Mark Michelson [Mon, 21 Oct 2013 21:05:37 +0000 (21:05 +0000)]
Remove a noisy debug message from bridging code.
This particular debug message, during a stress test, was logged so
often that it appeared that there may be a memory leak in the logger
code. In actuality, there was no memory leak, but the logger thread
was having a hard time keeping up with the demands of the rest of the
system.
Since this debug message has no value at all, the best way to fix the
problem was to just remove the message.
(closes issue AST-1225)
reported by John Bigelow
Patches:
spammy_log.diff uploaded by Mark Michelson (License #5049)
Kinsey Moore [Sat, 19 Oct 2013 21:53:08 +0000 (21:53 +0000)]
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.
Joshua Colp [Sat, 19 Oct 2013 14:43:41 +0000 (14:43 +0000)]
Return a channel snapshot when originating using ARI, and subscribe the Stasis application to it.
This change allows a user of ARI to know what channel it has originated and also follow any
progress. If a Stasis application is provided it will be automatically subscribed to the
originated channel immediately.
(closes issue ASTERISK-22485)
Reported by: David Lee
David M. Lee [Fri, 18 Oct 2013 21:49:48 +0000 (21:49 +0000)]
This is just a quick script for dumping swagger-ui into static-http,
so that it can be served by the Asterisk web server.
I had to change the Makefile in order to recursively install content
from the static-http directory, hence the code review instead of just
putting it in.
This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
Richard Mudgett [Fri, 18 Oct 2013 16:52:32 +0000 (16:52 +0000)]
Add channel lock protection around translation path setup.
Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge. With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.
* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.
* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper(). The call to
ast_translator_best_choice() got them backwards.
* Updated some callers of ast_channel_make_compatible() and the function
documentation. There is actually a difference between the two channels
passed in.
* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible(). The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.
(closes issue ASTERISK-22542)
Reported by: Matt Jordan
Michael L. Young [Fri, 18 Oct 2013 15:13:02 +0000 (15:13 +0000)]
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)
Walter Doekes [Fri, 18 Oct 2013 14:47:12 +0000 (14:47 +0000)]
Properly copy/remove the device state cache flag over a masquerade.
In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells
the devstate system to not cache states for non-real devices. However,
when optimizing away channels (ast_do_masquerade), that flag wasn't
copied.
In my case, using Local devices as queue members created a situation
where the endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The endpoint
channel was optimized into the (previously) Local channel, but kept
the do-not-cache flag. The end result being that the queue member
apparently stayed in use forever.
(closes issue ASTERISK-22718)
Reported by: Walter Doekes
Michael L. Young [Thu, 17 Oct 2013 20:37:10 +0000 (20:37 +0000)]
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)
Kinsey Moore [Wed, 16 Oct 2013 14:01:04 +0000 (14:01 +0000)]
Clarify documentation for channel and bridge list
This makes it clear that the ARI API calls for listing channels and
bridges will list all channels or bridges in the system and not just
those that are in or are controlled by a Stasis application.
(closes issue ASTERISK-22635)
Reported by: Kevin Harwell
Walter Doekes [Wed, 16 Oct 2013 12:12:42 +0000 (12:12 +0000)]
Don't check all realtime queues when doing "queue show some_queue".
When using realtime queues, queues have to be fetched from the database
every now and then to see if any info has been changed or to see if the
queue has been removed. When fetching info for an individual queue, the
pruning of other queues is unnecessarily costly.
Kinsey Moore [Tue, 15 Oct 2013 20:02:08 +0000 (20:02 +0000)]
Ensure bridge record error responses validate
This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.
Mark Michelson [Tue, 15 Oct 2013 15:21:56 +0000 (15:21 +0000)]
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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David M. Lee [Tue, 15 Oct 2013 13:43:05 +0000 (13:43 +0000)]
My doc correction in r400842 had a silly bug.
Because I added a wiki_description to models and not their properties, the
rendered wiki page had the model description instead of the property
descriptions, which looks very silly indeed.
Mark Michelson [Mon, 14 Oct 2013 21:52:24 +0000 (21:52 +0000)]
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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Kevin Harwell [Mon, 14 Oct 2013 15:52:28 +0000 (15:52 +0000)]
pjsip outbound registration: Log message says received a 408 when we didn't
If the server didn't exist that we are trying to register to the log message
would say that a 408 was received from that server when in reality one wasn't.
Added log messages stating no response was received if the response does not
exist.
(closes issue ASTERISK-22554)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2893/
Joshua Colp [Sun, 13 Oct 2013 15:41:37 +0000 (15:41 +0000)]
Fix a race condition in res_pjsip_session with rapidly terminating the session.
