Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.
This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.
(issue ASTERISK-19192)
Review: https://reviewboard.asterisk.org/r/1681/
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Add an AMI event in the Call category that is issued when a call is terminated
due to either RTP stream inactivity or SIP session timer expiration.
Event description:
Event: SessionTimeout
Source: source
Channel: channel-name
Uniqueid: channel-unique-id
`source` can be either RTPTimeout or SIPSessionTimer
(closes issue ASTERISK-16467)
Patch-by: Kirill Katsnelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix corner cases in get_calleridname() parsing and ensure that the
output buffer is nul terminated.
* Make get_calleridname() truncate the name it parses if the given buffer
is too small rather than abandoning the parse and not returning anything
for the name. Adjusted get_calleridname_test() unit test to handle the
truncation change.
* Fix get_in_brackets_test() unit test to check the results of
get_in_brackets() correctly.
* Fix parse_name_andor_addr() to not return the address of a local buffer.
This function is currently not used.
* Fix potential NULL pointer dereference in sip_sendtext().
* No need to memset(calleridname) in check_user_full() or tmp_name in
get_name_and_number() because get_calleridname() ensures that it is nul
terminated.
* Reply with an accurate response if get_msg_text() fails in
receive_message(). This is academic in v1.8 because get_msg_text() can
never fail.
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If a Contact or a Record-Route header had a quoted string with an
item in angle brackets, then we would mis-parse it. For instance,
"Bob <1234>" <1234@example.org>
would be misparsed as having the URI "1234"
The fix for this is to use parsing functions from reqresp_parser.h
since they are heavily tested and are awesome.
(issue ASTERISK-18990)
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r351183 | russell | 2012-01-16 20:43:19 -0500 (Mon, 16 Jan 2012) | 29 lines
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r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines
Add some missing locking in chan_sip.
This patch adds some missing locking to the function
send_provisional_keepalive_full(). This function is called from the scheduler,
which is processed in the SIP monitor thread. The associated channel (or pbx)
thread will also be using the same sip_pvt and ast_channel so locking must be
used. The sip_pvt_lock_full() function is used to ensure proper locking order
in a safe manner.
In passing, document a suspected reference counting error in this function.
The "fix" is left commented out because when the "fix" is present, crashes
occur. My theory is that fixing it is exposing a reference counting error
elsewhere, but I don't know where. (Or my analysis of this being a problem
could have been completely wrong in the first place). Leave the comment in
the code for so that someone may investigate it again in the future.
Also add a bit of doxygen to transmit_provisional_response().
(closes issue ASTERISK-18979)
Review: https://reviewboard.asterisk.org/r/1648
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When handling a non-2xx final response on an INVITE transaction, we have to
keep the transaction around after we send an ACK in case we receive a
retransmission of the response so we can re-transmit the ACK, but also tear
down the ast_channel as soon as we transmit the ACK. Before this patch, we
could fail at both of these things. Calling sip_alreadygone/needdestroy
prevented us from keeping the transaction up and retransmitting the ACK, and
queueing CONGESTION was not sufficient to cause the channel to be torn down
when originating calls via the CLI, for example.
This patch queues a hangup with CONGESTION instead of just queueing CONGESTION
for these responses and removes the sip_alreadygone and sip_needdestroy calls
from handle_response_invite on non-2xx responses. It relies on the hangup
calling sip_scheddestroy.
For more information, see section 17.1.1.1 of RFC 3261.
(closes issue ASTERISK-17717)
Review: https://reviewboard.asterisk.org/r/1672/
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When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry.
(closes issue ASTERISK-18996)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches: session_timer_fix.diff by Terry Wilson (License #5357)
based on session_timer.patch by Thomas Arimont (License #5525)
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Prior to this patch, the udptl struct was allocated and initialized when a
dialog was associated with a peer that supported T.38, when a new SIP
channel was allocated, or what an INVITE request was received. This resulted
in any dialog associated with a peer that supported T.38 having udptl support
assigned to it, including the UDP ports needed for communication. This
occurred even in non-INVITE dialogs that would never send image media.
This patch creates and initializes the udptl structure only when the SDP
for a dialog specifies that image media is supported, or when Asterisk
indicates through the appropriate control frame that a dialog is to support
T.38.
(closes issue ASTERISK-16698)
Reported by: under
Tested by: Stefan Schmidt
Patches: udptl_20120113.diff uploaded by mjordan (License #6283)
(closes issue ASTERISK-16794)
Reported by: Elazar Broad
Tested by: Stefan Schmidt
review: https://reviewboard.asterisk.org/r/1668/
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r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines
Ensure that two prerequisites are properly installed on Debian-style distributions.
