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(closes issue #16760)
Reported by: fiddur
Patches:
248394.diff uploaded by fiddur (license 678)i with modifications by me
Tested by: fiddur, phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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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
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(closes issue #14352)
Reported by: fiddur
Patches:
trunk-14352-2.diff uploaded by phsultan (license 73)
Tested by: fiddur
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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(closes issue #15359)
Reported by: snuffy
Patches:
bug_res_jabber_unused_defines.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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"initialize"
(closes issue #15571)
Reported by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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The 'pglobal' tool is quite handy indeed :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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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
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a result of the const-ify the world patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14405)
Reported by: snuffy
Patches:
xml_jabber.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13990)
Reported by: eliel
Patches:
array_len.diff uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- 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
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branch, and add the ones needed for all the new code here too
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13411)
Reported by: caio1982
Patches:
res_jabber_autoprune1.diff uploaded by caio1982 (license 22)
Tested by: caio1982
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Changing debug messages from VERBOSE to DEBUG channel
- Adding a few todo's
- Adding a few more "XMPP"'s to compliment Jabber...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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by Noel Morais on the users mailing list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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the corresponding roster item has a subscription value set to "none"
or "from".
Make the code more readable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) | 2 lines
Ignore appended resource when comparing JIDs.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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them to the parser ;
- report Gtalk error messages from a buddy to the console.
This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.
Thank you to PH for his great help!
(closes issue #12647)
Reported by: PH
Patches:
trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008) | 2 lines
Use keepalives effectively in order diagnose bug #12432.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008) | 1 line
Fix a potential memory leak
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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done inside iks_delete), thus making the code conform with coding guidelines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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type subscribed.
(closes issue #12066)
Reported by: ffadaie
Patches:
branch-1.4-12066-1.diff uploaded by phsultan (license 73)
trunk-12066-1.diff uploaded by phsultan (license 73)
Tested by: ffadaie, phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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integer. Modify calls to various ast_*_flag macros accordingly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ast_verb
(closes issue #11934)
Reported by: mvanbaak
Patches:
20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This authentication mechanism is implemented under the iksemel API, which
makes use of GnuTLS, whereas we use OpenSSL. Note : there's ongoing
dicsussion at the SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
http://ietfreport.isoc.org/ids-wg-sasl.html.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When the XMPP over TLS/SSL connection resets for some reason, it is
wrongly believed as being secured, which makes the re-connection
process endlessly fail. This was reported by mvanbaak in issue #11644.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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were included almost everywhere.
Remove some of the instances.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ast_variable_new parameters
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 line
In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Closes issue #9972
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Closes issue #9972, patch by phsultan. Copied from branch at http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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