The INVITE session state callback wrongly assumes that a session will always exist, but
when rapidly terminating the session this assumption goes out the window. As all handler
code for the INVITE session state callback requires the session it will now just exit
immediately if no session exists.
(closes issue ASTERISK-22668)
Reported by: John Bigelow
Kinsey Moore [Sat, 12 Oct 2013 16:49:00 +0000 (16:49 +0000)]
Fix realm comparison for outbound auth
When generating the list of authentication credentials to pass to
PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not
always NULL-terminated. This sometimes resulted in incorrect text for
the realm and a failure to match the realm for authentication purposes
which was causing the outbound nominal auth pjsip basic call test to
bounce. This now uses the pj_str_t that contains the realm instead of
generating a new one. Thanks to John Bigelow for helping to narrow this
down.
Richard Mudgett [Fri, 11 Oct 2013 16:25:05 +0000 (16:25 +0000)]
Softmix: Fix crash when switching from softmix to another bridge technology.
The crash is caused by a race condition when switching between native RTP
and softmix bridging technologies. In this situation, the bridging
technology is switched from native RTP to softmix, and then back to native
RTP fast enough that the softmix private data gets destroyed before the
softmix mixing thread gets started.
Thanks to Kinsey Moore for the crash analysis.
* Fix race condition when starting the softmix mixing thread and switching
to another bridge technology.
(closes issue ASTERISK-22678)
Reported by: John Bigelow
Patches:
jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett
Tested by: John Bigelow
Joshua Colp [Thu, 10 Oct 2013 12:25:44 +0000 (12:25 +0000)]
Fix an assertion in res_pjsip when specifying an invalid outbound proxy.
This change fixes two issues when setting an outbound proxy:
1. The outbound proxy URI was not parsed and validated during configuration.
2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would
occur because the usage count on the dialog was not decremented.
The documentation has also been updated to specify that a full URI must be specified for
the outbound proxy.
(closes issue ASTERISK-22672)
Reported by: Antti Yrjola
Matthew Jordan [Wed, 9 Oct 2013 11:00:47 +0000 (11:00 +0000)]
Use 'z' as the format specifier for size_t
Using 'lu' will produce a compiler warning for some versions of gcc and on some
architectures. 'z' should be portable as a format specifier for size_t.
Matthew Jordan [Tue, 8 Oct 2013 22:58:04 +0000 (22:58 +0000)]
Add PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stack
This patch adds support to the PJSIP stack in Asterisk for SIP header
manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader.
For PJSIP_HEADER, an incoming supplemental session callback is registered that
takes the pjsip_hdrs from the incoming session and stores them in a linked
list in the session datastore. Calls to PJSIP_HEADER traverse over the list
and return the nth matching header where 'n' is the 'number' argument to the
function.
When adding a header, the first call creates a datastore and linked list and
adds the datastore to the session. The header is then created as a pjsip_hdr
and added to the list. An outgoing supplemental session callback then
traverses the list and adds the headers to the outgoing pjsip_msg.
When removing a header, the list created with PJSIP_HEADER(add,...) is
traversed and all matching entries are removed.
(closes issue ASTERISK-22498)
Reported by: George Joseph
patch:
res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322)
Kinsey Moore [Tue, 8 Oct 2013 22:30:37 +0000 (22:30 +0000)]
Add warning when compiling with iODBC support
When running configure, libiodbc2 development headers will fulfill the
requirement for ODBC development headers, but will not function
properly. This adds a warning when libiodbc2 development headers are
detected instead of unixodbc development headers.
(closes issue ASTERISK-22459)
Reported by: Patrick Maille
Tested by: Walter Doekes
Patches:
issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674)
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Mark Michelson [Tue, 8 Oct 2013 20:51:21 +0000 (20:51 +0000)]
Switch from using pjsip_strerror to pj_strerror.
pjsip_strerror is only aware of PJSIP-specific error
codes. pj_strerror() is aware of all PJProject error
codes and OS-specific error codes.
This specifically fixes an oft-seen error in transport
configuration code where EADDRINUSE would result in
"Unknown PJSIP error 120098" instead of a useful
message.
Richard Mudgett [Tue, 8 Oct 2013 20:16:55 +0000 (20:16 +0000)]
app_confbridge: Can now set the language used for announcements to the conference.
ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
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* Fixed looking in the wrong profiles container to see if the default_user
profile is already created in verify_default_profiles(). The bridge
profile container is never going to hold user profiles. :)
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Kinsey Moore [Tue, 8 Oct 2013 18:19:23 +0000 (18:19 +0000)]
Fix func_config list entry allocation
The AST_CONFIG dialplan function defined in func_config.c allocates its
config file list entries using ast_malloc. List entry allocations
destined for use with Asterisk's linked list API must be ast_calloc()d
or otherwise initialized so that list pointers are set to NULL. These
uses of ast_malloc have been replaced by ast_calloc to prevent
dereferencing of uninitialized pointer values when traversing the list.
