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This only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer...
(closes issue #18075)
Reported by: mdu113
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: twilson, mdu113
Review: https://reviewboard.asterisk.org/r/1005/
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possible use of an eswitch.
(closes issue #16228)
Reported by: jlaguilar
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ABE-2656
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@299220 f38db490-d61c-443f-a65b-d21fe96a405b
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commit.
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This change also plugs a few memory leaks that can occur when parking sip calls.
ABE-2656
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Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING
message is not received. The debug output shows that the DTMF begin event
is seen, but the DTMF end event is missing. When the DTMF begin happens,
the call is muted so we now have one way audio (until a DTMF end event is
somehow seen).
* Made set the proceeding flag when the PRI_EVENT_ANSWER event is
received.
* Made absorb the DTMF begin and DTMF end events if we are overlap dialing
and have not seen a PROCEEDING message.
* Added a debug message when absorbing a DTMF event.
JIRA SWP-2690
JIRA ABE-2697
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If a REGISTER request with a Call-ID matching an existing transaction is received
it was possible that the REGISTER request would overwrite the initreq of the
private structure. This info is used to generate messages for other responses in
the transaction. This patch ignores REGISTER requests that match non-REGISTER
transactions.
(closes issue #18051)
Reported by: eeman
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/1050/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297959 f38db490-d61c-443f-a65b-d21fe96a405b
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The changes here attempt to more strictly follow RFC 3261 section 10.3.
Basically the following will now cause a 400 Bad Response to be returned, if:
- multiple Contact headers are present with one set to expire all bindings ("*")
- wildcard parameter is specified for Contact without Expires header or Expires
header is not set to zero.
ABE-2442
ABE-2443
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481, not bad event.
If we answer 481 the subscription that we don't want will be cancelled.
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The problem here is only present when local channels are used with the MOH
passthru option as well as no optimization (/nm). I will describe the slightly
bizarre scenario that was used to test, where phones B and C are queue members:
Phone A dials into a queue with two members using local channels and the above
options. Phone B answers. Phone A blind transfers phone B into the same queue.
Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH.
In this scenario, the unhold frame that should have gotten to phone B never
arrived due to the masquerade from the blind transfer. This is usually fine
since app_queue manages the starting and stopping of MOH. However, with the
passthrough option enabled when app_queue attempts to stop MOH it tries to do
so on the local channel rather than the real channel. The easiest solution
was to just make sure to send an unhold frame during the transfer since it
wouldn't make sense to have MOH playing after a transfer anyway. This only
modifies SIP transfers, but the other transfers did not seem to be a problem.
If DTMF based transfers were a problem it might be okay to add ast_moh_stop
to finishup, but I didn't want to have to add that unless required.
ABE-2624
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This mainly deals with the problem of constructors on platforms where an
explicit constructor order cannot be specified (any system with gcc 4.2
or less). However, this is only a problem on those systems where we need
to initialize mutexes with a constructor, because we have other code that
also relies upon constructors, and we cannot specify that mutexes are
initialized first (and destroyed last).
There are two approaches to dealing with this issue, related to whether the
code exists in the core Asterisk binary or in a separate code module. In the
core case, constructors are run immediately upon load, and the file_versions
list mutex needs to be already initialized, as it is referenced in the first
constructor within each core source file. In this case, we use pthread_once
to ensure that the mutex is initialized immediately before it is used for the
first time. The only caveat is that the mutex is not ever destroyed, but
because this is the core, it makes no real difference; the only time when
destruction is safe would be just prior to process destruction, which takes
care of that anyway. And due to using pthread_once, the mutex will never be
reinitialized, which means only one structure has leaked at the end of the
process. Hence, it is not a problematic leak.
The second approach is to use the load_module and unload_module routines,
which, for obvious reasons, exist only in loadable modules. In this second
case, we don't have a problem with the constructors, but only with destructor
order, because mutexes can be destroyed before their final usage is employed.
