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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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
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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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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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/
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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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* 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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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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* 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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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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* 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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r340165 | mjordan | 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) | 20 lines
Merged revisions 340164 via svnmerge from
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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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r339626 | rmudgett | 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) | 25 lines
Merged revisions 339625 via svnmerge from
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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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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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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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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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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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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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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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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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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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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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r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines
Misc minor changes in chan_sip.
* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().
* Removed a redundant static prototype.
* Some typos.
* Some whitespace.
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If a SDP does not modify the session, we ignore it. However, we were defaulting
no text and video support to true before checking to see if the sdp modified
anything or not. This would result in process_sdp ignoring an sdp but removing
video and text from the call during direct media reinvites.
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r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines
When subscribing MWI to an unsolicited mailbox the first notification is incorrect.
A remote peer subscribed to MWI with the unsolicited option and a local
phone subscribed to the remote mailbox. The notify message-summary events
are sent correctly except for the first one when subscribing, which will
always be 0. This means the phone MWI indicator will be wrong until the
mailbox read/unread count changes and the event is fired.
Looks like this is a regression from ASTERISK-16149.
* Fix the logic to check the cache and if allowed then fallback to
manually counting mailbox messages.
(closes issue ASTERISK-17997)
Reported by: rsw686
Patches:
jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett
Tested by: rsw686
JIRA SWP-3551
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r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
Stop sending IPv6 link-local scope-ids in SIP messages
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
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Also fixed a reference leak in an error path in sip_msg_send().
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r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines
Timout or error on INFO or MESSAGE transaction causes call to be lost.
When exchanging INFO messages within a call, 4xx error causes the call to
be disconnected although RFC 2976 explicitly states that such transactions
do not modify the state of the dialog.
When exchanging MESSAGE messages within a call, 4xx error causes the call
to be disconnected. To provide least surprise, we should not disconnect
the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428
Section 2)
(closes issue ASTERISK-17901)
Reported by: neutrino88
Review: https://reviewboard.asterisk.org/r/1257/
Review: https://reviewboard.asterisk.org/r/1258/
JIRA SWP-3486
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r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines
Ignore media offers with a port of 0
Section 5.1 of RFC3264 states:
A port number of zero in the offer indicates that the stream is offered
but MUST NOT be used.
(closes issue ASTERISK-17845)
Reported by: jacco
Patches:
issue19281_2.patch uploaded by jacco (license 1277)
Tested by: jacco, twilson
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r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines
Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT
It turned out that this was causing NAT=Yes to always use rport when present which was
against 1.6.2 behavior and the check itself was redundant since the only way this
segment of code could be reached was if RPORT_PRESENT was already evaluated as true
earlier.
(closes issue ASTERISK-17789)
Reported by: byronclark
Patches:
use_sip_nat_force_rport.patch uploaded by byronclark (license 1200)
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r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines
Make handle_request_publish do dialog expiration and destruction.
This patch fixes handle_request_publish so that it does dialog expiration and destruction.
Without this patch the incoming PUBLISH requests will get stuck in the dialog list.
Restarting asterisk is the only way to remove them.
Personal observation on one system the server hung up while looping through the channels
rendering asterisk unusable and all sip phones unregisterd when they try reregister
more requests are added.
(closes issue #18898)
Reported by: gareth
Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot
Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915
Review: https://reviewboard.asterisk.org/r/1253
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Don't send all messages to 's'. Get the destination from the request URI.
(Found using automated test cases).
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Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
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r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines
Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic.
Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails.
(closes issue #19346)
Reported by: kobaz
Tested by: kobaz,JonathanRose
Review: [full review board URL with trailing slash]
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r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines
ast_sockaddr_resolve() in netsock2.c may deref a null pointer
Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables
(closes issue #19346)
Reported by: kobaz
Patches:
netsock2.patch uploaded by kobaz (license 834)
Tested by: kobaz, Marquis
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r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines
Native SIP CCSS sends bad CC cancel SUBSCRIBE message.
The SUBSCRIBE message used to cancel a CC request has incorrect To/From
SIP headers. They are reversed and the dialog tags are the same when they
should not be. If pedantic mode was disabled, then the cancel would have
succeeded despite the incorrect message.
* The SIP_OUTGOING flag was not set correctly for the dialog and I had to
move some CC subscribe handling code as a result.
* Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a
CC request SUBSCRIBE message comes in and the CC instance is not found,
the 404 response was duplicated.
JIRA AST-568
JIRA SWP-3493
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r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines
This commit modifies the way polling is done on TLS sockets.
Because of the buffering the TLS layer does, polling is unreliable. If poll is
called while there is data waiting to be read in the TLS layer but not at the
network layer, the messaging processing engine will not proceed until something
else writes data to the socket, which may not occur. This change modifies the
logic around TLS sockets to only poll after a failed read on a non-blocking
socket. This way we know that there is no data waiting to be read from the
buffering layer.
(closes issue #19182)
Reported by: st
Patches:
ssl-poll-fix3.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines
Adds legacy_useroption_parsing to address interoperability concerns.
