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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* 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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When an inital INVITE occurs that contains image media, a channel
is not yet associated with the SIP dialog. The file descriptor
associated with the udptl session needs to be set in
initialize_udptl or in sip_new to account for this scenario.
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350550 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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When someone does Set(CALLERPRES()=unavailable) (or
Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows
"Anonymous" <anonymous@anonymous.invalid>. When sendrpid=yes/pai, the From
header will still display the callerid info, even though we supply an rpid
header with the anonymous info. It seems like we shouldn't leak that info in
any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems
to indicate that one shouldn't send identifying info in the From in this case.
This patch anonymizes the From header as well even when sendrpid=yes/pai.
(closes issue ASTERISK-16538)
Review: https://reviewboard.asterisk.org/r/1649/
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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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Some ISDN switches complain or block the call if the RDNIS number is
empty.
* Made chan_iax2 not save a RDNIS number into the ast_channel if the
string is blank. This is what other channel drivers do.
(closes issue ASTERISK-17152)
Reported by: rmudgett
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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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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is fortunate that the typo does not alter generated code since the
e->restart.channel and e->ring.channel members are in the same position.
(closes issue ASTERISK-18868)
Reported by: zvision
Patches:
sig_pri.c.diff (License #5755) patch uploaded by zvision
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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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Residual changes for Asterisk v10 branch from ASTERISK-18747 after
https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs
callid hash key change fix.
* Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from
dialogs_rtpcheck. This is an optimization to avoid an unneeded
lock/unlock and object search when using ao2_unlink.
* Prevent crash in check_rtp_timeout() if dialog->rtp is NULL.
Review: https://reviewboard.asterisk.org/r/1557/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343900 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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The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command
pulled the flags from the incorrect global flags page. These are now read from
sip global flags page 0.
(closes issue AST-711)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changing an object value used as a container key requires removing the
object from the container and reinserting it.
* Created change_callid_pvt() to call instead of build_callid_pvt(). The
change_callid_pvt() will correctly change the dialog callid so the ao2
conainter can explicitly unlink it.
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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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The original REGISTERTRYING flag, in addition to being impossible to
check, also encroached on the space for the flag above it. This
patch moves the flags that were below REGISTERTRYING back to where
they were as though we had just removed the REGISTERTRYING option.
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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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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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This behavior matches up more closely with the way invite/register/etc are handled.
This patch also modifies some adjacent code for code style compliance. Pretty minor.
(closes issue ASTERISK-17616)
Reported by: Jeremy Kister
Patches:
chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232)
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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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If a SIP dial string contains a numeric hostname that is not a peer name,
don't try to resolve it as it is unlikely that someone really means
Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that
create_addr returns -1 if an address isn't resolved so that we don't
attempt to send SIP requests to an address that doesn't resolve.
(closes issue ASTERISK-17146, ASTERISK-17716)
Review: https://reviewboard.asterisk.org/r/1532/
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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 parse_uri was called with an empty URI, some pointers would be
modified and an invalid read could result. This patch avoids calling
parse_uri with an empty contact uri when parsing REGISTER requests.
AST-2011-012
(closes issue ASTERISK-18668)
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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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lastmsgssent was removed from chan_sip and the old behavior of
sending a mwi notify on register [except when subscribemwi is set]
was restored but this must only happen when registration succeeds.
leaking information for unsuccessful registrations is not secure.
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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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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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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 (in trunk) and myself.
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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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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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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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r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) | 6 lines
Update documentation for SIP_HEADER.
The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated
in trunk, but not in 1.8 or 10, so I'm making them match.
(Closes issue ASTERISK-18640)
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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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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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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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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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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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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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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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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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r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines
lock the channel before calling ast_bridged_channel() to prevent a seg fault.
AMI agents list called on shutdown causes a segfault, introducing proper locking
will prevent this.
(closes issue ASTERISK-18092)
Reported by: agustina
Patches: chan_agent.patch (License #5041) patch uploaded by irroot
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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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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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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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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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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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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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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>,
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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines
Use the device name and not the channel name to initialize the device state.
Correct ASTERISK-11323 implementation as I don't see how it ever worked as
claimed when it used the channel name and not the device name.
(issue ASTERISK-11323)
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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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https://origsvn.digium.com/svn/asterisk/branches/1.8
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER.