(closes issue ASTERISK-22483)
Reported by: Brian Scott
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Mark Michelson [Tue, 8 Oct 2013 15:36:08 +0000 (15:36 +0000)]
Push CLI qualify into the threadpool.
If you run Asterisk in the background and then connect to
it through a separate console, the thread that runs CLI commands
is not registered with PJLIB. Thus PJLIB does not like it when
you attempt to send OPTIONS requests from that thread. So now
we push the task into the threadpool, which we know to be registered
with PJLIB.
Richard Mudgett [Tue, 8 Oct 2013 15:11:04 +0000 (15:11 +0000)]
Make app_queue and res_agi independent of AMI being enabled.
The https://reviewboard.asterisk.org/r/2888/ review changes manager to not
subscribe to stasis when it is disabled for performance reasons. When
manager is disabled app_queue and res_agi decline to load and fail to
clean up what they have already allocated.
* Made app_queue and res_agi clean up allocated resources when they
decline to load.
* Made app_queue and res_agi use their own subscriptions to the stasis
topics instead of borrowing manager's message router structure
inappropriately.
app_queue: Fix Queuelog EXITWITHKEY only logging two of four fields
Commit r62462 added two extra fields for logging "the original position the
caller entered the queue at, and the amount of time the caller was waiting in
the queue." But when r75969 was merged from 1.4 into trunk (r75977), these two
fields disappeared. Those two extra fields were not logged in 1.4 and when the
patch was merged, those fields went away.
Therefore, this is a regression and was caught by the reporter because he was
reading the awesome "Asterisk: The Definitive Guide" book.
(closes issue ASTERISK-22197)
Reported by: Dalius M.
Tested by: Dalius M.
Patches:
asterisk-22197-q-log-exitwithkey.diff
uploaded by Michael L. Young (license 5026)
Jonathan Rose [Fri, 4 Oct 2013 18:42:06 +0000 (18:42 +0000)]
chan_pjsip: Make logger togglable without loading/unloading
This patch makes the res_pjsip_logger do a few things... First, it
will be built and installed by default now, so end users won't need
to enable it in menuselect. Second, while it is loaded, it no longer
will immediately issue log messages. Upon loading, it is in the
disabled state and must be turned on with the new CLI command. The
CLI command 'pjsip set logger <on/off/host> has been added and can be
used to do the following:
pjsip set logger on:
Enables logger for all PJSIP traffic
pjsip set logger off:
Disables logger for all PJSIP traffic
pjsip set logger host <host>:
Enables logger for the specific host
Matthew Jordan [Fri, 4 Oct 2013 15:54:57 +0000 (15:54 +0000)]
ARI: Add subscription support
This patch adds an /applications API to ARI, allowing explicit management of
Stasis applications.
* GET /applications - list current applications
* GET /applications/{applicationName} - get details of a specific application
* POST /applications/{applicationName}/subscription - explicitly subscribe to
a channel, bridge or endpoint
* DELETE /applications/{applicationName}/subscription - explicitly unsubscribe
from a channel, bridge or endpoint
Subscriptions work by a reference counting mechanism: if you subscript to an
event source X number of times, you must unsubscribe X number of times to stop
receiveing events for that event source.
Matthew Jordan [Thu, 3 Oct 2013 21:40:20 +0000 (21:40 +0000)]
Remove publication of a channel snapshot when the technology is set
This patch removes said publication for a few reasons:
(1) It is unnecessary. Association of the channel technology with a specific
channel is an implementation detail that should be assumed to "just happen",
and consumers of Stasis don't need to be informed about it.
(2) Publication of said message can now cause crashes, as the actual creation
of a channel in normal locations now stages its messages. As a result, things
that create dummy channels (such as the SIP RTP QOS unit test) and associate
them with a channel technology were now crashing, as the channel itself was
not known by Stasis.
Kinsey Moore [Thu, 3 Oct 2013 19:29:49 +0000 (19:29 +0000)]
Fix security events for AMI invalid password
In r337595, additional security events were added for chan_sip
authentication failures. The new IEs added to the existing invalid
password event were defined as required IEs, but existing users of the
event did not set the new IEs and could not since they didn't apply to
existing uses. They are now marked as optional IEs.
(closes issue ASTERISK-22578)
Reported by: Matt Jordan
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Mark Michelson [Thu, 3 Oct 2013 19:11:22 +0000 (19:11 +0000)]
Fix assumption in bridge_native_rtp.c regarding number of participants in a bridge.
