There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r339298 | jrose | 2011-10-04 09:09:50 -0500 (Tue, 04 Oct 2011) | 19 lines
Merged revisions 339297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines
Reverting revision 333265 due to component connection problems it introduces.
I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this
problem, but first it seems prudent to remove this rather broad attempt to fix it and
instead approach this problem either from the same angle but looking only at canceling
(or possibly rescheduling) the send when we absolutely know it will cause a segfault
or, if that can't be easily accomplished, strictly from the devstate side of things.
Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe.
(issue ASTERISK-18626)
(issue ASTERISK-18078)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r333410 | jrose | 2011-08-26 11:28:03 -0500 (Fri, 26 Aug 2011) | 19 lines
Merged revisions 333378 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | 13 lines
[patch] Buddies are always auto-registered when processing the roster
Reporter said autoregister flag was ignored for registering 'buddies' which
had a subscription to us. Verified that this was the case and observed how
the patch addressed this and made sure it didn't break anything.
(closes issue ASTERISK-14233)
Reported by: Simon Arlott
Patches:
asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott
Tested by: Jonathan Rose
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r333266 | jrose | 2011-08-25 14:00:05 -0500 (Thu, 25 Aug 2011) | 20 lines
Merged revisions 333265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines
Segfault when publishing device states via XMPP and not connected
When using publishing device state with res_jabber, Asterisk will attempt
to send a device state using the unconnected client using iks_send_raw
and crash. This patch checks the validity of the connection before
attempting to send the device state.
(closes issue ASTERISK-18078)
Reported by: Michael L. Young
Patches:
res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young
Tested by: Jonathan Rose
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
Add the attribute "type" to each "<use>" for menuselect.
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines
Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL.
This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings.
(closes issue #18759)
Reported by: bklang
Patches:
null-strings.patch uploaded by bklang (license 919)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r291905 | twilson | 2010-10-15 09:39:58 -0700 (Fri, 15 Oct 2010) | 14 lines
Merged revisions 291904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) | 7 lines
Don't crash or deadlock on module unload
We can't hold the lock while pthread_join is called since aji_log_hook will
attempt to lock from the other therad. We reorder the pthread_join and
ast_aji_disconnect so that we don't do an SSL_read() while SSL_shutdown is
running, causing a crash.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new JabberStatus event gives a concise view of the status change to the AMI
clients. Thanks fiddur!
(closes issue #16760)
Reported by: fiddur
Patches:
244498.2.diff uploaded by fiddur (license 678)
Tested by: fiddur, phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
aji_io_recv takes the maximum number of bytes to read (instead of the total
buffer size), so we have to subtract 1 from our buffer size. Without this, when
we receive packets that are larger than our buffer, iksemel will choke and
things get wonky.
(closes issue #15232)
Reported by: lp0
Patches:
05302009_res_jabber.c.patch uploaded by seanbright (license 71)
Tested by: seanbright, lp0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines
Properly terminate AMI JabberSend response messages.
The response message (either Error or Success) needs an extra trailing \r\n
after the fields to inform the client that the message is complete.
(closes issue #14876)
Reported by: srt
Patches:
05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71)
asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Terry Wilson
Tilghman Lesher
Kinsey Moore
Jonathan Rose
Kevin P. Fleming
Leif Madsen
Russell Bryant
Paul Belanger
Brad Watkins
David Vossel
Philippe Sultan
Jeff Peeler
Sean Bright
David Brooks
Eliel C. Sardanons
Michiel van Baak