When the "ignorebusy" setting was deprecated, we added some code to allow us to
be compatible with older setups that are still using the "ignorebusy" setting
instead of "ringinuse". We set a char *variable with the column name to use,
which helps the realtime functions to use the correct column in their SQL
queries. When "persistentmembers" is enabled, we are not setting this variable
before the realtime functions were called to load members. This results in the
variable being NULL and therefore causing a segfault when loading members during
the module's process of loading.
The solution was to move the code that sets that variable to be before these
realtime functions are called during the loading of the module.
(closes issue ASTERISK-21738)
Reported by: JoshE
Tested by: JoshE
Patches:
asterisk-21738-rt-ringinuse-field-not-set.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2499/
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A GCC bug[1] can, in some cases, pop up an unsuppressible pedwarn when
using a static inline standard library function from a non-static
inline function.
This normally doesn't show up, but can occur if you're running an
upgrade version of GCC (such as GCC 4.8 on OS X, which normally runs
GCC 4.2).
[1]: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The <arpa/nameser_compat.h> was introduced way back in OS X Panther, which
itself was end-of-lifed back in 2007. We can assume that any OS X machine
we build on will need that header file :-)
Why bother removing it? The flag we're checking (__APPLE_CC__) is actually
Apple's build number. Self-compiled versions of GCC (such as installing the
latest version of GCC from homebrew) sets the value to 0, making it useless
for this sort of compile flaggery.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, a call to ast_load_realtime_multientry could get away with
passing a NULL parameter to the function, even though it really isn't
supposed to do that. After the change over to using ast_variable instead
of variadic arguments, the realtime engine gets unhappy if you do this.
This was always an unintended function call in app_directory anyway - now,
we just don't call into the realtime function calls if we don't have anything
to query on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we first introduced the channel blob types, the JSON blobs were
self identifying by a required "type" field in the JSON object
itself. This, as it turns out, was a bad idea.
When we introduced the message router, it was useless for routing based
on the JSON type. And messages had two type fields to check: the
stasis_message_type() of the message itself, plus the type field in the
JSON blob (but only if it was a blob message).
This patch corrects that mistake by removing the required type field
from JSON blobs, and introducing first class stasis_message_type objects
for the actual message type.
Since we now will have a proliferation of message types, I introduced a
few macros to help reduce the amount of boilerplate necessary to set
them up.
Review: https://reviewboard.asterisk.org/r/2509
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An endpoint is an external device/system that may offer/accept
channels to/from Asterisk. While this is a very useful concept for end
users, it is surprisingly not a core concept within Asterisk itself.
This patch defines ast_endpoint as a separate object, which channel
drivers may use to expose their concept of an endpoint. As the channel
driver creates channels, it can use ast_endpoint_add_channel() to
associate channels to the endpoint. This updated the endpoint
appropriately, and forwards all of the channel's events to the
endpoint's topic.
In order to avoid excessive locking on the endpoint object itself, the
mutable state is not accessible via getters. Instead, you can create a
snapshot using ast_endpoint_snapshot_create() to get a consistent
snapshot of the internal state.
This patch also includes a set of topics and messages associated with
endpoints, and implementations of the endpoint-related RESTful
API. chan_sip was updated to create endpoints with SIP peers, but the
state of the endpoints is not updated with the state of the peer.
Along for the ride in this patch is a Stasis test API. This is a
stasis_message_sink object, which can be subscribed to a Stasis
topic. It has functions for blocking while waiting for conditions in
the message sink to be fulfilled.
(closes issue ASTERISK-21421)
Review: https://reviewboard.asterisk.org/r/2492/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription
The problem is that the State Notify requests rely on the 200OK reponse for pacing control
and to not confuse the notify susbsystem.
The issue is, the pendinginvite isn't cleared if a response isn't received,
thus further notify's are never sent.
The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure.
(closes issue ASTERISK-21677)
Reported by: Dan Martens
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2475/
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Reload support was originally not included for SLA. It was added later,
but in a fairly non-traditional way. It basically sets a flag
indicating that a reload is pending, and then waits for a time where it
thinks everything SLA related is idle and unused, and *then* executes
the reload. It does this because the reload process is destructive. It
starts by throwing everything away and starting over.
There are a number of problems with this approach. One of them is that
the check to see if anything in use was incomplete. This patch makes it
more complete and thus less likely for a crash to occur during reload
processing. However, this approach still has problems so some much more
significant reworking of this code will need to come in as a next step.
Patch credit and testing by CoreDial, LLC.
