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https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) | 5 lines
Make sure that tcptls_session is properly initialized.
(issue #18598)
Reported by: ksn
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r316918 | seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 lines
Look at the correct buffer for our digest info instead of an empty one.
(issue #18598)
Reported by: ksn
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r316919 | seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 lines
Use the correct HTTP method when generating our digest, otherwise we always fail.
When calculating the 'A2' portion of our digest for verification, we need the
HTTP method that is currently in use. Unfortunately our mapping function was
incorrect, resulting in invalid hashes being generated and, in turn, failures
in authentication.
(closes issue #18598)
Reported by: ksn
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r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines
Merged revisions 314620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines
Fix consistency of CRLFs on HTTP headers that get sent out.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
18186-httpheadernewline.diff uploaded by qwell (license 4)
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Guessed the log levels based on info that level 3
is the soft roof. Can we create a page / document
to define the levels?
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enable reporting of the request URI / URL in debugging
change funny debug note to a serious note.
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Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308331 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18556)
Reported by: kkm
Review: https://reviewboard.asterisk.org/r/1071/
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Address content type for BSD and other platforms
(closes issue #18456)
Reported by: alexo
Patches:
asterisk18_http.patch uploaded by alexo (license 1175)
Tested by: alexo
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305084 f38db490-d61c-443f-a65b-d21fe96a405b
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For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
escaped.
This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.
The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.
The unit tests for these functions have also been updated.
ABE-2705
Review: https://reviewboard.asterisk.org/r/1081/
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Review: https://reviewboard.asterisk.org/r/986
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279726 | pabelanger | 2010-07-26 21:53:38 -0400 (Mon, 26 Jul 2010) | 9 lines
Use ast_sockaddr_setnull() when http is not enabled.
Otherwise, ast_tcptls_server_start() will still start http.
(closes issue #17708)
Reported by: pabelanger
Patches:
http.patch uploaded by pabelanger (license 224)
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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
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If there is a problem with a firmware file, Polycom phones will close the
connection. We were continuing to send the file anyway. There should be no
reason to continue sending a file if there is an error writing it.
(closes issue #16682)
Reported by: lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270692 f38db490-d61c-443f-a65b-d21fe96a405b
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fwrite returns the number of items written, not the number of bytes
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The 'pglobal' tool is quite handy indeed :-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200620 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though.
Review: http://reviewboard.digium.com/r/237/
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Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up.
Review: http://reviewboard.digium.com/r/234/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190545 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #10961)
Reported by: ys
Patches:
digest_auth_r148468_v5.diff uploaded by ys (license 281)
SVN branch http://svn.digium.com/svn/asterisk/team/group/manager_http_auth
Tested by: ys, twilson, tilghman
Review: http://reviewboard.digium.com/r/223/
Reviewed by: tilghman,russellb,mmichelson
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(closes issue #14192)
Reported by: pabelanger
Patches:
asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176)
Tested by: jamesgolovich
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(closes issue #13990)
Reported by: eliel
Patches:
array_len.diff uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161218 f38db490-d61c-443f-a65b-d21fe96a405b
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branch, and add the ones needed for all the new code here too
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1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines
2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines)
3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines)
4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied
5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address
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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
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(closes issue #13230)
Reported by: eliel
Patches:
res_http_post_leak.patch uploaded by eliel (license 64)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul 2008) | 3 lines
accomodate users who seem to lack a sense of humor :-)
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locked.
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ago (does not affect 1.4), where you would pass
a pointer to the end of a character array, and
ast_uri_decode would do no good.
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http support
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counts so that they only get destroyed when all threads are done using
them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
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for any path that includes a directory named 'private'. This patch also
requires manager authentication for any POST's being sent to the server as well to help secure uploads.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118161 f38db490-d61c-443f-a65b-d21fe96a405b
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same name in http queries, which might confuse the manager.
Replace calls to ast_uri_decode() with a local function that also
replaces '+' with ' ', as this is the normal encoding for
spaces in http requests.
This allows passing cli commands to the manager through the
http interface.
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the same thing.
Add an ast_debug() call to help debugging the url matching.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) | 6 lines
Improve some broken cookie parsing code. Previously, manager login over HTTP
would only work if the mansession_id cookie was first. Now, the code builds
a list of all of the cookies in the Cookie header. This fixes a problem
observed by users of the Asterisk GUI.
(closes AST-20)
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comparison was dropped
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to reload any file that references a given configuration file.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@111012 f38db490-d61c-443f-a65b-d21fe96a405b
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reduce duplication of data when generating responses using ast_http_error()
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