* Don't specify a specific version of libgmime; newer versions are available
now and acceptable.
* Install libsrtp so that res_srtp can be built.
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r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines
Correct some 'set-but-not-used' variable warnings.
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This patch addresses two issues in ConfBridge and the channel bridge layer:
1. It fixes a race condition wherein the bridge channel could be hung up
2. It removes the deadlock avoidance from the bridging layer and makes the
bridge_pvt an ao2 ref counted object
Patch by David Vossel (mjordan was merely the commit monkey)
(issue ASTERISK-18988)
(closes issue ASTERISK-18885)
Reported by: Dmitry Melekhov
Tested by: Matt Jordan
Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628)
(closes issue ASTERISK-19100)
Reported by: Matt Jordan
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1654/
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There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The iax2_process_thread() can exit without anyone waiting to join the
thread. If noone is waiting to join the thread then a large memory leak
occurs.
* Made iax2_process_thread() deatach itself if nobody is waiting to join
the thread.
(closes issue ASTERISK-17339)
Reported by: Tzafrir Cohen
Patches:
asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified)
(closes issue ASTERISK-17825)
Reported by: wangjin
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Adds two new options to SIP peers allowing them to specify features (dynamic or builtin)
to use when sending INFO/record requests. Recordonfeature activates whatever feature
is specified when recieving a record: on request while recordofffeature activates
whatever feature is specified when receiving a record: off request. Both of these
features can be disabled by setting the feature to an empty string.
(closes issue ASTERISK-16507)
Reported by: Jon Bright
Review: https://reviewboard.asterisk.org/r/1634/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves destruction of sip peers to immediately after the general section of
sip.conf is read so that autocreatepeer setting can be read before deletion of peers.
If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting
will be skipped when purging the current SIP peer list.
(closes ASTERISK-16508)
Reported by: Kirill Katsnelson
Patches:
017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Chan_sip gives a dialog reference to the extension state callback and
assumes that when ast_extension_state_del() returns, the callback cannot
happen anymore. Chan_sip then reduces the dialog reference count
associated with the callback. Recent changes (ASTERISK-17760) have
resulted in the potential for the callback to happen after
ast_extension_state_del() has returned. For chan_sip, this could be very
bad because the dialog pointer could have already been destroyed.
* Added ast_extension_state_add_destroy() so chan_sip can account for the
sip_pvt reference given to the extension state callback when the extension
state callback is deleted.
* Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy()
and handle_statechange() now that the struct ast_state_cb has a destructor
to call.
* Ensure that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration.
* Fixed pbx.c statecbs_cmp() to compare the correct information. The
passed in value to compare is a change_cb function pointer not an object
pointer.
* Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
deadlocking when those locks are held during the callback.
* Removed unused lock declaration for the pbx.c store_hints list.
(closes issue ASTERISK-18844)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/1635/
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Fix a segfault if an attempt to answer a call is made between when
the inbound call gives up (and the channel is removed) and when the
device is notified and removes the call from the device.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid. When an MWI event would occur, this would cause a seg fault.
(closes issue ASTERISK-18663)
Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan
Patches:
blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
Review: https://reviewboard.asterisk.org/r/1610/
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Extracting sig_analog from chan_dahdi lost call progress detection
functionality.
* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
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The code that allowed admins to create users with domain-only uri's had
stopped to work in 1.8 because of the reqresp parser rewrites. This is
fixed now: if you have a [mydomain.com] sip user, you can register with
useraddr sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in the
allow list as well.
Reviewboard r1606 shows a list of registration combinations and which
SIP response codes are returned.
Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson
(closes issue ASTERISK-18389)
(closes issue ASTERISK-18741)
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The previous patch (r346040) incorrectly parsed the URI in the presence
of a port, e.g., user@hostname:port would fail as the port would be
double appended to the SIP message. This patch uses the parse_uri function
to correctly parse the URI into its username and hostname parts, and places
them in the correct fields in the sip_pvt structure.
(issue ASTERISK-18903)
Review: https://reviewboard.asterisk.org/r/1597/
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Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
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This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using the MessageSend application to send a SIP MESSAGE to a non-peer,
chan_sip attempted to validate the hostname or IP Address. In the process,
it stripped off the extension and failed to add it back to the sip_pvt
structure before transmitting. This patch adds the full URI passed in
from the message core to the sip_pvt structure.
(closes issue ASTERISK-18903)
Reported by: Shaun Clark
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1597/
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The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.
See ASTERISK-18702 it has a very good description of the issue.
I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.
* Added 'dtmf' enum value to sip.conf allowoverlap config option. The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.
* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.
* Fixed get_destination() inconsistency with the pickup extension
matching.
