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r198957 | seanbright | 2009-06-03 16:39:10 -0400 (Wed, 03 Jun 2009) | 11 lines
Fix a possible crash in pbx_spool.
We were trying to reference members of a struct that had previously been freed.
This patch makes sure that we free the struct after it has been removed from
the spooler queue.
(closes issue #15072)
Reported by: garlew
Patches:
spool.diff uploaded by garlew (license 376)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r198891 | dvossel | 2009-06-03 10:49:46 -0500 (Wed, 03 Jun 2009) | 10 lines
Generic call forward api, ast_call_forward()
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and res_feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop.
(closes issue #15216)
Reported by: oxymoron
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r198665 | tilghman | 2009-06-01 15:07:04 -0500 (Mon, 01 Jun 2009) | 7 lines
If using the old deprecated format, a reload would cause the class to disappear.
(closes issue #14759)
Reported by: lidocaineus
Patches:
20090518__issue14759.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved more static docs to XML (pplications and manager actions):
Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198661 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 code was there because of the AgentCallbackLogin() application.
->loginchan[] member was only used by AgentCallbackLogin().
Agent where dumped to astdb if they where logged in using AgentCallbacklogin()
so they are not being dumper anymore.
Review: https://reviewboard.asterisk.org/r/267/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The situation that caused this problem was when continuous mode was being
turned on and off while a rate was set for a timing interface. A very easy
way to replicate this bug was to do a Playback() from behind a Local channel.
In this scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about every 20 ms
as frames get queued on to the PBX side channel from the other side of the
Local channel.
Essentially, this module treated continuous mode and a set rate as mutually
exclusive states for the timer to be in. When I dug deep enough, I observed
the following pattern:
1) Set timer to tick every 20 ms.
2) Wait almost 20 ms ...
3) Continuous mode gets turned on for a queued up frame
4) Continuous mode gets turned off
5) The timer goes back to its tick per 20 ms. state but starts counting
at 0 ms.
6) Goto step 2.
Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick,
but not most of the time. This is what produced the choppy sound (or sometimes
no sound at all).
Now, the module treats continuous mode and a set rate as completely independent
timer modes. They can be enabled and disabled independently of each other and
things work as expected.
(closes issue #14412)
Reported by: dome
Patches:
issue14412.diff.txt uploaded by russell (license 2)
issue14412-1.6.1.0.diff.txt uploaded by russell (license 2)
Tested by: DennisD, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Instead of specifying every object file, use make's magic to generate
it.
This will generate less conflicts in team branches when a new file is
added in trunk.
(closes issue #15226)
Reported by: eliel
Patches:
makefile uploaded by eliel (license 64)
Review: http://reviewboard.asterisk.org/r/269/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This branch splits all the analog signaling logic out of chan_dahdi.c into
sig_analog.c. Functionality in theory should not change at all. As noted
in the code, there is still some unused code remaining that will be cleaned
up in a later commit.
Review: https://reviewboard.asterisk.org/r/253/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines
Fix 'make config' target for Slackware.
There was a missing semi-colon after the echo statement in the Makefile that was
causing problems for some users. Fix suggested by reporter.
(closes issue #15225)
Reported by: pdavis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Set the invitestate to INV_CALLING when we send a connected line reinvite.
This prevents us from potentially rapid-firing reinvites to a single peer.
* Use the astdb to store a peer's allowed methods. This prevents us from sending
an UPDATE during the interval between startup and the peer's first registration
if the peer does not support the UPDATE method.
* Handle Polycom's method of indicating allowed methods in REGISTER. Instead of
using an Allow header, they place the allowed methods in a methods= parameter
in the Contact header.
ABE-1873
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I still need to set up tests to make sure my examples are completely correct,
but I ran out of time tonight and felt that they at least would give an idea as
to how to use calendaring. I will try to test the examples and do some cleanup
on the docs tomorrow night.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize away the Local
channel when using this option.
(closes issue #14829)
Reported by: licedey
Tested by: mmichelson, licedey, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
