to be used within the dial app, before a call is bridged.
Many thanks to sobomax for submitting this patch.
Quoting from bug 11582:
"So the goal of the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function is not well suited
for this purpose as it uses much call bridge specific data and doesn't separate a
detection of feature from a feature handler call. So a new function ast_feature_detect()
has been extracted off the ast_feature_interpret() function but keeping the original
logic intact except some insignificant changes to locking.
"Having created the ast_feature_detect() function the possibility to use feature detection
in almost any place of the asterisk code. So a call to this function has been added to
wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler
however and uses old call leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature currently is the only one
supported during call setup as other features as call parking and call transfer don't make much
sense during call setup. However if need in some of the features would arise it is much easier to
implement as the infrastructure changes are already in place with this patch."
I have cleaned up the patch somewhat, and verified that the existing functionality is not
harmed, and that the new functionality works. Terry has committed his stuff, and there were
no conflicts (see 14274).
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license 359)
patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax (license 359)
11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
supporting devices. The devices (snoms, specifically) need to receive a SIP
URI instead of just an extension. This adds that functionality.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r172639 | twilson | 2009-01-30 18:15:09 -0600 (Fri, 30 Jan 2009) | 5 lines
Rename new parkedcallparking option to parkedcallreparking
Since this option actually already existed in 1.6.0+, use the same name so as
not to confuse people when they upgrade
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
Reported by: nemo
Patches:
20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added doxygen comments to the major dahdi structures.
* Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
* Fixed PRI not handling unknown TON/NPI prefix letters correctly.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Also, change a function in app.c to return a userful value instead of always returning 0.
Patch by fnordian, changed by Corydon76 and myself.
This does not close the bug report, as fnordian had an additional change we're still discussing.
(related to issue #14059)
Reported by: fnordian
Patches:
chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines
Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.
The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!
(closes issue #14294)
related to issue #13385
Reported by: klaus3000 and adomjan
Patches:
bug14294b.diff uploaded by oej (license 306)
Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing
will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines
Fix devicestate problems for "always-on" agent channels
A revision to chan_agent attempted to "inherit" the device
state of the underlying channel in order to report the
device state of an agent channel more accurately.
The problem with the logic here is that it makes no sense to
use this for always-on agents. If the agent is logged in, then
to the underlying channel, the agent will always appear to be
"in use," no matter if the agent is on a call or not. The reason
is that to the underlying channel, the channel is currently in use
on a call to the AgentLogin application.
The most common cause that I found for this issue to occur was for
a SIP channel to be the underlying channel type for an Agent channel.
If the SIP phone re-registers, then the registration will cause the
device state core to query the device state of the SIP channel. Since the
SIP channel is in use, the Agent channel would also inherit this status.
Once the agent channel was set to "in use" there was no way that the device
state could change on that channel unless the agent logged out.
The solution for this problem is a bit different in 1.4 than it is in the
other branches. In 1.4, there will be a one-line fix to make sure that only
callback agents will inherit device state from their underlying channel type.
For the other branches of Asterisk, since callback support has been removed, there
is also no need for device state inheritance in chan_agent, so I will simply be
removing it from the code.
In addition, the 1.4 source is getting a new comment to help the next person who
edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be
used to determine if the agent is a callback agent or not.
(closes issue #14173)
Reported by: nathan
Patches:
14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines
Prevent a crash from occurring when a jitter buffer interpolated frame is
removed from a slinfactory
slinfactory used the "samples" field of an ast_frame in order to determine
the amount of data contained within the frame. In certain cases, such as
jitter buffer interpolated frames, the frame would have a non-zero value for
"samples" but have NULL "data"
This caused a problem when a memcpy call in ast_slinfactory_read would attempt
to access invalid memory. The solution in use here is to never feed frames into
the slinfactory if they have NULL "data"
(closes issue #13116)
Reported by: aragon
Patches:
13116.diff uploaded by putnopvut (license 60)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.
To fix this problem, there are two measures taken
1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c
(closes issue #14260)
Reported by: ccesario
Patches:
14260.patch uploaded by putnopvut (license 60)
Tested by: ccesario
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Merged revisions 171527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines
Use the same branch tag in CANCEL as in INVITE
Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems.
Thanks Fredrik for pointing out where the bug in the SIP messaging was.
(closes issue #14346)
Reported by: oej
Patches:
bug14346.diff uploaded by oej (license 306)
Tested by: oej
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r171452 | russell | 2009-01-26 15:31:59 -0600 (Mon, 26 Jan 2009) | 13 lines
Resolve some synchronization issues in chan_iax2 scheduler handling.
The important changes here are related to the synchronization between threads
adding items into the scheduler and the scheduler handling thread. By adjusting
the lock and condition handling, we ensure that the scheduler thread sleeps no
longer and no less than it is supposed to. We also ensure that it does not
wake up more often than it has to.
There is no bug report associated with this. It is just something that I found
while putting scheduler thread handling into a reusable form (review 129).
Review: http://reviewboard.digium.com/r/131/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Steve Murphy
Russell Bryant
Olle Johansson
Sean Bright
Tilghman Lesher
Terry Wilson
Mark Michelson
Richard Mudgett
Leif Madsen
Matthew Fredrickson
David Vossel
Doug Bailey