* Fixed initialization of PAGE3 of global_flags in reload_config().
(closes issue ASTERISK-18702)
Reported by: Pavel Troller
Review: https://reviewboard.asterisk.org/r/1517/
Review: https://reviewboard.asterisk.org/r/1582/
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Fix bug ASTERISK-16558 which dealt with the order of responses to incoming
streams defined by SDP.
Fix unreported bug where offering multiple same-type streams would cause
Asterisk to reply with an incorrect SDP response missing one or more streams
without a proper declination.
Fix bugs related to a single non-audio stream being offered with responses
requesting codecs that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Review: https://reviewboard.asterisk.org/r/1516/
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Another deadlock between the conlock/hints and channels/channel locking
orders.
* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().
(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
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The domain matching code prior to 1.8 used to manually remove the port
from the host:port string when determining if an incoming request
matched the list of domains. When switching to the new parsing
functions, the documentation implied that the "domain" was being
returned by these functions, when instead it was returning the
"hostport" as defined by RFC 3261. This led to confusion and resulted
in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when
domain=x.x.x.x was set in sip.conf.
This patch renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport strings
when dealing with domain matching.
Review: https://reviewboard.asterisk.org/r/1574/
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r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines
Fixes regression caused by r343635
There was a missing unlock for a function return that is only
present in Asterisk 10 and Asterisk Trunk.
(closes issue ASTERISK-18839)
Reported by: Michael L. Young
Patches:
asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
because of a deadlock between the dialogs container lock and the RWLOCK of
the events subscription list.
* Create dialogs_to_destroy container to hold dialogs that will be
destroyed.
* Ensure that the event subscription callback will never happen with an
invalid peer pointer by making the event callback removal the first thing
in the peer destructor callback.
NOTE: This particular deadlock will not happen with Asterisk 10, but some
of the changes still apply.
(closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Review: https://reviewboard.asterisk.org/r/1564/
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This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.
Review: https://reviewboard.asterisk.org/r/1562/
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Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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There were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during the
addition of the sipregs functionality, several possibilities for memory
leaks had been introduced. The insecure=port matching has always been
broken for anyone using the sipregs family. And, related, the broken
implementation forced those using sipregs to *still* have an ipaddr
column on their sippeers table.
Thanks Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which caused
the realtime_peer to have a completely unused code path.
This changeset fixes the leaks, the lookup inconsistenties and that
you won't need an ipaddr column on your sippeers table anymore (when
you're using sipregs). Beware that when you're using sipregs, peers
with insecure=port will now start matching!
(closes issue ASTERISK-17792)
(closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1395
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When an extension is removed from a context, its entry in the pattern match
tree is not deleted. Instead, the extension is marked as deleted. When an
extension is removed and re-added, if that extension is also a prefix of
another extension, several log messages would report an error and did not
check whether or not the extension was deleted before accessing the memory.
Additionally, if the extension was already in the tree but previously
deleted, and the pattern was at the end of a match, the findonly flag was
not honored and the extension would be erroneously undeleted.
Additionaly, it was discovered that an IAX2 peer could be unregistered
via the CLI, while at the same time it could be scheduled for unregistration
by Asterisk. The unregistration method now checks to see if the peer
was already unregistered before continuing with an unregistration.
(closes issue ASTERISK-18135)
Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1526
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The "No D-channels available! Using Primary channel as D-channel anyway!"
WARNING message has been confusing on non-NFAS setups. The message refers
to things that are NFAS specific.
* Changed the warning to several different warnings to be more accurate
for the situation and less confusing as a result:
"No D-channels up! Switching selected D-channel from X to Y.",
"No D-channels up!", and
"D-channel is down!".
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Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Google has recently make some changes (again) to their protocol. Rather then
patching asterisk to flip between the two different methods, we now allow both.
Lets hope this keeps Google Voice happy for a while.
(closes issue ASTERISK-18714)
Reported by: Iordan Iordanov
Patches:
chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311)
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Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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* Fix potential deadlocks in SIP and IAX blind transfer to parking.
* Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter). Created ast_park_call_exten() and ast_masq_park_call_exten()
to maintian API compatibility.
* Made masq_park_call() handle a failed ast_channel_masquerade() setup.
* Reduced excessive struct parkeduser.peername[] size.
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If a simple switch was started on a device and then a specific call
made (such as redial or speed dial), on timeout of the simple switch
the call would be attempted again. This patch only allows the simple
switch to make a call if the substate is still in the collecting
digits mode.
Also added small debug message to dialAndAactivate sub.
Tested by snuff and myself.
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r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
Merged revisions 340970 via svnmerge from
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r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
Quiet RTCP Receiver Reports during fax transmission
RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
code was added to support the bug fix.