However, we need the mutexes to still be destroyed, in certain scenarios: if
the module is unloaded prior to the process ending, it should be clean, with
no allocations by the module hanging around after that point in time.
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(closes issue #18398)
Reported by: pabelanger
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The FXS connected phone has to have CW/CID support to fail, as it will
send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal
phone with no CID never fails. Also the SIP phone does not hear MOH when
the CW call is answered.
The DTMF end frame is suppressed when the phone acknowledges the CW signal
for CID. The problem is the DTMF begin frame needs to be suppressed as
well. The DTMF begin frame is causing SIP to start sending the DTMF RTP
frames. Since the DTMF end frame is suppressed, SIP will not stop sending
those DTMF RTP packets.
* Suppress the DTMF begin and end frames when the channel driver is
looking for DTMF digits.
* Fixed a couple issues caused by not cleaning up the CID spill if you
answer the CW call while it is sending the CID spill.
* Fixed not sending CW/CID spill to the phone when the call is natively
bridged. (Fixed by not using native bridge if CW/CID is possible.)
* Suppress received audio when sending CW/CID spills. The other parties
involved do not need to hear the CW/CID spills and may be confused if the
CW call is for them.
(closes issue #18129)
Reported by: alecdavis
Patches:
issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
NOTE:
* v1.4 does not have the main problem fixed by suppressing the DTMF start
frames. The other three items fixed are relevant.
* If you really must restore native bridging between analog ports, you
need to disable CW/CID either by configuring chan_dahdi.conf
callwaitingcallerid=no or dialing *70 before dialing the number to
temporarily disable CW.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296165 f38db490-d61c-443f-a65b-d21fe96a405b
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This is not a perfect solution as headers that are joined via commas are not
detected. This is a parsing issue that to fix "correctly" would necessitate
a new SIP parser.
Review: https://reviewboard.asterisk.org/r/1019/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295628 f38db490-d61c-443f-a65b-d21fe96a405b
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seconds.
Asterisk is just whining too much with this message: "No D-channels
available! Using Primary channel XXX as D-channel anyway!".
Filtered the message so it only comes out once if there is no D channel
available without an intervening D channel available period.
(closes issue #17270)
Reported by: jmls
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294821 f38db490-d61c-443f-a65b-d21fe96a405b
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The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time. This
scenario has the potential to progress to the point of saturating a link just
from options packets. The fix was to ensure that the poke scheduler checks to
see if a peer is in the peer list before continuing to poke. The reason a peer
must be in the peer list to be able to properly manage an options dialog is
because otherwise the call pointer is lost when the peer is regenerated from
the database, which is how existing qualify dialogs are detected.
(closes issue #16382)
Reported by: lftsy
Patches:
bug16382-3.patch uploaded by jpeeler (license 325)
Tested by: zerohalo
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294688 f38db490-d61c-443f-a65b-d21fe96a405b
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scheduler entries if we get a new 491.
This prevents a scheduler entry from leaking if we receive a 491 response when one is pending. If a scheduler entry leaks, the pvt it is associated my get destroyed before the scheduler entry fires, and then memory corruption and crashes can occur when the scheduled reinvite attempts to access and modify the memory of the destroyed pvt.
ABE-2543
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294163 f38db490-d61c-443f-a65b-d21fe96a405b
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C answers.
All parties are analog FXS ports.
1) A calls B.
2) A flash hooks to call C.
3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback)
4) C answers
5) A continues to hear ringback during the 3-way call. (All parties can hear each other.)
* Fixed use of wrong variable in dahdi_bridge() that stopped ringback on
the wrong subchannel.
* Made several debug messages have more information.
A similar issue happens if B and C are SIP channels. B continues to hear
ringback. For some reason this only affects v1.8 and trunk.
