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This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.
To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.
I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.
* Move JSON support from res_json.c to main/json.c
* Made libjansson-dev a required dependency
* Added an ast_channel_blob message type, which has a channel
snapshot and JSON blob of data.
* Changed UserEvent and Newexten events so that they are dispatched
via ast_channel_blob messages on the channel's topic.
* Got rid of the ast_channel_varset message; used ast_channel_blob
instead.
* Extracted the manager functions converting Stasis channel events to
AMI events into manager_channel.c.
(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Unsubscribe from MWI stasis event on channel reload.
(closes issue ASTERISK-21216)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-mwiind02.diff uploaded by snuffy (license 5024)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Thanks, rmudgett!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Review: https://reviewboard.asterisk.org/r/2403/
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Review: https://reviewboard.asterisk.org/r/2402/
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Because of the way that topics were handled when publishing, it was
possible to dispatch a message to a subscription after that
subscription had been unsubscribed such that the dispatched message
arrived at the callback after the callback had received its final
message. In callbacks that cleaned up user data, this would often cause
a segfault. This has been resolved by locking the topic during the
entirety of dispatch. To prevent long publishing and topic locking
times, forwarding subscriptions have been made to be standard
subscriptions instead of mailboxless subscriptions which were
dispatched at publishing time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When using scoped locks, the unref of an AO2 object should happen after
the unlock occurs which requires usage of scoped refs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r383341 | dlee | 2013-03-19 10:57:29 -0500 (Tue, 19 Mar 2013) | 5 lines
Removed codecs/g722/*.i on make clean
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Merged revisions 383340 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r383342 | dlee | 2013-03-19 10:58:33 -0500 (Tue, 19 Mar 2013) | 1 line
Remove codecs/speex/*.i on make clean
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.
Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This change allows you to use XMPP buddy state in places where device state
can be used be used, such as dialplan hints. If at least one resource is
available the buddy is considered available. Now your phone can reflect
their IM status too!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Often times, when subscribing to a topic, one wants to handle
different message types differently. While one could cascade if/else
statements through the subscription handler, it is much cleaner to
specify a different callback for each message type. The
stasis_message_router is here to help!
A stasis_message_router is constructed for a particular stasis_topic,
which is subscribes to. Call stasis_message_router_unsubscribe() to
cancel that subscription.
Once constructed, routes can be added using
stasis_message_router_add() (or stasis_message_router_set_default()
for any messages not handled by other routes). There may be only one
route per stasis_message_type. The route's callback is invoked just as
if it were a callback for a subscription; but it only gets called for
messages of the specified type.
(issue ASTERISK-20887)
Review: https://reviewboard.asterisk.org/r/2390/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue ASTERISK-20887)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Unsubscribing things in Asterisk seems to very commonly follow with
NULLing out the variable that was unsubscribed. This change makes that
a bit simpler.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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AMI, HTTP, and chan_sip all support TLS in some way, but none of them
support all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from setting
TLS/SSL options that they do not support.
This also gets tlsverifyclient closer to a working state by requesting
the client certificate when tlsverifyclient is set. Currently, there is
no consumer of main/tcptls.c in Asterisk that supports this feature and
so it can not be properly tested.
Review: https://reviewboard.asterisk.org/r/2370/
Reported-by: John Bigelow
Patch-by: Kinsey Moore
(closes issue AST-1093)
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When a session timer expires during a dialog that has re-negotiated to T.38
and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP
containing audio media only. This causes some hilarity with the poor fax
session under weigh.
This patch corrects that by sending T.38 parameters if we are in the middle of
a T.38 session.
(closes issue ASTERISK-21232)
Reported by: Nitesh Bansal
patches:
dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418)
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Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In certain situations, call files are not processed when using KQueue with
pbx_spool. Asterisk was sending an invalid timeout value when the spool
directory is empty, causing the call to kevent to error immediately. This
can create a tight loop, increasing the CPU load on the system.
(closes issue ASTERISK-21176)
Reported by: Carlton O'Riley
patches:
kqueue_osx.patch uploaded by coriley (License 6473)
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Not informing the RTP engine of the instance data creates shrapnel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Push some cleanups upstream before testing another ticket.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When retrieving the parking lots from a MySQL database table, the current order
is "filename, cat_metric desc, var_metric asc, category". If there are multiple
parking lots with the same cat_metric but different categories, everything is
being sorted on cat_metric first resulting in errors when loading the parking
lots.
This patch fixes the problem by sorting on the category field first, then the
cat_metric field.
(closes issue ASTERISK-21035)
Reported by: Alex Epshteyn
Patches:
asterisk-21035-orderby.diff Michael L. Young (license 5026)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This commit updates some fields in the contributed realtime schema files to
handle IPv6 addresses.
(closes issue ASTERISK-21173)
Reported by: Torrey Searle
Patches:
realtime_sql.patch Torrey Searle (license 5334)
asterisk-21173-update-ip-fields.diff Michael L. Young (license 5026)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue ASTERISK-21156)
Reported by: amsoft2001
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ICE/STUN/TURN support in res_rtp_asterisk is also now optional.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed
to include the Username field. Somehow, one of the events was missed. This
patch corrects that - the Username field should be included in all AMI Registry
events involving SIP registrations.
