This patch does two things:
1. It moves the debug statement that shows the HTTP sub-protocols being
compared after the string length calculation such that it shows the correct
string length in the output
2. It adds some additional debug that displays when it matches on a
sub-protocol and when it fails
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The recent changes to update stasis_cache_topics directly from the
publisher thread uncovered a race condition, which was causing asserts
in the /stasis/core tests.
If the caching topic's subscription is the last reference to the
caching topic, it will destroy the caching topic after the final
message has been processed. When dispatching to a different thread,
this usually gave the unsubscribe enough time to finish before
destruction happened. Now, however, it consistently destroys before
unsubscription is complete.
This patch adds an extra reference to the caching topic, to hold it
for the duration of the unsubscription.
This patch also removes an extra unref that was happening when the
final message was received by the caching topic. It was put there
because of an extra ref that was put into the caching topic's
constructor. Both have been removed, which makes the destructor a bit
less confusing.
Review: https://reviewboard.asterisk.org/r/2675/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394686 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If there is an error streaming an audio file, the current return status makes it
difficult for an AGI script to determine that there was an error with the audio
file.
This patches changes the result to return -1 and the function returns
RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of
res_agi, this would appear to be the proper way to handle an error.
(closes issue ASTERISK-21903)
Reported by: Ariel Wainer
Tested by: Ariel Wainer
Patches:
asterisk-21903-return-stream-res_1.8.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2625/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since ast_hangup() is effectively a channel destructor, it should be a
void function.
* Make the few silly callers checking the return value no longer do so.
Only the CDR and CEL unit tests checked the return value.
* Make all callers take advantage of the NULL safe change and remove the
NULL check before the call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.
This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.
This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).
It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.
Review: https://reviewboard.asterisk.org/r/726/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability to kick all users out of a conference from the
ConfBridge kick CLI command. It is invoked by passing 'all' as the channel
parameter to the CLI command, i.e., "confbridge kick <conf> all".
Note that this patch was modified slightly to conform to trunk.
(closes issue ASTERISK-21827)
Reported by: dorianlogan
patches:
kickall-patch_v2.diff uploaded by dorianlogan (License 6504)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a channel is hungup, both an APP_END event and a HANGUP event can be
fired. To ensure that HANGUP events occur after APP_END events, the method
callbacks for the APP_END event should be processed prior to the callbacks
for the HANGUP event.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This ensures that code that was only meant to be run on a reinvite failure
only runs on a reinvite failure.
(closes issue ASTERISK-22061)
reported by Rusty Newton
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch attempts to fix some possible race conditions in shutdown of the
CDR engine. It:
* Adds a cleanup handler to only unsubscribe and join on stasis messages during
graceful shutdown. The cleanup handler should execute before the regular atexit
handler, as we want to unsubscribe for any further messages before dispatching
the CDRs.
* The CDRs are now locked when we dispatch them on shutdown.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2657/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When QUEUE_MEMBER is used and the member specified is not in the queue,
Asterisk provides an ERROR message that indicates that the option specified
is not valid. This patch now properly displays an ERROR message that the
member is not in the queue if an interface is specified.
(closes issue ASTERISK-21980)
Reported by: Avraam David
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Peter J Philipp pointed out that there are two checks that ensure that len is
not less than 0. If len is less than 0, the function returns. Having both of
them is clearly redundant.
This removes the second and attempts to clarify (slightly) the error condition.
(closes issue ASTERISK-21772)
Reported by: Peter J Philipp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is not apparent to the average user that the PASSTHRU function should not
be passed as ${PASSTHRU(string)} but just as PASSTHRU(string) to functions
which take a variable name and not its contents.
This patch clarifies the behavior in the documentation and provides an example.
(closes issue ASTERISK-21717)
Reported by: Richard Miller
patches:
func_strings.diff uploaded by Richard Miller (license 5685)
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Merged revisions 394303 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It simplifies the Dial handling in CDRs. As a rule, the caller in a dial
relationship is always the Party A. There was some logic present in the
handling of the dial message that could, conceivably, pick the caller
as Party A for the beginning of the dial and the peer as Party A for the
end of the dial. This shouldn't have happened if the code in the bridging
framework was doing its job; however, that was broken and it led to the
FRACK. As it is, this code was overly ocmplex and not needed: the caller,
if present, should always be Party A. Period.
* It properly checks to see if a channel will continue on in the dialplan.
ast_check_hangup - much like cake at the end - is a lie. It will tell
you that you are hungup when you are not. Do not believe it.
I would make this function tell the truth, but I'm nervous that we've been
depending on it sitting on its throne of lies for far too long, and it would
probably break lots of things. So I'm just checking the "internal" soft
hangup flags, like everyone else.
