Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling
ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the
NOTIFY request that informs the transferor if the transfer succeeded or failed.
This patch also clears said flag from the off nominal NOTIFY paths in the
local_attended_transfer code, as once we've sent the NOTIFY request it is safe
to send by the BYE request.
This was caught by the blind-transfer-accountcode test in the Asterisk Test
Suite.
(closes issue ASTERISK-23290)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3214/
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* Move route code to sip/route.c + sip/include/route.h
* Rename functions to sip_route_*
* Replace ad-hoc list code with macro's from linkedlists.h
* Create sip_route_process_header() to processes Path and Record-Route headers
(previously done with different code in build_route and build_path)
* Add use of const where possible
* Move struct uriparams, struct contact and contactliststruct from sip.h to
reqresp_parser.h. sip/route.c uses reqresp_parser.h but not sip.h, this was
a problem. These moved declares are not used outside of reqresp_parser.
* While modifying reqprep() the lack of {} caused me trouble. I added them.
* Code outside route.c treats sip_route as an opaque structure, using macro's
or procedures for all access.
(closes issue ASTERISK-22582)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3173/
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Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
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This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
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The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
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This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal. That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...". This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.
(closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek
Review: https://reviewboard.asterisk.org/r/2988/
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While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing. The display of
"N" used to mean NAT (i.e. yes). The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear. That was a great suggestion.
Therefore, this patch does the following:
* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.
* A column for the 'Comedia' setting has been added. It too will display the
setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.
* UPGRADE.txt has been updated to document this change.
(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
asterisk-forcerport-display-clarification_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2941
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A condition was added in a commit to fix ASTERISK-21374, that, if the
SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT
flag to the dialog. This condition should not have been there since it assumed
that if Asterisk is in an environment where NAT is involved, that the auto_* nat
settings or force_rport setting would be on in the global settings. If the nat
setting in the global setting is set to 'nat=no' and then turned on for peers
(which is not quite the recommended way, although it is allowed) this flag is
never copied to the dialog resulting in problems like, REGISTER replies going
to the wrong port.
This patch removes this conditional check and will now always use the peer's
flag which by this point in the code the checks on whether the peer is behind
NAT or not (if using auto_force_rport) have already been run.
(closes issue ASTERISK-22236)
Reported by: Filip Frank
Tested by: Michael L. Young
Patches:
asterisk-2236-always-set-rport.diff uploaded
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2919/
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When a 200 OK for an initial INVITE is received, we were doing
the right thing by ACKing and sending an immediate BYE. However,
we also were doing the wrong thing and queuing an answer frame,
thus causing the call to be answered. This would cause the call
to be hung up by the channel thread, thus resulting in a second
BYE being sent out.
In this fix, I also have set the hangupcause to be correct since
the initial BYE being sent by Asterisk had an unknown hangup
cause. I have changed to using "Bearer capabilty not available"
since the call was hung up due to an SDP offer/answer error.
(closes issue ASTERISK-22621)
reported by Kinsey Moore
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Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
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r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
Minor performance bump by not allocate manager variable struct if we don't need it
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r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
Stasis performance improvements
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
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r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
Taskprocessor optimization; switch Stasis to use taskprocessors
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
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r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
Optimize how Stasis forwards are dispatched
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
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r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
Remove dispatch object allocation from Stasis publishing
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
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When Asterisk receives a 200 OK in response to an invite, that peer should have
sent an SDP at some point by then. If the channel has never received an SDP,
media won't have been set and the remote address won't be known. Endpoints in
general should not be doing this. This patch makes it so that Asterisk will
simply hang up a call if it sends a 200 OK at this point. So far this odd
behavior for endpoints has only been observed in tests which involved manually
created SIP transactions in SIPp.
(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/
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1st Issue
When a realtime peer sends an un-REGISTER request, Asterisk
un-registers the peer but the database table record still has regseconds and
fullcontact for the peer. This results in calls attempting to be routed to the
peer which is no longer registered. The expected behavior is to get
busy/congested when attempting to call an un-registered peer through the
dialplan.
What was discovered is that we are clearing out the peer's registration in the
database in parse_register_contact() when calling expire_register() but then
upon returning from parse_register_contact(), update_peer() is run which stores
back in the database table regseconds and fullcontact.
2nd Issue
The reporter pointed out that the 200 ok being returned by Asterisk
after un-registering a peer contains a Contact header with ;expires= and the
Expires header is not set to 0. This is actually a regression.
Tests were created for this second issue (ASTERISK-22548). The tests have been
reviewed and a Ship It! was received on those tests.
This patch does the following:
* Do not ignore the Expires header value even when it is set to 0. The patch
sets the pvt->expiry earlier on in the function so that it is set properly and
used.
* If pvt->expiry is 0, do not call update_peer since that means the peer has
already been un-registered and there is no need to update the database record
again since nothing has changed.
(closes issue ASTERISK-22428)
Reported by: Ben Smithurst
Tested by: Ben Smithurst, Michael L. Young
Patches:
asterisk-22428-rt-peer-update-and-expires-header.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2869/
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The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
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With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].
This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.
For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.
Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)
The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.
Other changes made as a part of this patch:
* The stubs for http_websocket that wrap system calls set errno to
ENOSYS.
* res_http_websocket now properly increments module use count.
* In loader.c, the while() wrappers around dlclose() were removed. The
while(!dlclose()) is actually an anti-pattern, which can lead to
infinite loops if the module you're attempting to unload exports a
symbol that was directly linked to.
* The special handling of nonoptreq on systems without weak symbol
support was removed, since we no longer rely on weak symbols for
optional_api.
[1]: https://wiki.asterisk.org/wiki/x/wACUAQ
(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
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If the SIP channel driver processes an invalid SDP that defines media
descriptions before connection information, it may attempt to reference
the socket address information even though that information has not yet
been set. This will cause a crash.
This patch adds checks when handling the various media descriptions that
ensures the media descriptions are handled only if we have connection
information suitable for that media.
Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing
the solution to this problem.
(closes issue ASTERISK-22007)
Reported by: wdoekes
Tested by: wdoekes
patches:
issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674)
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A remote exploitable crash vulnerability exists in the SIP channel driver if an
ACK with SDP is received after the channel has been terminated. The handling
code incorrectly assumed that the channel would always be present.
This patch adds a check such that the SDP will only be parsed and applied if
Asterisk has a channel present that is associated with the dialog.
Note that the patch being applied was modified only slightly from the patch
provided by Walter Doekes of OSSO B.V.
(closes issue ASTERISK-21064)
Reported by: Colin Cuthbertson
Tested by: wdoekes, Colin Cutherbertson
patches:
issueA21064_fix.patch uploaded by wdoekes (License 5674)
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