(closes issue ASTERISK-18400)
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r340578 | twilson | 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) | 16 lines
Merged revisions 340534 via svnmerge from
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r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines
Update SIP realtime fullcontact regardless of caching
We should update the fullcontact field in the realtime table whether or
not rtcachefriends is set. There is no reason to treat a non-cached
realtime entity differently than a cached in this regard.
(closes issue ASTERISK-18446)
Reported by: wdoekes
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* Added a CLI "ss7 show channels" command that might prove useful for
future debugging.
* Made the incoming SS7 channel event check and gripe message uniform.
* Made sure that the DNID string for an incoming call is always
initialized.
(issue ASTERISK-17966)
Reported by: Kenneth Van Velthoven
Patches:
jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett
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* Fixed deadlock potential calling dialog_unlink_all() in
__sip_autodestruct(). Found by helgrind.
* Fixed deadlock potential in handle_request_invite() after calling
sip_new(). Found by helgrind.
* The sip_new() function now returns with the created channel already
locked.
* Removed the dead code that starts a PBX in in sip_new(). No sip_new()
callers caused that code to be executed and it was a bad thing to do
anyway.
* Removed unused parameters and return value from dialog_unlink_all().
* Made dialog_unlink_all() and __sip_autodestruct() safely obtain the
owner and private channel locks without a deadlock avoidance loop.
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Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r340165 | mjordan | 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) | 20 lines
Merged revisions 340164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines
Updated chan_sip to place calls on hold if SDP address in INVITE is ANY
This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::.
In this case, the call should be placed on hold. Previously, we checked for
the address being null; this patch keeps that behavior but also checks for
the ANY IP addresses.
Review: https://reviewboard.asterisk.org/r/1504/
(closes issue ASTERISK-18086)
Reported by: James Bottomley
Tested by: Matt Jordan
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Added some data to skinny packet structures to make compatible
with v17. Added protocolversion to device, set on registration
based on the version provided by device.
v17 includes some increased ip space for ip6. This patch increases
ip space in the packets but still only uses ip4. Some packet
structures duplicated (ip4 and ip6 types). ip4 type used unless
version is greater or equal to 17.
Tested by snuff and myself on 7961 with recent 8.5 firmware. Also
tested compatible with old 7960 and older 30VIPs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages
in v17 that are greater than the old 1000 bytes. Also add some
useful logging regarding packet and session handling.
A device (with protocol v17) was sending a packet with length
greater than 1000 which resulted in the TCP session being
destroyed and registration being retryed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) | 8 lines
Return -1 to skinny_session if register rejected.
If device registration is rejected, return -1 so that the session is
destroyed immediately. Previously, a segfault would occur on a
graceful shutdown if a register is rejected and the skinny_session
has not yet timed out.
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r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) | 9 lines
Remove log message on traverse session list.
On destroying a session, a list of sessions is traversed to find the
matching session. For each session not matching, skinny erroneously
logged that the session was not matched. While technically correct
the message was misleading, and tended to indicate errors that
were not there.
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There was an issue that the cap and confcap pointers for each line and device
were being memcpy'd so they all pointed to the same ast_format_cap. On
destroying, a segfault occured on the second call to the same struct.
skinny reload now works again as well.
Tested by snuff and myself.
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r339626 | rmudgett | 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) | 25 lines
Merged revisions 339625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines
Fix debugging messages generated by 'udptl debug'.
* Makes chan_sip set the tag to the channel name.
* Fixes received debug message sequence number.
* Removed tx/rx debug message type since it was hard coded to 0.
* Made udptl.c logged message header consistent if possible: "UDPTL (%s): ".
* Removed unused rx_expected_seq_no from struct ast_udptl.
(closes issue ASTERISK-18401)
Reported by: Kevin P. Fleming
Patches:
jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Matthew Nicholson
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r339088 | twilson | 2011-10-03 11:44:27 -0700 (Mon, 03 Oct 2011) | 17 lines
Merged revisions 339086 via svnmerge from
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r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines
Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places
After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame
is sent when a re-invite happens. If we receive a re-invite from a device
the waitstream_core was not aware of the new control frame and would drop
the call.
(closes issue ASTERISK-18610)
Reported by: Kristijan_Vrban
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r338801 | rmudgett | 2011-09-30 17:06:48 -0500 (Fri, 30 Sep 2011) | 19 lines
Merged revisions 338800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) | 12 lines
Fix segfault in analog_ss_thread() not checking ast_read() for NULL.
NOTE: The problem was reported against v1.6.2. It is unlikely to ever
happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely
to be used. The version in sig_analog.c has largely replaced it.
