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
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: pj
Tested by: pj
Set transport to SIP_TRANSPORT_UDP mode if not specified which fixes calls to get_transport returning UNKNOWN.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@144025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(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: https://origsvn.digium.com/svn/asterisk/trunk@143799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@143609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines
Fix pedantic mode of chan_sip to only check the
remote tag of an endpoint once a dialog has
been confirmed. Up until that point, it is possible
and legal for the far-end to send provisional
responses with a different To: tag each time. With
this patch applied, these provisional messages
will not cause a matching problem.
(closes issue #11536)
Reported by: ibc
Patches:
11536v2.patch uploaded by putnopvut (license 60)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line
This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes applied by this patch:
- Fix autocompletion in 'sip prune realtime', sip peers where never auto completed. Now we complete this command with:
'sip prune realtime peer' -> all | like | sip peers
Also I have modified the syntax in the usage, was wrong...
- Pass ast_cli_args->argv and ast_cli_args->argc while running autocompletion on CLI commands (CLI_GENERATE).
With this we avoid comparisons on ast_cli_args->line like this:
strcasestr(a->line, " description")
strcasestr(a->line, "descriptions ")
strcasestr(a->line, "realtime peer"), and so on..
Making the code more confusing (check the spaces in description!).
The only thing we must be sure is to first check a->pos or a->argc.
- Fix 'iax2 prune realtime' autocompletion, now we autocomplete this command with 'all' & 'iax2 peers', check a look that iax2 peers where all the peers, now only the ones in the cache..
(closes issue #13133)
Reported by: eliel
Patches:
clichanges.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