With the new option engaged, Asterisk should interpret user fields with useroptions
contained within the userfield of the uri by stripping them out of the original message
whenever a semicolon is encountered in the userfield string.
(closes issue #18344)
Reported by: danimal
Tested by: jrose
Review: https://reviewboard.asterisk.org/r/1223/
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r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines
Merged revisions 319653 via svnmerge from
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r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines
Merged revisions 319652 via svnmerge from
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r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines
Make sure everyone gets an unhold when a transfer succeeds
Some phones, like the Snom phones, send a hold to the transfer target after
before sending the REFER. We need to make sure that we unhold the parties
that are being connected after the masquerade. If Local channels with the /nm
option are used when dialing the parties, hold music would still be playing on
the transfer target, even after being connected with the transferee.
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r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines
Unbreak the storing of registrations for restart
The fix for issue 18882 broke retrieving non-realtime peers from the ast_db
on restart/reload. This patch tries to unbreak things while leaving the intent
of the original fix intact.
(closes issue #19318)
Reported by: remiq
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: lmadsen, remiq
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state of the channel reverts to unknown this should be rejected.
this is important for negotiating T.38 gateway see #13405
This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected.
Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states.
(closes issue #18889)
Reported by: irroot
Tested by: irroot, darkbasic, mnicholson
Review: https://reviewboard.asterisk.org/r/1115
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r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines
Fix directed group pickup feature code *8 with pickupsounds enabled
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.
1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
2). dialplan applications for directed_pickups shouldn't beep.
3). feature code for directed pickup should beep on success/failure if configured.
Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
Moved app_directed:pickup_do() to features:ast_do_pickup().
Functions below, all now use the new ast_do_pickup()
app_directed_pickup.c:
pickup_by_channel()
pickup_by_exten()
pickup_by_mark()
pickup_by_part()
features.c:
ast_pickup_call()
(closes issue #18654)
Reported by: Docent
Patches:
ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett
Review: https://reviewboard.asterisk.org/r/1185/
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r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines
Merged revisions 318548 via svnmerge from
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r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines
Clean up several chan_sip reference leaks
Several situations in the code could lead to peers or sip_pvt references
being leaked. This would cause RTP ports to never be destroyed (leading
to exhaustion of all available RTP ports) and memory leaks.
The original patch for this issue from rgagnon was the result of an
obscene amount of testing and hard work, for which I am very grateful. I
did some cleanup and added a few additional refcount fixes that I found.
(closes issue #17255)
Reported by: kvveltho
Patches:
tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202)
Tested by: rgagnon, twilson, wdoekes, loloski
Review: https://reviewboard.asterisk.org/r/1101/
Review: https://reviewboard.asterisk.org/r/1207/
Review: https://reviewboard.asterisk.org/r/1210/
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r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines
Merged revisions 318331 via svnmerge from
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r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines
Don't offer video to directmedia callee unless caller offered it as well
Make sure that when directmedia is enabled, that video is not offered to the
callee even if it supports it. p->vrtp will not exist since the caller didn't
offer video.
(closes issue #19195)
Reported by: one47
Patches:
sip_cant_add_video_rtp uploaded by one47 (license 23)
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r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) | 10 lines
chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer.
Don't duplicate variables on the sip_pvt. Just reset the variable list each
time.
(closes issue #19202)
Reported by: wdoekes
Patches:
issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717)
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r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines
chan_sip: fix a deadlock in check_rtp_timeout.
Don't block doing silly deadlock avoidance. Just return and try again later.
The funciton gets called often enough that it's fine. Also, this change was
already made in trunk.
(closes issue #18791)
Reported by: irroot
Patches:
chan_sip.rtptimeout.patch uploaded by irroot (license 52)
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r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines
Fix SIP connected line updates.
This patch fixes a couple SIP connected line update problems:
1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured. Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.
2) The connected line should not be updated on initial connect if there is
no connected line information. Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.
(closes issue #18367)
Reported by: GeorgeKonopacki
Patches:
issue18367_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1199/
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r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines
Fix some consistency issues with jitterbuffer config.
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
(closes issue #19083)
Reported by: rgagnon
Patches:
issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
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r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | 10 lines
Resolves a deadlock that occurs during sip_new
This is based on an uncommitted patch by jpeeler for the issue. Instead of
relocking and then unlocking the channel though, we keep the lock on the channel
until we are finished doing what we need to the channel.
(closes issue #18441)
Reported by: Alric
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r317281 | russell | 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) | 29 lines
Merged revisions 317255 via svnmerge from
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r317255 | russell | 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) | 22 lines
Merged revisions 317211 via svnmerge from
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r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines
chan_sip: fix broken realtime peer count, fix memory leak
This patch addresses two bugs in chan_sip:
1) The count of realtime peers and users was off. The increment checked the
value of the caching option, while the decrement did not.
2) Add a missing regfree() for a regex.
(closes issue #19108)
Reported by: vrban
Patches:
missing_regfree.patch uploaded by vrban (license 756)
sip_object_counter.patch uploaded by vrban (license 756)
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r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) | 8 lines
Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer
abruptly disappears. This mostly occurs after a successful registration.