(closes issue ASTERISK-22060)
Reported by: Mark Michelson
(issue ASTERISK-21831)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Instead of formatting a NULL pointer into a "%s" format string (which is
usually not a good thing to do), we instead print "Unknown".
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
from mixing different variants or general MFC-R2 settings within the same E1 line.
Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or
the whole E1 has the same country variant and R2 settings.
In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1.
This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every
time a new channel => section is found and the configuration was changed.
(closes issue ASTERISK-21117)
Reported by: Rafael Angulo
Related Elastix issue: http://bugs.elastix.org/view.php?id=1612
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The JSON test attempted an overly clever use of RAII_VAR to run code
at the beginning and end of each test, in order to validate that no
JSON objects were leaked during the test.
The problem is that the validation code would run during the initial
load, when the tests were initialized. This happens during startup,
when other parts of the system might actively be allocating and
freeing JSON objects.
This patch changes the RAII_VAR to use the new
ast_test_register_{init,cleanup} functions to run the validations
properly.
(closes issue ASTERISK-21978)
Review: https://reviewboard.asterisk.org/r/2669/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Major changes are to pull all packet reading functions into skinny_session
and move timeout handling to scheduling arrangements. Thread cancelling is
now undertaken directly rather than waiting for the read to timeout
(cleanup is popped on thread cancel). Also added some keepalive timings in
debugging messages.
Keepalive timeout has been increased from 1.1 by keepalive to 3 times
keepalive. This seems to align (after keepalives stabilise) with when
devices reset after not receiving keepalives. Probably needs more work,
especially around the first and/or second keepalives that vary
significantly by device and firmware version.
Review: https://reviewboard.asterisk.org/r/2611/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When I corrected the CEL test crash in r394037, I didn't quite pay attention
to how the globals and locals were being shuffled around in the cleanup
callback. I removed the nulling of the global variables, which caused them
to be double cleaned.
This patch puts the global nulling code back (since the vars are cleaned up
by RAII_VARs), and removes the explicit ao2_cleanup() (since they were no-ops,
because the variables had just been nulled).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When I initially wrote the configuration support for ARI users, I
determined the section type by a category prefix (i.e., [user-admin]).
This is neither idiomatic Asterisk configuration, nor is it really
that user friendly. This patch replaces the category prefix with a
type field in the section, which is much cleaner.
Review: https://reviewboard.asterisk.org/r/2664/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The voicemail_api test had code like strncmp(a, b, sizeof(a)), but a was a
char pointer, instead of a literal or char array. This meant that sizeof was
the size of the pointer, not the length of the string.
Since the string is in a stringfield and should be null terminated, I just
changed it to a plain strcmp.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
** This script is in no way finished.
Started the initial "cut" at converting a sip.conf file to a res_sip.conf file.
Hopefully the bulk of the framework is in place and only a few minor adjustments
need to be made when an option mapping is added that "doesn't fit". This script
and supporting files should be executable against python version 2.5.
An OrderedDict class (backported from a newer version of python) is included.
A MultiOrderedDict class is implemented so options, when added, should be able
to be added in order and allowed to have multiple values.
Currently the scripts supports the majority of endpoint options found in
res_sip.conf. Support has also been added for Aor(s) and the ACL/security
sections. Inside the sip_to_res_sip.py file one can see a list of options
that still need to be mapped.
Also items that still need to be done: templates, includes, parsing '=>'
delimiter. Note that some code is hopefully in place already to support
templates (e.g. lookup/retrieving defaults from them). However, the
parsing of and adding of the section needs to be done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SLA code within app_meetme was written before asotbj2 had been
merged into Asterisk. Worse, support for reloads did not exist at first
and was added later as a bolt-on feature. I knew at the time that
reloading was not safe at all while SLA was in use, so the reload would
be queued up to execute when the system was idle. Unfortunately, this
approach was still prone to errors beyond the fact that this was the
only place in Asterisk where configuration was not reloaded
instantly when requested.
This patch converts various SLA objects to be reference counted objects
using astobj2. This allows reloads to be processed while the system is
in use. The code ensures that the objects will not disappear while one
of the other threads is using them. However, they will be immediately
removed from the global trunk and station containers so no new calls
will use them if removed from configuration.
Review: https://reviewboard.asterisk.org/r/2581/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matthew Jordan
David M. Lee
Michael L. Young
Richard Mudgett
Jonathan Rose
Tzafrir Cohen
Mark Michelson
Joshua Colp
Moises Silva
Jason Parker
Damien Wedhorn
Kevin Harwell
Russell Bryant