Commit Graph

8139 Commits

Author SHA1 Message Date
Kevin Harwell
28c0cb28d0 channel locking: Add locking for channel snapshot creation
Original commit message by mmichelson (asterisk 12 r403311):

"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."

The above was initially committed and then reverted at r403398.  The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed.  Fixed by unreffing the channels.

Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel.  Fixed by
unlocking "other->chan"

(closes issue ASTERISK-22709)
Reported by: John Bigelow
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Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-18 20:33:37 +00:00
Joshua Colp
e2630fcd51 channels: Return allocated channels locked.
This change makes ast_channel_alloc return allocated channels
locked. By doing so no other thread can acquire, lock, and manipulate
the channel before it is completely set up.

(closes issue AST-1256)

Review: https://reviewboard.asterisk.org/r/3067/
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2013-12-18 19:28:05 +00:00
Rusty Newton
f7c60b8fb6 Several components: fixing Typos in comments and code, "avaliable" instead of "available"
(issue ASTERISK-23021)
(closes issue ASTERISK-23021)
Reported by: Jeremy Lainé
Tested by: Rusty Newton
Patches:
   available.patch uploaded by Jeremy Lainé (license 6561)
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2013-12-17 23:38:02 +00:00
Matthew Jordan
4ddf45fd24 pjsip/dialplan_functions: Use the right buffer length when printing URIs
While entertaining, sizeof(buflen) is not the same as buflen. Doh.
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2013-12-15 01:39:20 +00:00
Kevin Harwell
84e1790beb bridge_native_rtp: Deadlock during 4-way conference creation
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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2013-12-13 18:33:25 +00:00
Matthew Jordan
fc8c0ef28f channels/Makefile: clean pjsip directory
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2013-12-13 05:00:53 +00:00
Jonathan Rose
5583d2c629 chan_pjsip: Revert r403587
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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2013-12-12 19:18:06 +00:00
Russell Bryant
90108b15a0 Reset peer outboundproxy on sip.conf reload
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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2013-12-11 19:22:05 +00:00
Matthew Jordan
ce423d2ea4 func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11 13:06:30 +00:00
Jonathan Rose
905a7de52b chan_pjsip: Fix a sticking channel lock caused by channel masquerades
(closes issue ASTERISK-22936)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3042/
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2013-12-09 22:59:14 +00:00
David M. Lee
1212906351 Reverting r403311. It's causing ARI tests to hang.
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2013-12-05 22:10:20 +00:00
Mark Michelson
8e8b329e14 Add channel locking for channel snapshot creation.
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03 17:07:29 +00:00
Kinsey Moore
95bc7626aa chan_dahdi: Fix crash during caller ID read
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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2013-11-12 15:03:18 +00:00
Scott Griepentrog
094db82a73 chan_sip: keep same local (from) tag for outgoing register requests
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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2013-11-08 23:07:50 +00:00
Matthew Jordan
029ce1e962 chan_sip: Use AST_AF* defined constant when calling ast_get_ip
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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2013-11-05 21:17:30 +00:00
Matthew Jordan
f6bd22b0fd chan_iax2: Fix incorrect usage of ast_get_ip involving uninitialized struct
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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2013-11-05 21:14:42 +00:00
Kevin Harwell
fe47684b43 chan_sip: notify dialog info ignores presentation indicator in callerid
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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2013-11-04 21:02:18 +00:00
Kinsey Moore
98dea21bc1 chan_sip: Fix RTCP port for SRFLX ICE candidates
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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2013-11-01 12:40:40 +00:00
Kevin Harwell
4746c068dc pjsip_messaging: Added debug for in dialog messaging
(issue ASTERISK-22777)
Reported by: Matt Jordan
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2013-10-30 17:54:26 +00:00
Michael L. Young
230141d677 chan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers"
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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2013-10-28 14:59:16 +00:00
Joshua Colp
26914adc00 chan_pjsip: Fix a crash when direct media is enabled and an ACK is received after the channel is hung up.
(closes issue ASTERISK-22731)
Reported by: Kinsey Moore
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2013-10-26 12:56:08 +00:00
Kevin Harwell
9ba7742431 chan_sip: Allow a sip peer to accept both AVP and AVPF calls
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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2013-10-25 16:09:05 +00:00
Jonathan Rose
4ca0f222e8 memory leaks: Memory leak cleanup patch by Corey Farrell (second set)
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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2013-10-24 17:00:27 +00:00
Jonathan Rose