Initial value must be the time to autoanswer in ms, then optionally :BEEP
to play a tone when answered and :MUTE to mute the mic when answering.
eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and
play a beep. just 3000 would answer afer 3 secs of ringing with no
beep and full two way audio.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything.
Review: https://reviewboard.asterisk.org/r/1256/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Don't send all messages to 's'. Get the destination from the request URI.
(Found using automated test cases).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines
Merged revision 319468 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines
The mISDN HDLC mode is prevented on dialed channels.
The use of mISDN HDLC mode is prevented if the mISDN dial technology
option 'h1' is used when config option astdtmf=yes.
There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC
mode. Instead of setting the channel to HDLC mode it is set to
transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the
logging message is correct, but the if condition is not.
Make check the nodsp and hdlc flags.
JIRA ABE-2787
JIRA SWP-3437
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The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.
Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.
(closes issue #19221)
Reported by: kenner
JIRA SWP-3396
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and
changes to the transmit_* functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines
PRI early media won't ring.
And another way to pass early media. Don't indicate that there is inband
information present, just assume that the B channel is connected.
* Restore clearing the dialing flag Rx squelch unconditionally when a
PROCEEDING message comes in.
(closes issue #19268)
Reported by: tbsky
Patches:
issue19268_v1.8.patch uploaded by rmudgett (license 664)
Tested by: tbsky
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Consolidate the functions and add some debugging info. Allows to be
able to set a substate without explicitly knowing what the state is.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add the setsubstate_onhook to complete the initial substate handling
procedures. Added dumpsub(sub, forcehangup) which is the common way of
calling setsubstate_onhook. Dumpsub attempts to activate another sub
after setting the current one onhook.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines
Unable to pickup DAHDI/PRI call because call state is reported as DIALING.
The channel state is not updated to RINGING when an ALERTING message is
received. Regression caused when sig_pri.c (also sig_ss7.c) extracted
from chan_dahdi.c.
* Added missing channel state update to RINGING when the
AST_CONTROL_RINGING frame is queued for ISDN and SS7.
(closes issue #19257)
Reported by: alecdavis
Patches:
issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
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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
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines
Don't get early media for ISDN on outgoing calls.
It looks to be a long-standing misinterpretation of the progress indicator
ie values:
1 - Call is not end-to-end ISDN; further call progress information may be
available in-band.
8 - In-band information or an appropriate pattern is now available.
Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled
as early media probably because the meaning of the second half of it's
description was overlooked.
* Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or
PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path.
(closes issue #18868)
Reported by: isrl
Patches:
issue18868_19246_v1.8.patch uploaded by rmudgett (license 664)
Tested by: satish_lx
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No inband progress on PRI_EVENT_RINGING even if inband flag set.
My ISDN-PRI provider sends an ALERTING with "Inband information or
appropriate pattern now available", but Asterisk only generates and passes
the RING to the SIP extension, not the inband message. Unfortunately, the
inband message is not a ringback tone but a prompt that says the number is
not in service. The SIP extension then hears two rings and the call is
hungup which confuses the caller.
* Post an AST_CONTROL_PROGRESS as well as opening the media path if inband
audio is indicated with an ALERTING message.
(closes issue #19246)
Reported by: cristiandimache
Patches:
issue19246_v1.8.patch uploaded by rmudgett (license 664)
Tested by: cristiandimache
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a call is made to a line that already has a call and the device is
offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Long time coming, finally moving the hookstate from line to device.
This may fix some issues where a device has multiple lines. Previously
we had to run through all lines on a device to see if it was actually
onhook or not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
skinny_hold moved to setsubstate_hold and skinny_unhold integrated into
setsubstate_connected. Removed sub->onhold and replaced with
SUBSTATE_HOLD.
Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on
answering a SUBSTATE_RINGIN sub (was a typo).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mark Michelson
Jonathan Rose
Richard Mudgett
Joshua Colp
Stefan Schmidt
Matthew Jordan
Russell Bryant
Terry Wilson
Kevin P. Fleming
Walter Doekes
Alexandr Anikin
Kinsey Moore
Gregory Nietsky
David Vossel
Matthew Nicholson
Paul Belanger
Damien Wedhorn
Igor Goncharovskiy
Leif Madsen
Jason Parker
Olle Johansson
Tzafrir Cohen
Sean Bright
Mark Murawki
Tilghman Lesher
Brett Bryant
Alec L Davis