(closes issue ASTERISK-18648)
Reported by: Stephan Bosch
Patches:
jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Stephan Bosch
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r338323 | rmudgett | 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) | 12 lines
Merged revisions 338322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines
Make duplicate call ptr warning message more helpful.
* Adds the value of the call ptr to the duplicate call ptr message to help
trace why there is a duplicate call ptr.
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r337721 | rmudgett | 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) | 25 lines
Merged revisions 337720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines
Made ISDN not add numbering plan prefix strings to empty numbers.
When the Caller-ID is restricted, the expected behavior is for the
Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto
the Caller-ID number even if it is restricted (empty) causing the
Caller-ID to be the national prefix rather than blank.
This behavior was lost when sig_pri was extracted from chan_dahdi.
* Made not add prefix strings to empty connected line, calling, and ANI
number strings.
(closes issue ASTERISK-18577)
Reported by: Kris Shaw
Patches:
jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Kris Shaw
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r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines
Generate Security events in chan_sip using new Security Events Framework
Security Events Framework was added in 1.8 and support was added for AMI to generate
events at that time. This patch adds support for chan_sip to generate security events.
(closes issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
Review: https://reviewboard.asterisk.org/r/1362/
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r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines
Forgot to svn add new files to r337595
Part of Generating security events for chan_sip
(issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (License #5026) by Michael L. Young
Reviewboard: https://reviewboard.asterisk.org/r/1362/
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r337487 | irroot | 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) | 16 lines
Merged revisions 337486 via svnmerge from
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r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines
If IP address is used in chan_h323 host parameter of peer configuration.
module tries to resolve IP address to IP address and fails.
Simple fix to set family of socket this is a hangover from ipv6 changes.
(closes issue ASTERISK-18237)
(issue ASTERISK-17278)
(issue ASTERISK-17500)
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r337008 | rmudgett | 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) | 22 lines
Merged revisions 337007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines
Check if a channel was created before using the pointer in sig_ss7_new_ast_channel().
Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace.
* Added some missing libss7 access lock protection.
* Prevent cancelling the ss7_linkset() thread at inoportune times just
like the pri_dchannel() thread.
(issue ASTERISK-17955)
Reported by: Ian M Sherman
Patches:
jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett
(attached to related ASTERISK-17966)
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r336978 | rmudgett | 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) | 28 lines
Merged revisions 336977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines
Fix deadlock from not releasing SS7 linkset lock.
sig_ss7_hangup() failed to release the SS7 linkset lock if the call had
the alreadyhungup flag set.
* Made unlock the SS7 linkset lock in sig_ss7_hangup() if the
alreadyhungup flag is set.
* Made ss7_start_call() not hold any locks while creating the channel for
an incoming call to prevent deadlock.
* Made ss7_grab() a void function, since it could never fail, to simplify
calling code.
* Made obtain the channel lock to do softhangup in some places.
Patches:
jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett
JIRA AST-668
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r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
Allow Setting Auth Tag Bit length Based on invite or config option
Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
Curently only 80 bit is supported.
The outgoing invite will use the taglen of the incoming invite preventing
one-way audio.
(Closes issue ASTERISK-17895)
Review: https://reviewboard.asterisk.org/r/1173/
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r336500 | irroot | 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) | 19 lines
Merged revisions 336499 via svnmerge from
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r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines
A long time ago in a galaxy far far away a IPv6 update was made,
chan_h323 was not updated causeing all to flee to chan_ooh323.
the brave Jedi [asterisk developers] pondered this miscarrige of justice
and restored order to the force for the sake of closing out 2 old issues.
(closes issue ASTERISK-17278)
(closes issue ASTERISK-17500)
Reported by: dread, sybasesql
Tested by: irroot
Reviewed by: IRC (russellb, kpfleming)
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r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines
Merged revisions 336294 via svnmerge from
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r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
break when starting a call with directmedia. This patch queues a new type of control frame
so that our RTP bridge loop can properly detect when these situations occur and check to see
if peers need to be updated in order to send their media to the proper location.
(Closes issue ASTERISK-18340)
Reported by: Thomas Arimont
(Closes issue ASTERISK-17725)
Reported by: kwk
Tested by: twilson, jrose
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r336167 | irroot | 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) | 22 lines
Merged revisions 336166 via svnmerge from
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r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines
The round robin routing routine in chan_misdn.c is broken.
it rotates between ports but never checks the channels in the ports.
i have extensivly tested it and verified it works on 1 upto 4 ports.
before the patch only 1 out of each port was used now all are used as
expected.
