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necessary to allow for a sip_pvt to maintain a reference
to a sip_peer's outboundproxy even after the peer has
been freed.
(closes issue #13700)
Reported by: fnordian
Patches:
13700.patch uploaded by putnopvut (license 60)
Tested by: fnordian
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Ao2 world...
We're going to try to get time to fix this and kpfleming believes that there's code in ao2
so that we can solve it...
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct 2008) | 3 lines
fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r149266 | mmichelson | 2008-10-14 18:43:58 -0500 (Tue, 14 Oct 2008) | 4 lines
Change this warning to an error message. Suggestion
comes from Sean Bright. Thanks Sean!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines
Call register_peer_exten even in the case that the peer's
IP/port does not change.
(closes issue #13309)
Reported by: dimas
Patches:
v2-13309.patch uploaded by dimas (license 88)
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leaks found with these changes.
(Closes issue #13505, closes issue #13543)
Reported by: mav3rick, triccyx
Patches:
20081001__bug13505.diff.txt uploaded by Corydon76 (license 14)
Tested by: mav3rick, triccyx
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines
Don't allow reserved characters to be used in register
lines in sip.conf.
(closes issue #13570)
Reported by: putnopvut
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct 2008) | 9 lines
Deadlock prevention in chan_local.
(closes issue #13676)
Reported by: tacvbo
Patches:
13676.patch uploaded by putnopvut (license 60)
Tested by: tacvbo
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Bump the reference count up before we add it to the scheduler, duh.
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it; has_to_tag was not cleared, which caused the second incoming call over a TCP socket to fail if pedantic checking was enabled
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- Fixing language
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(found at SIPit)
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in the "kill the user" commit and caused calls relying
on the insecure setting to not work properly. I changed
for finding a peer back to how it was prior to that
commit.
(closes issue #13644)
Reported by: pj
Patches:
13644_trunkv2.patch uploaded by putnopvut (license 60)
Tested by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148376 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well.
(closes issue #13668)
Reported by: mjc
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Thanks rmudget for letting me know and providing hints on how to fix it best.
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Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
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making the new/old message count available to local devices. (issue #AST-77)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines
when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected)
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so we don't have to hardcode anything.
(closes issue #13636)
Reported by: seanbright
Patches:
13636.diff uploaded by seanbright (license 71)
Reviewed by: russellb, putnopvut
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copy_request accesses dst->data w/o being able to tell whether it is proerly initialized
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r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
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r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06 Oct 2008) | 9 lines
Check whether an extension exists in the _call method, rather than the _alloc
method, because we need to evaluate the callerid (since that data affects
whether an extension exists).
(closes issue #13343)
Reported by: efutch
Patches:
20080915__bug13343.diff.txt uploaded by Corydon76 (license 14)
Tested by: efutch
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r146448 | qwell | 2008-10-05 16:17:44 -0500 (Sun, 05 Oct 2008) | 1 line
Fix silly formatting.
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Reported by: pj
Tested by: pj
Set transport to SIP_TRANSPORT_UDP mode if not specified which fixes calls to get_transport returning UNKNOWN.
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call-id in the dialog-info event package used with extension state subscriptions
on Snom phones. Then, when the phone sends an INVITE with Replaces for the
special callid, Asterisk will perform a pickup on the extension that was
subscribed to.
The original code on this issue was submitted by xylome. However, contributions
have been made by (at least) mgernoth and pkempgen. The final patch was written
by seanbright, and includes the necessary logic to allow this work in a
technology independent way.
(closes issue #5014)
Reported by: xylome
Patches:
issue5014-trunk.diff uploaded by seanbright (license 71)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
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This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
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Reported by: nickpeirson
The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;
chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008) | 1 line
improved helptext of misdn_set_opt.
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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
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(closes issue #13524)
Reported by: wedhorn
Patches:
unload.diff uploaded by wedhorn (license 30)
With small tweak by me to prevent a crash
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1 line
A micro-fix, in sip_park_thread, where d is freed before the func is done using it.
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r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15 Sep 2008) | 6 lines
Set the raw formats at the same time as the other formats.
(closes issue #13240)
Reported by: jvandal
Patches:
20080813__bug13240.diff.txt uploaded by Corydon76 (license 14)
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(closes issue #13452)
Reported by: pj
Patches:
memleak5.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, pj, mvanbaak
(closes issue #13294)
Reported by: pj
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(closes issue #13354)
Reported by: adomjan
Patches:
20080828__bug13354.diff.txt uploaded by Corydon76 (license 14)
20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, adomjan
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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)
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r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12 Sep 2008) | 6 lines
(closes issue #12965)
Reported by: rlsutton2
Prevents local channels from playing MOH at each other which was causing ast_generic_bridge to loop much faster.
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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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