passed directly into the HASH() function. Requested via the -users list, and
committed at Astricon in the Code Zone.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) | 29 lines
(closes issue #13489)
Reported by: DougUDI
Tested by: murf
(closes issue #13490)
Reported by: seanbright
Tested by: murf
(closes issue #13467)
Reported by: edantie
Tested by: murf, edantie, DougUDI
This crash happens because we are unsafely handling old pointers.
The channel whose cdr is being handled, has been hung up and
destroyed already. I reorganized the code a bit, and tried not
to lose the fork-cdr-chain concepts of the previous code.
I now verify that the 'previous' channel (the channel we
had when the bridge was started), still exists, by looking it up
by name in the channel list. I also do not try to reset the
CDR's of channels involved in bridges.
Testing shows it solves the crash problem, and should not
negatively impact previous fixes involving CDR's generated
during/after blind transfers. (The reason we need to reset
the CDR's on the "beginning" channels in the first place).
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144067 f38db490-d61c-443f-a65b-d21fe96a405b
host portion in the Contact URI and specifies a
transport, the parsing done in parse_moved_contact
resulted in a malformed URI.
This commit fixes the parsing so that a proper
Dial string may be formed when the forwarded
call is placed.
(closes issue #13523)
Reported by: mattdarnell
Patches:
13523v2.patch uploaded by putnopvut (license 60)
Tested by: mattdarnell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144025 f38db490-d61c-443f-a65b-d21fe96a405b
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r143964 | murf | 2008-09-23 08:22:10 -0600 (Tue, 23 Sep 2008) | 21 lines
In at least one machine, we noted that the timestr
was not getting set in the STMT; it was coming out,
usually, as binary garbage to an mssql server.
These changes fixed the problem. The only thing
I can venture forth as a guess, is that the pointer
is being stored in the interface, not a copy of the
string. Because we ripped the build process into a
subroutine, the timestr became a temp. stack variable,
and between the time the STMT got built and the
time it was executed on the server, the string being
pointed to was damaged. At any rate, even if this
theory is false, and some mechanism was at fault,
this fix worked reliably where it didn't before.
Why this bug didn't bite last week, I have no idea.
This change basically defines the timestr buffer
in the calling function, extending the life of the
buffer to cover both the STMT's building and
processing to the server.
I AM BLOCKING THIS FIX FROM TRUNK, because we
tested out a trunk release, and it is working
fine.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143974 f38db490-d61c-443f-a65b-d21fe96a405b
(closes issue #13524)
Reported by: wedhorn
Patches:
unload.diff uploaded by wedhorn (license 30)
With small tweak by me to prevent a crash
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143799 f38db490-d61c-443f-a65b-d21fe96a405b
subscription if we have previously subscribed. Otherwise
a segfault will occur.
(closes issue #13476)
Reported by: jonnt
Patches:
13476.patch uploaded by putnopvut (license 60)
Tested by: jonnt
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143609 f38db490-d61c-443f-a65b-d21fe96a405b
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r143475 | jpeeler | 2008-09-18 16:01:23 -0500 (Thu, 18 Sep 2008) | 1 line
compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143476 f38db490-d61c-443f-a65b-d21fe96a405b
is set, don't bother searching for a region to free, just
immediately exit.
This has the dual benefit of suppressing a warning message
about freeing memory at (nil) and of optimizing the free()
replacement by not having to do any futile searching for
the proper region to free.
(closes issue #13498)
Reported by: pj
Patches:
13498.patch uploaded by putnopvut (license 60)
Tested by: pj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143400 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) | 6 lines
Allow for "G.729" if offered in an SDP even though
it is not RFC 3551 compliant. Some Cisco switches
will send this in an SDP, and it doesn't hurt to
be able to accept this.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143340 f38db490-d61c-443f-a65b-d21fe96a405b
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r143270 | jpeeler | 2008-09-16 16:44:21 -0500 (Tue, 16 Sep 2008) | 9 lines
(closes issue #13493)
Reported by: mdu113
Patches:
bug13493.diff uploaded by jpeeler (license 325)
Tested by: mdu113
Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced.
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r143204 | jpeeler | 2008-09-16 11:14:25 -0500 (Tue, 16 Sep 2008) | 10 lines
(closes issue #13465)
(closes issue #13425)
Reported by: mdu113
Patches:
bug13425.diff uploaded by jpeeler (license 325)
Tested by: mdu113
The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel.