(issue ASTERISK-17888)
(closes issue ASTERISK-21201)
Reported by: Dmitriy Serov
patches:
chan_sip.c.diff uploaded by Dmitriy Serov (license 6479)
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Merged revisions 382848 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Fix issue with 'unistim show info' CLI command when device connected not configured
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Added an option "discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event. This essentially stops telling the peer
to start music on hold.
(issue ABE-2899)
Reported by: Denis Alberto Martinez
Review: https://reviewboard.asterisk.org/r/2336/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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struct conference_bridge_user -> struct confbridge_user
struct conference_bridge -> struct confbridge_conference
struct conference_state -> struct confbridge_state
struct conference_bridge_user *conference_bridge_user -> struct confbridge_user *user
struct conference_bridge_user *cbu -> struct confbridge_user *user
struct conference_bridge *conference_bridge -> struct confbridge_conference *conference
The names are now generally shorter, consistently used, and don't conflict
with the struct names.
This patch handles the renaming part of the issue.
(issue ASTERISK-20776)
Reported by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Prior to this change, certain conditions for sending the message would
result in an address of '(null)' being used in the via header of the
SIP message because a NULl value of pvt->ourip was used when initially
generating the via header. This is fixed by adding a call to build_via
when the address is set before sending the message.
(closes issue ASTERISK-21148)
Reported by: Zhi Cheng
Patches:
700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475)
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(issue ASTERISK-20887)
(issue ASTERISK-20959)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Fixes test failure introduced in r382685.
(issue ASTERISK-20887)
(issue ASTERISK-20959)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The cache dump mechanism allows the developer to retreive multiple
items of a given type (or of all types) from the cache residing in a
stasis caching topic in addition to the existing single-item cache
retreival mechanism. This also adds to the caching unit tests to
ensure that the new cache dump mechanism is functioning properly.
Review: https://reviewboard.asterisk.org/r/2367/
(issue ASTERISK-21097)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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For the initial use of this bus, I took some work kmoore did creating
channel snapshots. So rather than create AMI events directly in the
channel code, this patch generates Stasis events, which manager.c uses
to then publish the AMI event.
This message bus provides a generic publish/subscribe mechanism within
Asterisk. This message bus is:
- Loosely coupled; new message types can be added in seperate modules.
- Easy to use; publishing and subscribing are straightforward
operations.
In addition to basic publish/subscribe, the patch also provides
mechanisms for message forwarding, and for message caching.
(issue ASTERISK-20887)
(closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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After r382670, get_ip_and_port_from_sdp was no longer used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Originally, way back in r201583, we added the alternate RTP address so
that the RTP engine would expect to receive audio from a new source
when a glare re-INVITE occurred. In r382589, we remove the alternate
RTP source, as the 'secret' probation mode allows for switching to a new
RTP source when a previous source stops sending RTP. At the time, it
seemed appropriate to set the RTP source based on the information in the
glared re-INVITE.
Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs
with no SDP - such as in a connected line update that glances - we'll set
the RTP source to an invalid address. In subsequent re-INVITE requests from
this Asterisk instance, we'll then send an invalid media address, which will
result in the remote side sending a 488. Whoops.
There isn't any need to reset the RTP source - if we're using strictrtp, we'll
simply synchronize to a new source when we stop getting packets from the old
one. If we aren't using strictrtp, then again there shouldn't be a problem.
Note that the Asterisk Test Suite's connectedline test caught this error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot
that Urgent also "counts" as new messages. This fixed the problem when any of
the three folders was specified and the combine option was used.
It missed the case where the folder isn't specified and we build a snapshot of
all folders. This patch corrects that.
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Merged revisions 382617 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Another instance of attribute retrieval not being freed properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Many places that allocated to pull out an attribute are now freed
properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Often, Asterisk may realize that a change in the source of an RTP stream is
about to occur and ask that the RTP engine reset it's lock on the current RTP
source. In certain scenarios, it may take awhile for the new remote system to
send RTP packets, while the old remote system may continue providing RTP during
that time period. This causes Asterisk to re-lock onto the old source, thereby
rejecting the new source when the old source stops sending RTP and the new
source begins.
This patch prevents that by having a constant secondary, 'secret' probation
mode enabled when an RTP source has been chosen. RTP packets from other sources
are always considered, but never chosen unless the current RTP source stops
sending RTP.
Review: https://reviewboard.asterisk.org/r/2364
(closes issue AST-1124)
Reported by: John Bigelow
Tested by: John Bigelow
(closes issue AST-1125)
Reported by: John Bigelow
Tested by: John Bigelow
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Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Strings retrieved via ast_xml_get_text() must be freed with
ast_xml_free_text().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Messages sent while the logger thread is shutting down will now have
their associated callid freed properly.
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Merged revisions 382574 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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If ast_threadpool_set_size with a size equal to the current size, a
reference to a set_size_data structure would be leaked.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Ownership of the listener reference is not transferred because the
listener is reffed when placed into the taskprocessor. Ensure that the
listener is dereffed properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.
A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.
Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/
(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
oolong-path-support-trunk in team branch by oej (License 5267)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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