(closes issue #17544)
Reported by: marcelloceschia
Patches:
(modified) tcptls.patch uploaded by st (license 907)
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r316617 | dvossel | 2011-05-04 08:44:41 -0500 (Wed, 04 May 2011) | 19 lines
Merged revisions 316616 via svnmerge from
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r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) | 12 lines
Fixes session-timers=refuse not being enforced for *caller*
During handle_request_invite, the session timer mode was retrieved from
a cached variable. This patch forces a peer lookup of the session timer
mode in the case of an incoming invite.
(closes issue #18804)
Reported by: wdoekes
Patches:
issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717)
issue_18804_v2.diff uploaded by dvossel (license 671)
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The original implementation of this did the merging of all contexts with the
same name in the realtime layer, but that implementation severely breaks
drivers which use the same context name (e.g. iax.conf, type={peer,user}).
Therefore, the implementation needs to do the merging for particular entries
only, based upon what contexts would allow that in the channel driver itself.
This implementation is for chan_sip only, but others could be added in the
future.
(closes issue #17957)
Reported by: marcelloceschia
Patches:
chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079)
Tested by: tilghman
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r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) | 9 lines
Never put the Require: timer header in an Invite.
This has already been discussed and should have been resolved earlier. View
revsion 285565's log for more information about why it is important to not
put timer in the Require header.
(closes issue #18704)
Reported by: mfrager
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r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines
Merged revisions 315893 via svnmerge from
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r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
Merged revisions 315891 via svnmerge from
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r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
Fix our compliance with RFC 3261 section 18.2.2.
This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.
(closes issue #18951)
Reported by: jmls
Patches:
issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines
Merged revisions 314620 via svnmerge from
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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r314550 | twilson | 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) | 13 lines
Merged revisions 314549 via svnmerge from
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r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just allocate
enough for the sip_pkt.
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r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines
Remove the need for deadlock avoidance in chan_sip do_monitor.
Deadlock avoidance between the sip pvt and the pvt->owner is
very difficult. Now that channel's are ao2 objects, this complication
is no longer necessary. It turns out the pvt's msg queue only
exists because of deadlock avoidance (when deadlock avoidance fails
msgs were added to a queue to be processed later), so this goes away as well.
The technique used in the new sip_lock_pvt_full() function should
be used as a template for replacing all locations where deadlock
avoidance occurs between a channel tech_pvt and the pvt's owner.
My hope is that this will begin a reversal of the invalid channel
driver locking architecture we have been using for so long.
This patch also resolves an issue where the pvt->owner gets
unlocked during processing the msg queue.
(closes issue #18690)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/1182/
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r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
sip codec negotiation of dynamic rtp payloads error fix
This patch fixes how chan_sip handles dynamic rtp payload types
it does not understand. At the moment if a dynamic payload's mime
type does not match one we understand, the payload does not get
removed from our payload table. As a result of this, the payload
is set to whatever dynamic codec we use internally for that payload
number on outgoing INVITES. This is incorrect.
This patch fixes this by properly checking the rtpmap set function's
return code to make sure it was found. The function can return both
-1 and -2 depending on the source of the mismatch. We were just
checking -1 explicitly.
Review: https://reviewboard.asterisk.org/r/1169/
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r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) | 15 lines
Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension.
The get_destination() function was not using the "s" extension when the
request URI did not specify an extension. This is a regression caused
when the URI parsing code was extracted into parse_uri().
Made get_destination() substitute the "s" extension when the parsed URI
results in an empty string.
(closes issue #18348)
Reported by: shmaize
Patches:
issue18348_v1.8.patch uploaded by rmudgett (license 664)
Tested by: shmaize
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r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines
Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL.
This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings.
(closes issue #18759)
Reported by: bklang
Patches:
null-strings.patch uploaded by bklang (license 919)
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r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines
Merged revisions 309255 via svnmerge from
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r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines
Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.
Since it's a duplicate, nothing is going to be done, so delme doesn't need to
be set at all. Strangely, when this was added, this was being set to 1 in 1.6,
and 0 in trunk.
(issue AST-439)
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r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines
Fix Deadlock with attended transfer of SIP call
Call path
sip_set_rtp_peer (locks chan then pvt)
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper (locks p->owner)
But by the time p->owner lock was attempted, seems as though chan and p->owner were different.
So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.
(closes issue #18837)
Reported by: alecdavis
Patches:
bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, Irontec, ZX81, cmaj
Review: [https://reviewboard.asterisk.org/r/1126/]
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-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
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Terry Wilson
Richard Mudgett
Mark Michelson
Kinsey Moore
Jonathan Rose
Joshua Colp
Stefan Schmidt
Matthew Jordan
Russell Bryant
Walter Doekes
Tilghman Lesher
David Vossel
Matthew Nicholson
Paul Belanger
Leif Madsen
Olle Johansson
Gregory Nietsky
Jason Parker
Sean Bright
Mark Murawki
Kevin P. Fleming
Brett Bryant
Alec L Davis