https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines
Prevent phantom calls to queue members.
If a caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would incorrectly
indicate that the caller was still in the queue. With these changes,
the problem does not occur.
(closes issue #14631)
Reported by: latinsud
Patches:
queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
(with small modification from me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Park() was not respecting the arguments passed to it. Any extension/context/priority
given to it was being ignored. This patch remedies this.
(closes issue #15380)
Reported by: DLNoah
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r205149 | russell | 2009-07-08 10:54:21 -0500 (Wed, 08 Jul 2009) | 8 lines
Make OpenSSL usage thread-safe.
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@205150 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
This change allows applications that request T.38 negotiation on a channel that
does not support it to get the proper indication that it is not supported, rather
than thinking that negotiation was started when it was not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines
Removed confusing warning message "Got Busy in Connected State"
If an incoming mISDN call is answered with the Answer application and a
subsequent Dial gets a busy endpoint then it is valid for that already
connected channel to get the busy indication. Asterisk will play the busy
tones until the dialplan plays something else or hangs up the call.
(closes issue #11974)
Reported by: fvdb
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
(closes issue #15022)
Reported by: greenfieldtech
Patches:
20090519__issue15022.diff.txt uploaded by tilghman (license 14)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
First of all, the code was unnecessary. The goal was to lock a channel
which was already locked. Second, the assumption of the deadlock avoidance
loop was that the sip_pvt was already locked and we were trying to get the
channel lock. The problem is that the sip_pvt was unlocked a few lines above.
Basically, I'm removing 5 lines of no-op.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ensure that add-on modules can be embedded, fix up Makefile.moddir_rules
to allow module directory Makefiles to more easily specify the modules to
be built, and explicitly list the addons modules in its Makefile, since
the module names don't follow any pattern.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines
Add error message so that it is clear why a SIP peer was not processed when
a DNS lookup fails on a host or outboundproxy.
(closes issue #13432)
Reported by: p_lindheimer
Patches:
outboundproxy.patch uploaded by p (license 558)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines
Fix a problem where chan_sip would ignore "old" but valid responses.
chan_sip has had a problem for quite a long time that would manifest when
Asterisk would send multiple SIP responses on the same dialog before receiving
a response. The problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two requests out,
and a response arrived for the first request sent, then Asterisk would ignore
the response. The result was that Asterisk would continue retransmitting the
requests and ignoring the responses until the maximum number of retransmissions
had been reached.
The fix here is to rearrange the code a bit so that instead of simply comparing
the sequence number of the response to our latest outgoing sequence number, we
walk our list of outstanding packets and determine if there is a match. If there is,
we continue. If not, then we ignore the response.
In doing this, I found a few completely useless variables that I have now removed.
(closes issue #11231)
Reported by: flefoll
Review: https://reviewboard.asterisk.org/r/298
........
r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines
Fix build oops.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also, the code in this module is horrendous and we should remove it from the
tree. I'm not sure who is supposed to be maintaning this thing, but they
clearly are not. I don't see the sense of leaving it in the main tree. If it
lives *anywhere* it should be in addons.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This might seem like a legitimate comment that merely needed semicolon
prefixes, but in reality, the adaptive layer is designed to allow arbitrary
CDR variables, without needing the use of a userfield to store multiple items.
It's therefore not only invalid syntax but also goes against the intent of the
adaptive method.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
David Vossel
Mark Michelson
Jason Parker
David Brooks
Tilghman Lesher
Sean Bright
Russell Bryant
Luigi Rizzo
Matthew Nicholson
Kevin P. Fleming
Richard Mudgett
Ryan Brindley