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when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines
Tested by: sergee, murf, chris-mac, andrew, KNK
This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from
the ground up!
This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.
Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.
While I dearly hope that this patch overcomes all problems, and
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.
** trunk note: some code to suppress the h exten being run
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.
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r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | 20 lines
(closes issue #13364)
Reported by: mdu113
Well, fundamentally, the problems revealed in 13364 are
because of the ForkCDR call that is done before the dial.
When the bridge is in place, it's dealing with the first
(and wrong) cdr in the list.
So, I wrote a little func to zip down to the first non-locked
cdr in the chain, and thru-out the ast_bridge_call, these
results are used instead of raw chan->cdr and peer->cdr pointers.
This shouldn't affect anyone who isn't forking cdrs before a
dial, and should correct the cdr's of those that do.
So, this change ends up correcting the dstchannel
and userfield; the disposition was fixed by a previous
patch, it was OK coming into this problem.
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r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | 30 lines
(closes issue #12318)
Reported by: krtorio
I made a small change to the code that handles local channel situations.
In that code, I copy the answer time from the peer cdr, to the bridge_cdr,
but I wasn't also copying the disposition from the peer cdr.
So, Now I copy the disposition, and I've tested against
these cases:
1. phone 1 never answers the phone; no cdr is generated at all.
this should show up as a manager command failure or something.
2. phone 2 never answers. CDR is generated, says NO ANSWER
3. phone 2 is busy. CDR is generated, says BUSY
4. phone 2 answers: CDR is generated, times are correct; disposition
is ANSWERED, which is correct. The start time is the time that
the manager dialed the first phone. The answer time is the time
the second phone picks up.
I purposely left the cid and src fields blank; since this call really
originates from the manager, there is no 'easy' data to put in these
fields. If you feel strongly that these fields should be filled in,
re-open this bug and I'll dig further.
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r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed, 10 Sep 2008) | 9 lines
Fix detection of PWLIB and OpenH323 version when spacing in the headers
isn't consistent.
(closes issue #13426)
Reported by: bamby
Patches:
detect_openh323.diff uploaded by bamby (license 430)
(Modified by me to use sed instead of tr)
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r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) | 7 lines
It is a normal situation that a task gets put in the scheduler that should run
as soon as possible. Accept "0" as an acceptable time to run, and also treat
negative as "run now", and don't print a debug message about it.
(inspired by a message asking about the "request to schedule in the past"
debug message on the -dev list)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142355 f38db490-d61c-443f-a65b-d21fe96a405b
issue 12979. The difference between this and the
1.6.0 and 1.6.1 versions is that this is a much more
invasive change. With this, we completely get rid
of the interfaces list, along with all its helper
functions.
Let me take a moment to say that this change personally
excites me since it may mean huge steps forward regarding
proper lock order in app_queue without having to strew
seemingly unnecessary locks all over the place. It also
results in a huge reduction in lines of code and complexity.
Way to go Brett!
(closes issue #12979)
Reported by: sigxcpu
Patches:
20080710_issue12979_queue_custom_state_interface_trunk_2.diff uploaded by bbryant (license 36)
Tested by: sigxcpu, putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142146 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep 2008) | 21 lines
When determining if codecs used by SIP peers allow
the media to be natively bridged, use the jointcapability
instead of the peercapability.
It seems that the intent of using the peercapability was to
expand the choice of codecs for the call to increase the
chances of being able to native bridge the channels. The
problem is that if a codec were settled on for the native
bridge and that wasn't a codec that was configured to be used
by Asterisk for that peer, then Asterisk would send a
REINVITE with no codecs in the SDP which is a bug no matter
how you slice it.
(closes issue #13076)
Reported by: ramonpeek
Patches:
13076.patch uploaded by putnopvut (license 60)
Tested by: tbelder
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r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines
Ensure that the stored CDR reference is still valid after the bridge before
poking at it. Also, keep the channel locked while messing with this CDR.
(fixes crashes reported in issue #13409)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142064 f38db490-d61c-443f-a65b-d21fe96a405b
messages weren't showing up when I had set the debug level
high for just app_queue.c. It's because we were only checking
the global option_debug variable instead of using the awesome
macro which checks both the global and file-specific value
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@141998 f38db490-d61c-443f-a65b-d21fe96a405b
crichter