https://origsvn.digium.com/svn/asterisk/trunk
........
r164801 | murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines
(closes issue #14076)
Reported by: toc
Tested by: murf
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r164733 | file | 2008-12-16 12:50:11 -0400 (Tue, 16 Dec 2008) | 6 lines
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r164525 | russell | 2008-12-15 16:25:46 -0600 (Mon, 15 Dec 2008) | 6 lines
Open a timer before loading configuration so that the trunking configuration option
will take effect.
(closes issue #14082)
Reported by: seandarcy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r164415 | murf | 2008-12-15 12:42:05 -0700 (Mon, 15 Dec 2008) | 16 lines
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
which this fix silences
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
................
r164344 | file | 2008-12-15 13:46:32 -0400 (Mon, 15 Dec 2008) | 10 lines
Blocked revisions 164343 via svnmerge
........
r164343 | file | 2008-12-15 13:43:59 -0400 (Mon, 15 Dec 2008) | 4 lines
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r164312 | file | 2008-12-15 13:24:28 -0400 (Mon, 15 Dec 2008) | 4 lines
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r164309 | russell | 2008-12-15 11:21:38 -0600 (Mon, 15 Dec 2008) | 2 lines
Fix a couple more build issues related to ast_str_opaque
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
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.
(closes issue #13747)
Reported by: davidw
Tested by: russell
Review: http://reviewboard.digium.com/r/90/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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.
(closes issue #13776)
Reported by: kombjuder
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r164268 | mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 lines
Fix up a few issues with regards to queues
* 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
(closes issue #14033)
Reported by: cristiandimache
Patches:
14033.patch uploaded by putnopvut (license 60)
Tested by: cristiandimache
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r163951 | seanbright | 2008-12-12 22:00:26 -0500 (Fri, 12 Dec 2008) | 1 line
Use actual tables instead of ASCII art ones.
........
r163952 | seanbright | 2008-12-12 22:03:15 -0500 (Fri, 12 Dec 2008) | 1 line
This shouldn't have gotten commited. We might want to generate this into a separate file instead of the version controlled one.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r163912 | file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines
Only detach and destroy the whisper audiohooks if they are actually in use.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r163873 | twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines
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.
(closes issue #14034)
Reported by: cristiandimache
Tested by: otherwiseguy, cristiandimache
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r163828 | russell | 2008-12-12 17:06:55 -0600 (Fri, 12 Dec 2008) | 2 lines
Add a note to indicate why this only supports one channel for now.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r163762 | tilghman | 2008-12-12 16:04:26 -0600 (Fri, 12 Dec 2008) | 14 lines
Merged revisions 163761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines
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)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r163629 | file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r163582 | mvanbaak | 2008-12-12 18:14:13 +0100 (Fri, 12 Dec 2008) | 11 lines
Fix codec capability setup in chan_skinny
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
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.
(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r163384 | tilghman | 2008-12-11 17:38:56 -0600 (Thu, 11 Dec 2008) | 16 lines
Merged revisions 163383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r163254 | russell | 2008-12-11 15:48:08 -0600 (Thu, 11 Dec 2008) | 16 lines
Merged revisions 163253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) | 8 lines
Fix some observed slowdowns in dialplan processing.
The change is to remove autoservice usage from dialplan functions that do not
need it because they do not perform operations that potentially block.
(closes issue #13940)
Reported by: tbelder
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r163241 | russell | 2008-12-11 15:21:31 -0600 (Thu, 11 Dec 2008) | 8 lines
Fix a problem where continuous mode will get inadvertently get turned off if set_rate()
is used while continuous mode was already turned on.
(closes issue #13738)
Reported by: smurfix
Patches:
res.patch.fixed uploaded by smurfix (license 547)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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.
(closes issue #14035)
Reported by: erogoza
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