(closes issue ASTERISK-18413)
Reported by: irroot
Tested by: irroot
Reviewed by: irroot
Review: https://reviewboard.asterisk.org/r/1410/
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r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines
Merged revisions 335319 via svnmerge from
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r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines
Lock the peer->mvipvt to avoid crashes with SIP history enabled
After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
which cause issues with SIP history additions in combination with the max limit for
number of history entries.
Review: https://reviewboard.asterisk.org/r/1373/
(closes issue ASTERISK-18288)
Thanks to irrot for peer review. Work with this bug funded by IPvision AS
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r335321 | kmoore | 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) | 16 lines
Merged revisions 335320 via svnmerge from
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r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines
Prevent IAX2 from getting IPv6 addresses via DNS
IAX2 does not support IPv6 and getting such addresses from DNS can cause error
messages on the remote end involving bad IPv4 address casts in the presence of
IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6
addresses via DNS queries.
(closes issue ASTERISK-18090)
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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* Make check_rtp_timeout() remember the values returned by
ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling them.
(closes issue ASTERISK-18319)
Reported by: Rob Gagnon
Patches:
issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1377/
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r334013 | rmudgett | 2011-08-31 11:00:49 -0500 (Wed, 31 Aug 2011) | 30 lines
Merged revisions 334012 via svnmerge from
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r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines
No DAHDI channel available for conference, user introduction disabled.
The following error will consistently occur when trying to dial into a
MeetMe conference when the server does not have DAHDI hardware installed:
app_meetme.c: No DAHDI channel available for conference, user introduction
disabled (is chan_dahdi loaded?)
While chan_dahdi is loaded correctly during compilation and install of
Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf
configuration file in /etc/asterisk is not created by FreePBX if hardware
does not exist, causing MeetMe to be unable to open a DAHDI pseudo
channel.
* Allow chan_dahdi to create a pseudo channel when there is no
chan_dahdi.conf file to load.
(closes issue ASTERISK-17398)
Reported by: Preston Edwards
Patches:
jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
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r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines
Merged revisions 334009 via svnmerge from
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r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines
Call pickup race leaves orphaned channels or crashes.
Multiple users attempting to pickup a call that has been forked to
multiple extensions either crashes or fails a masquerade with a "bad
things may happen" message.
This is the scenario that is causing all the grief:
1) Pickup target is selected
2) target is marked as being picked up in ast_do_pickup()
3) target is unlocked by ast_do_pickup()
4) app dial or queue gets a chance to hang up losing calls and calls
ast_hangup() on target
5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
ast_channel_masquerade(), ast_hangup() completes successfully and the
channel is no longer in the channels container.
6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the
masquerade on the dead channel.
7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel
8) bad things happen while doing the masquerade and in the process
ast_do_masquerade() puts the dead channel back into the channels container
9) The "orphaned" channel is visible in the channels list if a crash does
not happen.
This patch does the following:
* Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel
and not release the channel lock until that has happened.
* Made __ast_channel_masquerade() not setup a masquerade if either channel
has AST_FLAG_ZOMBIE set.
* Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work.
(closes issue ASTERISK-18222)
Reported by: Alec Davis
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
(closes issue ASTERISK-18273)
Reported by: Karsten Wemheuer
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
Review: https://reviewboard.asterisk.org/r/1400/
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r334007 | kmoore | 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) | 14 lines
Merged revisions 334006 via svnmerge from
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r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines
Correct an AMI protocol violation with SIPshowpeer
The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are
ended by using \r\n this confuses any interfacing script.
(closes issue ASTERISK-17486)
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r333837 | twilson | 2011-08-29 16:41:13 -0500 (Mon, 29 Aug 2011) | 22 lines
Merged revisions 333836 via svnmerge from
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r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines
Refresh peer address if DNS unavailable at peer creation
If Asterisk starts and no DNS is available, outbound registrations will fail
indefinitely. This patch copies the address from the sip_registry struct, which
will be updated, to the peer->addr when necessary.
If dnsmgr is enabled, the registration fails without the patch because even
though the address on the registry is updated via dnsmgr, the address is just
copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update
the address that is copied to the sip_pvt or peers.
Closes issue ASTERISK-18000
Review: https://reviewboard.asterisk.org/r/1335/
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GCC 4.6 detects variables that get assined to, but never used later.
Also removes some remmed-out lines that become invalid.
(closes issue ASTERISK-18336)
Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>,
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r332504 | kmoore | 2011-08-18 14:29:15 -0500 (Thu, 18 Aug 2011) | 15 lines
Merged revisions 332503 via svnmerge from
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r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | 8 lines
CRC4 in "dahdi show status" gives wrong impression to T1 users
Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in
more situations without confusing users, especially since T1 lines use CRC6
instead of CRC4.
