time.
The fix turned out to be quite simple. The dynamic features are now stored
in a read/write list instead of a list using a mutex.
(closes issue #13478)
Reported by: neutrino88
Fix suggested by file
When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it.
(closes issue #13525)
Reported by: pj
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Fix reloads of aliased CLI commands. Due to changes done to turn it into a single memory allocation we can't just use the existing CLI alias structure. We have to destroy all existing ones and then create new ones.
(closes issue #14054)
Reported by: pj
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Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams.
Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
(closes issue #12560)
Reported by: vsauer
Patches:
patch001.diff uploaded by ramonpeek (license 266)
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When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
(closes issue #13599)
Reported by: hjourdain
Patches:
chan_sip.c.diff uploaded by hjourdain (license 583)
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................
This patch was originally uploaded to issue #13783 by
jamessan. While the issue was closed for other reasons, this
patch is valid and fixes a separate problem, and is thus
being committed.
Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
(closes issue #12983)
Reported by: vt
Patches:
dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
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When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server.
(closes issue #13633)
Reported by: performer
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Finish conversion to using ARRAY_LEN and remove it as a janitor project.
(closes issue #14032)
Reported by: bkruse
Patches:
14032.patch uploaded by bkruse (license 132)
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Russell Bryant [Tue, 9 Dec 2008 22:38:55 +0000 (22:38 +0000)]
Blocked revisions 162418 via svnmerge
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r162418 | russell | 2008-12-09 16:38:41 -0600 (Tue, 09 Dec 2008) | 7 lines
Add some additional Asterisk project developer documentation.
After the nightly update of the documentation on asterisk.org, I'll post
an update to asterisk-dev with a pointer to the changes. This covers some
release branch and commit policy information. None of this should be a
surprise, since it's just documenting what we have already been doing.
Russell Bryant [Tue, 9 Dec 2008 22:30:52 +0000 (22:30 +0000)]
Merged revisions 162414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) | 16 lines
Merged revisions 162413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines
Remove the test_for_thread_safety() function completely.
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
We appear to have documented tz= in the [general] section of voicemail.conf,
without actually having implemented it. Oops.
(Reported by Olivier on the -users list)
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Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing.
(closes issue #14005)
Reported by: ddl
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Russell Bryant [Tue, 9 Dec 2008 21:02:01 +0000 (21:02 +0000)]
Merged revisions 162291 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) | 17 lines
Merged revisions 162286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.
If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated.
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Merged revisions 162264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line
In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this.
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projects / thirdparty / asterisk.git / log
