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This patch was intended to eliminate a deadlock that occurs when
masquerades occur in pjsip channels, but has some potential side
effects. Mark Michelson is currently working on addressing this
problem from another angle.
(issue ASTERISK-22936)
Reported by: Jonathan Rose
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If you set a peer's outboundproxy and then removed it from the config,
this would not get picked up in a config reload. This patch fixes that
by resetting it in set_peer_defaults().
Closes ASTERISK-19454
Review: https://reviewboard.asterisk.org/r/3065/
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This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
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(closes issue ASTERISK-22936)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3042/
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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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Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.
(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
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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 the structure passed to ast_get_ip should be set memset to 0, thus
initializing the ss_family member to 0, explicitly setting it to AST_AF_UNSPEC
is more portable.
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This started off as a fix for the failing IAX2 acl_call test in the Asterisk
Test Suite. When inspecting why that test was failing, it became clear that all
attempts to bind to any local loopback address was failing:
[Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding IAX2 to address
127.0.0.1:4569
[Nov 2 15:56:28] DEBUG[15787] netsock2.c: Splitting '127.0.0.1' into...
[Nov 2 15:56:28] DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''.
[Nov 2 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1", "(null)",
...): ai_family not supported
[Nov 2 15:56:28] WARNING[15787] acl.c: Unable to lookup '127.0.0.1'
While there's conceivably other ways for getaddrino to return EAI_FAMILY, the
most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not provided as the
desired family. The culprit was the call to ast_get_ip, defined in acl.h. This
function uses the family from the passed in addr object (which it will also
populate when it returns!) when it eventually calls getaddrinfo.
This patch fixes the use of ast_get_ip that were not specifying the family in
chan_iax2. This prevents uninitialized use of the structure, so that the
addresses resolve correctly.
Review: https://reviewboard.asterisk.org/r/2991
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The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring. Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow. If they are restricted then "anonymous" is used instead.
(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
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This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.
(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
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(issue ASTERISK-22777)
Reported by: Matt Jordan
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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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after the channel is hung up.
(closes issue ASTERISK-22731)
Reported by: Kinsey Moore
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Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.
(closes issue ASTERISK-22005)
Reported by: Torrey Searle
Patches:
optional_avpf_trunk.patch uploaded by tsearle (license 5334)
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Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
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(issue ASTERSIK-22467)
Reported by: Corey Farrell
Patches:
chan_sip-parse_contact_header_test-free-contacts.patch uploaded by coreyfarrell (license 5909)
cli-filename-completion-leak.patch uploaded by coreyfarrell (license 5909)
func_math.patch uploaded by corefarrell (license 5909)
main-test-cleanup.patch uploaded by coreyfarrell (license 5909)
test_dlinklists.patch uploaded by coreyfarrell (license 5909)
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This corrects a situation in which a media line was not parsed properly
and resulted in a crash.
(closes issue ASTERISK-21190)
Reported by: adomjan
Patches:
chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
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If preferred codecs included any non-audio format the code would
mistakenly add the audio format, even if it was not a joint capability
with the remote side.
(closes issue ASTERISK-21131)
Reported by: nbougues
Patches:
patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470)
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When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses
was removed by mistake. This patch restores this functionality and adds notes
about IPv6 addresses in the sample config.
(closes issue ASTERISK-22741)
Reported by: Joshua Colp
Tested by: Michael L. Young
Patches:
asterisk-22741-fix-binding-multiple-addr.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2945/
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This fixes address lookup for incoming calls without a peer definition.
The address family was unset instead of being set to AST_AF_UNSPEC
which was causing lookup failures on "127.0.0.1". This is one of the
causes of the current failure of the app_page integration test.
Review: https://reviewboard.asterisk.org/r/2933/
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When trying to determine if a peer is behind NAT, we should not be using the
ports when comparing addresses.
This patch removes the port from being checked and just useds the addresses
now.
(closes issue ASTERISK-22729)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-remove-using-port-for-nat-check.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2927/
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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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channels compatible.
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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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* Add hwtxgain and hwrxgain config options to chan_dahdi.conf with
documentation in chan_dahdi.conf.sample.