beb5cdbef5 memory leaks: Memory leak cleanup patch by Corey Farrell (first set)
(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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2013-10-23 20:10:30 +00:00
Kinsey Moore
dfd3f4ef46 chan_mgcp: Properly handle malformed media lines
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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2013-10-23 15:23:58 +00:00
Joshua Colp
f4e028a765 chan_sip: Fix an issue where an incompatible audio format may be added to SDP.
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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2013-10-23 11:16:44 +00:00
Michael L. Young
4ca92e3b8a chan_iax2: Fix Binding To Multiple Addresses Again
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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2013-10-23 02:36:01 +00:00
Kinsey Moore
3407f27c88 Fix IAX2 incoming call address lookups
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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2013-10-19 21:57:07 +00:00
Michael L. Young
32d758ed32 Remove Port Restriction When Checking For NAT
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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2013-10-18 15:14:36 +00:00
Michael L. Young
42f3cae1fd Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag
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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2013-10-17 20:39:29 +00:00
Richard Mudgett
66e7dc041d bridge_native_dahdi: Return channel join failure if could not make the channels compatible.
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2013-10-15 20:26:13 +00:00
Richard Mudgett
3208058831 chan_iax2: Fix channel left locked in off nominal code path.
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2013-10-15 20:05:47 +00:00
Mark Michelson
2c927b871f Prevent chan_sip from sending duplicate BYEs.
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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2013-10-15 15:26:06 +00:00
Richard Mudgett
2127848d6c chan_dahdi: Add config support for hwgain settings.
* 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14 22:52:42 +00:00
Richard Mudgett
edb437ee6c chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" 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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2013-10-14 22:06:01 +00:00
Mark Michelson
47e910bfe6 chan_sip: Do not increment the SDP version between 183 and 200 responses.
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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2013-10-14 22:03:22 +00:00
Richard Mudgett
e2a0a6ed19 chan_iax2: Fix compile error.
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2013-10-05 00:59:17 +00:00
Michael L. Young
2af53640c8 Add IPv6 Support To chan_iax2
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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2013-10-04 21:41:58 +00:00
Jonathan Rose
d35b5c9cb0 chan_sip: Don't ignore expires value in contact header if it lacks semicolon
(closes issue ASTERISK-22574)
Reported by: Filip Jenicek
Patches:
    chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
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2013-10-03 23:20:00 +00:00
Richard Mudgett
e3d69a5628 chan_vpb: Make compile again.
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2013-10-03 16:28:35 +00:00
Mark Michelson
ee21eee7e0 Cache string values of formats on ast_format_cap() to save processing.
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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2013-10-03 14:58:16 +00:00
Mark Michelson
addbf276f5 Multiple revisions 400318-400319
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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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2013-10-02 22:22:17 +00:00
Michael L. Young
e4ed9886e6 Cast Integer Argument To Unsigned Char
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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2013-10-02 21:33:42 +00:00
Richard Mudgett
d14869bcad sig_ss7: Fix compiler warnings.
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2013-10-02 17:10:39 +00:00
Joshua Colp
c1235f2639 Reduce channel snapshot creation and publishing by up to 50%.
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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2013-10-02 16:23:34 +00:00
Richard Mudgett
7d17d5fb04 chan_dahdi: Fix analog parking using flash-hook.
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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2013-10-01 21:19:13 +00:00
David M. Lee
2de42c2a25 Multiple revisions 399887,400138,400178,400180-400181
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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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2013-09-30 18:55:27 +00:00
Kinsey Moore
b44ce141e5 chan_sip: Allow Asterisk to retry after 403 on register
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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2013-09-30 15:57:11 +00:00
Richard Mudgett
9f19d096e3 chan_sip: Increase some scratch buffer sizes dealing with caller id.
* Eliminated an unnecessary initialization in check_user_full().

(closes issue ASTERISK-22477)
Reported by: Michael Shepelev
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2013-09-27 21:44:42 +00:00
Jonathan Rose
7e2a72771d chan_sip: Reject calls on 200 OKs if no SDP has been received
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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2013-09-27 17:46:16 +00:00