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This change adds the ability for modules to add themselves as observers
to sorcery object types. Observers can be notified when objects are
created, updated, or deleted as well as when the object type is loaded or
reloaded. Observer notifications are done using a thread pool in a serialized
fashion so the caller of the sorcery API calls is minimally impacted.
This also adds the ability to create JSON changesets of a sorcery object.
Tests are also present to confirm all of the above functionality.
Review: https://reviewboard.asterisk.org/r/2477/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch:
* Cleans up some doxygen
* Prevents leaking the system level Stasis topics and messages
on exit (users of valgrind will be happier)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes the direct call to AMI from the SHARED function
and instead call Stasis-Core. Stasis-Core delivers the notification
that a shared variable has changed on a channel to all interested
consumers.
(issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves VarSet events for local variables raised by GoSub
over to Stasis-Core. It also tweaks up the post-processing documentation
scripts to not combine parameters if both parameters are already documented.
(issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 4028 Section 10
if the side not performing refreshes does not receive a
session refresh request before the session expiration, it SHOULD send
a BYE to terminate the session, slightly before the session
expiration. The minimum of 32 seconds and one third of the session
interval is RECOMMENDED.
Prior to this asterisk would refresh at 1/2 the Session-Expires interval,
or if the remote device was the refresher, asterisk would timeout at interval end.
Now, when not refresher, timeout as per RFC noted above.
(closes issue ASTERISK-21742)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2488/
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RFC 4028 Section 7.2
"UACs MUST be prepared to receive a Session-Expires header field in a
response, even if none were present in the request."
What changed
After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered
a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher.
Symptom:
After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device
may respond with a much lower Session-Expires (180 in our case) value that it is now using.
Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE.
After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the
refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response.
Fix:
handle_response_invite() when 200OK, remove check for outbound and reinvite.
(closes issue ASTERISK-21664)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2463/
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* Remove t and ama local variables. There is no way they could be
anything other than default because p->owner can only be NULL at this
point.
* Rename tmp and tmp2 to owner and chan respectively.
* Remove redundant initialization of channel context, exten, priority.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In certain situations, when the RTP engine goes to send a DTMF end digit
it may be in a situation where the remote address is no longer available,
or the digit that was supposed to be sent is invalid. In such cases, we
need to clear the RTP counters appropriately. Otherwise, when the RTP
source is set again, we'll continue to think that we're in the middle of
sending a DTMF digit, which can confuse the remote party (signficantly).
(closes issue ASTERISK-21522)
Reported by: Corey Farrell
patches:
rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909)
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Pretending that chan_local locals container can have more than one bucket
is silly. The container has no key to help search.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a reason the heap comparison functions like qsort(), and other
comparison functions specify <0, >0, and =0 for the return values.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When you have lots of SIP peers (according to the issue reporter, around 3500),
the 'sip show peers' CLI command or AMI action can crash due to a poorly placed
string duplication that occurs on the stack. This patch refactors the command
to not allocate the string on the stack, and handles the formatting of a single
peer in a separate function call.
(closes issue ASTERISK-21466)
Reported by: Guillaume Knispel
patches:
fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492)
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Way back when in the dark days of Asterisk 1.8.9, blind transferring a call
in a context that included the 'h' extension would inadvertently execute the
hangup code logic on the transferred channel. This was a "bad thing". The fix
was to properly check for the softhangup flags on the channel and only execute
the 'h' extension logic (and, in later versions, hangup handler logic) if the
channel was well and truly dead (Jim).
Unfortunately, CDRs are fickle. Setting the softhangup flag when we detected
that the channel was leaving the bridge (but not to die) caused some crucial
snippet of CDR code, lying in ambush in the middle of the bridging code, to
not get executed. This had the effect of blowing away one of the CDRs that is
typically created during a blind transfer.
While we live and die by the adage "don't touch CDRs in release branches", this
was our bad. The attached patch restores the CDR behavior, and still manages to
not run the 'h' extension during a blind transfer (at least not when it's
supposed to).
Thanks to Steve Davies for diagnosing this and providing a fix.
Review: https://reviewboard.asterisk.org/r/2476
(closes issue ASTERISK-21394)
Reported by: Ishfaq Malik
Tested by: Ishfaq Malik, mjordan
patches:
fix_missing_blindXfer_cdr2 uploaded by one47 (License 5012)
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Michael L. Young
David M. Lee
Matthew Jordan
Richard Mudgett
Alec L Davis
Jason Parker
Russell Bryant
Joshua Colp
Jonathan Rose
Sean Bright