(closes issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett
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output.
* Remember the swgain setting from CLI "dahdi set swgain" command so the
CLI "dahdi show channel" output will reflect the current setting.
* Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
(issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
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Bumping the SDP version number can cause interoperability problems
since receivers of the responses will expect that a 200 SDP will
be identical to a previous 183 SDP.
(closes issue ASTERISK-21204)
reported by NITESH BANSAL
Patches:
dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418)
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This patch adds IPv6 support to chan_iax2. Yay!
(closes issue ASTERISK-22025)
Patches:
iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2660/
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(closes issue ASTERISK-22574)
Reported by: Filip Jenicek
Patches:
chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
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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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r400318 | mmichelson | 2013-10-02 17:08:49 -0500 (Wed, 02 Oct 2013) | 12 lines
Remove unnecessary waits from stasis.
Since caches are updated on publisher threads, there is no need
to wait for the cache updates to occur after a stasis message
is published.
In the case of chan_pjsip device state changes, this set of
changes caused an improvement to performance.
Review: https://reviewboard.asterisk.org/r/2890
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r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, 02 Oct 2013) | 3 lines
Remove svn:mergeinfo property.
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The member reg in the peercnt structure is an unsigned char and peercnt_modify()
is expecting an unsigned char argument which gets assigned to peercnt->reg.
This patch fixes that by casting the integer argument being passed to
peercnt_modify to unsigned char.
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This change introduces the ability to stage channel snapshot
creation and publishing by suppressing the implicit creation
and publishing that some functions have. Once all operations
are executed the staging is marked as done and a single snapshot
is created and published.
Review: https://reviewboard.asterisk.org/r/2889/
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Transferring an analog call using a flash-hook to parking would fail to
park the call and result in an invalid ao2 object unref.
* Park the correct bridged channel.
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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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This adds a global option in chan_sip to allow it to continue
attempting registration if a 403 is received, clearing the cached nonce
and treating it as a non-fatal response. Normally, this would cause
registration attempts to that endpoint to stop.
This also adds a similar per-outbound-registration option to chan_pjsip
which allows the retry interval to be altered for 403 responses to
REGISTER requests.
(closes issue ASTERISK-17138)
Review: https://reviewboard.asterisk.org/r/2874/
Reported by: Rudi
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* Eliminated an unnecessary initialization in check_user_full().
(closes issue ASTERISK-22477)
Reported by: Michael Shepelev
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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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The PRI and SS7 link control threads are not stopped correctly when the
chan_dahdi.so module is unloaded. The link control threads pri_dchannel()
and ss7_linkset() are not awakened from a poll() to cancel the thread.
* Added a SIGURG signal after requesting the thread cancel to break the
link control thread poll() immediately.
For SS7 it was slightly worse, the link poll() timeout would always be
whatever was the last libss7 scheduled event time used. If no libss7
scheduled event was pending, the thread could run more often than
necessary.
* Set nextms to 60 seconds for the ss7_linkset() poll() if there is no
other libss7 scheduled event.
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Header In 200ok
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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* Clean up some twisted code in the iax2_bridge() loop.
* Add AST_CONTROL_VIDUPDATE and AST_CONTROL_SRCCHANGE to a list of frames
to prevent the native bridge loop from breaking.
* Passing the AST_CONTROL_T38_PARAMETERS frame should also allow FAX over
a native IAX2 bridge.
(issue ABE-2912)
Review: https://reviewboard.asterisk.org/r/2870/
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For v12 and above this is really just documentation until IAX2 native
bridging is restored.
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(closes issue ASTERISK-22453)
Reported by: Corey Farrell
Patches:
chan_pjsip_session_unregister_supplement.patch uploaded by Corey Farrell (license 5909)
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Prior to this patch, Asterisk would incorrectly use the previous endpoint
addresses in SDP in spite of providing its own port. T38 is never meant to
be done through directmedia and Asterisk should always be in the media path
for these streams.
(closes issue ASTERISK-17273)
Reported by: Kevin Stewart
(closes issue ASTERISK-18706)
Reported by: Jeremy Kister
Review: https://reviewboard.asterisk.org/r/2853/
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