* Don't start ringback on the real and 3-way subchannels when creating the
3-way conference. Removing this code is benign on v1.6.2 and earlier.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@293805 f38db490-d61c-443f-a65b-d21fe96a405b
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When setting to zero/"no", the numeric default was shown making it not obvious
the disabled setting was respected.
(closes issue #18123)
Reported by: zerohalo
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@293722 f38db490-d61c-443f-a65b-d21fe96a405b
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Change "Unable to get index, and nullok is not asserted" to "Unable to get
index for '<channel-name>' on channel <number> (<function>(), line
<number>)".
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chan_local does some dangerous things involving deadlock avoidance.
tech_pvt functions like hangup and queue_frame are provided with a
locked channel upon entry. Those functions are completely safe as
long as you don't attempt to give up that channel lock, but that is
impossible to guarantee due to the required deadlock avoidance necessary
to lock both the tech_pvt and both channels involved.
In the past, we have tried to account for this by doing things like
setting a "glare" flag that indicates what function should destroy the
pvt. This was used in local_hangup and local_queue_frame to decided
who should destroy the pvt if they collided in separate threads. I
have removed the need to do this by converting all chan_local tech_pvts
to astobj2. This means we can ref a pvt before deadlock avoidance
and not have to worry about that pvt possibly getting destroyed under
us. It also cleans up where we destroy the tech_pvt. The only unlink
from the tech_pvt container occurs in local_hangup now, which is where
it should occur.
Since there still may be thread collisions on some functions like
local_hangup after deadlock avoidance, I have added some checks to detect
those collisions and exit appropriately. I think this patch is going to
solve quite a bit of weirdness we have had with local channels in the past.
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There is a deadlock between dahdi_exception() and dahdi_indicate() for
analog ports. The call-waiting and three-way-calling feature can
experience deadlock if these features are trying to do something and an
event from the bridged channel happens at the same time.
Deadlock avoidance code added to obtain necessary channel locks before
attemting an operation with call-waiting and three-way-calling.
(closes issue #16847)
Reported by: shin-shoryuken
Patches:
issue_16847_v1.4.patch uploaded by rmudgett (license 664)
issue_16847_v1.6.2.patch uploaded by rmudgett (license 664)
issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/971/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@291643 f38db490-d61c-443f-a65b-d21fe96a405b
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This fixes a crash due to a hangup race condition.
ABE-2601
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The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
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This commit takes a simliar approach to 288112 and checks the dialplan to
determine the proper action for an incoming contact header as to whether or not
it should be decoded or not. sip_new was blindly always decoding the extension,
which also caused the outgoing contact header to be incorrect as well as failing
to match the encoded extension in the dialplan.
(closes issue #17892)
Reported by: wdoekes
Patches:
bug17892-1.patch uploaded by jpeeler (license 325)
Tested by: wdoekes
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@289699 f38db490-d61c-443f-a65b-d21fe96a405b
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On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed.
(closes issue #17950)
Reported by: schmidts
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/901/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@289622 f38db490-d61c-443f-a65b-d21fe96a405b
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This avoids noise on some Local channel situations where we don't use /n.
Thanks to Alec Davis for the suggestion.
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If a local channel gets bridged to itself, it becomes orphaned with no devices
left to actually tell it to hang up. This patch modifies local_fixup() to detect
this case and deny it.
Review: https://reviewboard.asterisk.org/r/934
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with a 500 Server Internal Error response.
ABE-2458
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(closes issue #16647)
Reported by: kwemheuer
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invitestate to INV_CANCEL like in sip_hangup.
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unlocked channel.
Near the beginning of schedule_delivery(), ast_bridged_channel() is called
on iaxs[fr->callno]->owner. However, the channel is not locked, which can
result in ast_bridged_channel() crashing should owner->tech change to a
technology that doesn't implement bridged_channel.
I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since
the owner lock was not held there either.
Converted the existing channel deadlock avoidance to use
iax2_lock_owner(). Using the new function simplified some awkward code.