(closes issue AST-471)
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r332265 | rmudgett | 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) | 33 lines
Merged revisions 332264 via svnmerge from
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r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines
Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards.
France Telecom brings layer 2 and layer 1 down on BRI lines when the line
is idle. When layer 1 goes down Asterisk cannot make outgoing calls and
the HA8 and HB8 cards also get IRQ misses.
The inability to make outgoing calls is because the line is in red alarm
and Asterisk will not make calls over a line it considers unavailable.
The IRQ misses for the HA8 and HB8 card are because the hardware is
switching clock sources from the line which just brought layer 1 down to
internal timing.
There is a DAHDI option for the B410P card to not tell Asterisk that layer
1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp
teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards:
"modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear
up the IRQ misses when the telco brings layer 1 down.
* Add layer 2 persistence option to customize the layer 2 behavior on BRI
PTMP lines. The new option has three settings: 1) Use libpri default
layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer
brings it down. 3) Leave layer 2 down when the peer brings it down.
Layer 2 will be brought up as needed for outgoing calls.
JIRA AST-598
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r332119 | jrose | 2011-08-16 12:45:38 -0500 (Tue, 16 Aug 2011) | 23 lines
Merged revisions 332118 via svnmerge from
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r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines
ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs
Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on whichever mailbox
triggered the mwi event. Now all of them get counted regardless. Also fixes a bug
involving parsing of the mailbox option in sip.conf so that trailing and leading
spaces before/after commas are trimmed.
(closes issue ASTERISK-18067)
Reported by: aragon
(closes issue ASTERISK-15479)
Reported by: Ben Winslow
Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow
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r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines
In 10 and trunk this option is disabled by default.
Merged revisions 332021 via svnmerge from
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r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
AST-580
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r331956 | rmudgett | 2011-08-15 12:35:03 -0500 (Mon, 15 Aug 2011) | 20 lines
Merged revisions 331955 via svnmerge from
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r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) | 13 lines
Fix some minor chan_dahdi config load issues.
* Address chan_dahdi.conf dahdichan option todo item about needing line
number.
* Make ignore_failed_channels option also apply to dahdichan option.
* Don't attempt to create a default pseudo channel if the chan_dahdi.conf
channel/channels option is not allowed.
* Add a similar check for dahdichan in normal chan_dahdi.conf sections as
is done in users.conf.
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r331772 | rmudgett | 2011-08-12 13:59:45 -0500 (Fri, 12 Aug 2011) | 15 lines
Merged revisions 331771 via svnmerge from
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r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) | 8 lines
Suppress warning message when using DAHDITransfer or DAHDIHangup.
* The fake event should only be processed by the channel that currently
owns the private and not the associated call waiting or 3-way channel.
JIRA AST-620
JIRA SWP-3616
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r331715 | rmudgett | 2011-08-12 12:54:47 -0500 (Fri, 12 Aug 2011) | 29 lines
Merged revisions 331714 via svnmerge from
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r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) | 22 lines
AMI actions DAHDIHangup and DAHDITransfer have no effect.
The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI
channel. These two AMI actions are highly specialized to analog channels
and appear to make the channel behave like a jack port for headsets.
* Made the faked DAHDI event get processed before a normal media stream
read in dahdi_read() instead of trying to trigger an exception read by
setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long
ago that changed how AST_FLAG_EXCEPTION is processed in the core.
Unfortunately, the faked DAHDI events no longer worked when that happened.
* Updated the DAHDI AMI action documentation for the following actions:
DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff,
DAHDIShowChannels, and DAHDIRestart.
* Made use sscanf() instead of atoi() for better error checking of the
DAHDIChannel header string.
JIRA AST-620
JIRA SWP-3616
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r331518 | kmoore | 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) | 17 lines
Merged revisions 331517 via svnmerge from
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r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines
SIP Notify via AMI or CLI leaks SIP PVTs
Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref following it
corrects the issue as seen via REF_DEBUG. The unref existed in a distant
revision and it appears as though the wrong ref operation was removed.
(closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
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In order to address a compatability issue with certain features on certain devices
which rely on display name content to change behavior, initreqprep in chan_sip.c
has been changed to no longer substitute cid_number into the display name when
cid_name isn't present. Instead, it will send no display name in that case.
(closes issue ASTERISK-16198)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/1341/
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r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines
Merged revisions 331248 via svnmerge from
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r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines
Misc minor items found in code.
* Add some reentrancy protection in pbx.c when creating the contexts_table
hash table.
* Fix inverted test in chan_sip.c conditional code.
* Fix uninitialized variable and use of the wrong variable in chan_iax2.c.
* Fix test of return value in app_parkandannounce.c. Explicitly testing
for -1 is bad if the function does not actually return that value when it
fails.