When a party leaves a bridge, there may be more participants in the bridge than expected.
As such, it is important not to make assumptions regarding the list of channels in a
bridge.
This change makes it so that when a party leaves a native RTP bridge, we unbridge it and
the party it was bridged with. Previously, the first and last channels in the list were
unbridged since it was assumed that these were the two channels that had been bridged. As
previously stated, a new party had been inserted into the bridge, so this logic did not
work properly.
(closes issue ASTERISK-22615)
reported by Matt Jordan
(closes issue ASTERISK-22532)
reported by Matt Jordan
Kinsey Moore [Thu, 3 Oct 2013 18:28:57 +0000 (18:28 +0000)]
res_rtp_multicast: Ensure SSRC is set properly
This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.
Kinsey Moore [Thu, 3 Oct 2013 17:57:49 +0000 (17:57 +0000)]
Detect and use xsltCleanupGlobals when available
This introduces usage of an additional libxslt cleanup function,
xsltCleanupGlobals, when the configure script detects that it is
available. Early versions of the library did not include this function.
Matthew Jordan [Thu, 3 Oct 2013 17:55:50 +0000 (17:55 +0000)]
Update Alembic database scripts for external scripting and PostgreSQL, Oracle
This patch does the following:
1) The env scripts have been updated to be tolerant of a NULL configuration
file. This occurs when configuration is provided by an external script,
such that the actual config.ini file is not used.
2) Enum types have all been given names. This is needed for PostgreSQL script
generation.
3) The identifier meetme_confno_starttime_endtime is greater than 30
characters, and hence invalid for Oracle databases. This has been truncated
down to meetme_confno_start_end.
Joshua Colp [Thu, 3 Oct 2013 14:51:41 +0000 (14:51 +0000)]
Fix crashes in res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and external_media_address is set.
The callback function for changing the media address in streams wrongly assumes that a connection line
will always be present. This is false as no line is present if a stream has been rejected.
(closes issue ASTERISK-22645)
Reported by: Rusty Newton
Mark Michelson [Wed, 2 Oct 2013 22:34:05 +0000 (22:34 +0000)]
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.
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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Matthew Jordan [Wed, 2 Oct 2013 21:25:50 +0000 (21:25 +0000)]
Only create Stasis subscriptions when enabled
Subscribing to Stasis isn't free.
As such, this patch makes AMI, CDR, and CEL - the "big 3" - only subscribe
when enabled. Toggling their availability via a .conf file will
unsubscribe/subscribe as appropriate.
Matthew Jordan [Wed, 2 Oct 2013 18:57:15 +0000 (18:57 +0000)]
Fix the CDR CLI command 'cdr show active {channel}'
When the switch from channel names to channel unique IDs happened, the poor
CLI command got left in the dust. This fixes the command so that users can
once again see how Asterisk is messing up your billing information.
Richard Mudgett [Wed, 2 Oct 2013 17:11:42 +0000 (17:11 +0000)]
MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is enabled.
* There were several places in ARI where an external library was mallocing
memory that must always be released with free(). When MALLOC_DEBUG is
enabled, free() is redirected to the MALLOC_DEBUG version. Since the
external library call still uses the normal malloc(), MALLOC_DEBUG
complains that the freed memory block is not registered and will not free
it. These cases must use ast_std_free().
* Changed calls to asprintf() and vasprintf() to the equivalent
ast_asprintf() and ast_vasprintf() versions respectively.
Joshua Colp [Wed, 2 Oct 2013 16:20:25 +0000 (16:20 +0000)]
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.
Joshua Colp [Wed, 2 Oct 2013 15:31:31 +0000 (15:31 +0000)]
Fix a random one way audio issue in PJSIP.
Due to the asynchronous design of the PJMEDIA SDP negotiator it was possible for
the SDP to be negotiated *after* a channel was created and after it was being wait
on by an application. It is only after negotiation occurs that the file descriptors
for RTP are placed on the channel. Since the channel was already being waited on
these file descriptors were not monitored, causing incoming media to never be read.
This change wakes up any application waiting on the channel so that added file
descriptors end up being monitored.
Matthew Jordan [Mon, 30 Sep 2013 19:57:36 +0000 (19:57 +0000)]
Parse arguments passed to the CDR_PROP function correctly
I can only blame this on a bad merge, because this in no way worked properly
the way it was written. Mea culpa. The function should now parse its arguments
correctly and function properly. (Note that the API used by the CDR_PROP
function has working unit tests... this was merely bad coding of the actual
registered function)
(closes issue ASTERISK-22613)
Reported by: Private Name
projects / thirdparty / asterisk.git / log