* Fixup some comments and add some curly braces in features.c.
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r330706 | kmoore | 2011-08-03 08:39:06 -0500 (Wed, 03 Aug 2011) | 17 lines
Merged revisions 330705 via svnmerge from
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r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | 10 lines
Call pickup broken for DAHDI channels when beginning with #
The call pickup feature did not work on DAHDI devices for anything other than
feature codes beginning with * since all feature codes in chan_dahdi were
originally hard-coded to begin with *. This patch is also applied to
chan_dahdi.c to fix this bug with radio modes.
(closes issue AST-621)
Review: https://reviewboard.asterisk.org/r/1336/
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r330586 | dvossel | 2011-08-02 11:17:59 -0500 (Tue, 02 Aug 2011) | 15 lines
Merged revisions 330581 via svnmerge from
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r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) | 8 lines
Fixes crash in chan_iax2.
Fixes crash in chan_iax2 resulting from an edge case in the
way control frames are queued during calltoken negotiation is complete.
(closes issue ASTERISK-17610)
Reported by: mgrobecker
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There is a fairly common pattern making its way through the code base where we
put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER.
The purpose is so that it can be passed into the object hash function.
However, this really seems like a hack and potentially error prone. This patch
is a first stab at approach to avoid having to do that.
It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these
situations. Then, the hash function can know whether it was given an object or
some custom data to hash.
The patch also changes some uses of ao2_find() for iax2_user and iax2_peer
objects to reflect how OBJ_KEY would be used.
So long, and thanks for all the fish.
Review: https://reviewboard.asterisk.org/r/1184/
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r330051 | rmudgett | 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) | 29 lines
Merged revisions 330050 via svnmerge from
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r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines
Merged revisions 330033 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines
Datacalls with B410P fail.
Incoming and outgoing call legs of a data call are using different
formats: a-law, u-law. When the call is bridged, the media stream is run
through translation to convert the media formats. The translation is bad
for data calls.
* Make incoming call that does not explicitly specify u-law or a-law use
the DAHDI channel's default law. The outgoing call always uses the
default law from the DAHDI channel.
(closes issue ABE-2800)
Patches:
jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett
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r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
Merged revisions 328935 via svnmerge from
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r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
Inband DTMF regression
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
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r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328823 via svnmerge from
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r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
RTP bridge away with inband DTMF and feature detection
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
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r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328608 via svnmerge from
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r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
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r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines
INVITE 403 Forbidden response always retransmits the maximum times.
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
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This patch adds pass-through support for CELT. CELT
formats are defined in codecs.conf and can be configured
to any sample rate a CELT endpoint supports. This patch also
addresses a crash in channel.c resulting from a frame list being
freed incorrectly. This crash was discovered while testing a CELT
translator which had to split encoded audio into multiple frames.
The codec translator is not a part of this patch, but may be
contributed in the future.
Review: https://reviewboard.asterisk.org/r/1294/
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r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
Add the attribute "type" to each "<use>" for menuselect.
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
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r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines
Used auth= parameter freed during "sip reload" causes crash.
If you use the auth= parameter and do a "sip reload" while there is an
ongoing call. The peer->auth data points to free'd memory.
The patch does several things:
1) Puts the authentication list into an ao2 object for reference counting
to fix the reported crash during a SIP reload.
2) Converts the authentication list from open coding to AST list macros.
3) Adds display of the global authentication list in "sip show settings".
(closes issue ASTERISK-17939)
Reported by: wdoekes
Patches:
jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/1303/
JIRA SWP-3526
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r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines
Better way to get chan and pvt lock for issue ASTERISK-17431.
Redoes -r308945 for issue ASTERISK-17431 deadlock fix for
sip_set_udptl_peer() and sip_set_rtp_peer().
* Lock the channels in the defined order and avoid the need for a deadlock
avoidance loop.
* Lock the channel before getting the pointer to the private structure to
be sure that the pointer will not change due to a masquerade or channel
hangup.
* To preserve sanity, check that chan and p->owner are the same. (Pointer
rearangements should not happen without the protection of locks because
bad things tend to happen otherwise.)
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Terry Wilson
Mark Michelson
Richard Mudgett
Kinsey Moore
Jonathan Rose
Joshua Colp
Stefan Schmidt
Matthew Jordan
Russell Bryant
Kevin P. Fleming
Damien Wedhorn
Walter Doekes
Alexandr Anikin
Tilghman Lesher
Gregory Nietsky
David Vossel
Matthew Nicholson
Paul Belanger
Igor Goncharovskiy
Leif Madsen
Olle Johansson
Jason Parker
Tzafrir Cohen
Sean Bright
Mark Murawki