In the process of fixing the locking on ast_bridged_channel(), I also
found a memory leak in socket_process() for v1.6.2 and v1.8. The local
struct variable ies.vars is not freed on early/abnormal function exits.
(closes issue #17919)
Reported by: rain
Patches:
issue17919_v1.4.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/926/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288192 f38db490-d61c-443f-a65b-d21fe96a405b
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When a phone sends an encoded URI for a subscription, the URI is not matched
with the actual hint that is in decoded format. For example, if we have an
extension with a hint that is named: "#5601" or "*5601", the subscription will
work fine if the phone subscribes with an already decoded URI, but when it's
decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the
correct hint.
(closes issue #17785)
Reported by: ramonpeek
Patches:
20100831__issue17785.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288112 f38db490-d61c-443f-a65b-d21fe96a405b
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Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either.
(closes issue #17551)
Reported by: ricardolandim
Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96)
Tested by: ricardolandim, mnicholson
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Review: https://reviewboard.asterisk.org/r/914/
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disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset,
the outgoing leg may not hang up or be delayed in hanging up. (Causes:
PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.)
Just hang up the call if the incoming call leg hangs up before connecting
for any reason. It makes no sense to send a BUSY or CONGESTION control
frame to the outgoing call leg under these circumstances.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286113 f38db490-d61c-443f-a65b-d21fe96a405b
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PROBLEM:
In chan_sip, and all the other channel drivers, it is common for
us to hold the tech_pvt lock while we ask the Asterisk core about
an extension and context. Every time we do this the locking
order becomes, (1. tech_pvt lock ---> 2. global context lock). In
chan_sip when a dialog subscribes to a hint, that locking order
is reversed in the extensionstate callback which will occur outside
of the channel_driver's monitor loop. So, on an extension state
update we have (1. global context lock ----> 2. tech_pvt lock).
Typically when we have to do a reversed locking order like this
we'd just do some sort of deadlock avoidance to fix the problem...
That will not work here. There are more locks involved here than
just the context and tech_pvt. Those are the two that are colliding,
but it is impossible to give up the context lock because the global
hints list lock MUST be held as well and we can not give that lock
up during the extensionstate callback traversal... The locking order
for the context and hints are (1. global context lock ----> 2.
hints list lock). Deadlock avoidance is not an option here.
SOLUTION:
The solution this patch implements is to queue the extension state updates
into a list and send the NOTIFY messages out during the do_monitor pvt
traversal. This clears out the problem of having to hold the context
lock before the tech_pvt lock entirely.
(closes issue #17888)
Reported by: zerohalo
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Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
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Otherwise, you could get issues with DTMF timeouts causing hangups.
(closes issue #17370)
Reported by: makoto
Patches:
channel-readstring-silence-generator.patch uploaded by makoto (license 38)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@285742 f38db490-d61c-443f-a65b-d21fe96a405b
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the user.
(closes issue #16661)
Reported by: jstapleton
Patches:
20100414__issue16661.diff.txt uploaded by tilghman (license 14)
20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: jstapleton
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@280811 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279053 f38db490-d61c-443f-a65b-d21fe96a405b
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signaling.
When you are using chan_dahdi ISDN signaling with immediate=yes and a call
comes in without a DNID then you get the DNID of a previous call.
Chan_dahdi does not touch the DNID field on a new call if it does not have
a DNID.
Made always copy the DNID from the new call.
The patches backport the relevant changes from trunk -r210387.
(closes issue #17568)
Reported by: wuwu
Patches:
issue17568_v1.4.patch uploaded by rmudgett (license 664)
issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278701 f38db490-d61c-443f-a65b-d21fe96a405b
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Prior to 1.4.26.3 T.38 negotiation worked properly, in the case
of the reporter.
(issue #16852)
Reported by: cfc
(closes issue #16705)
Reported by: mpiazzatnetbug
Patches:
issue16705_2.diff uploaded by ebroad (license 878)
Tested by: vrban, ebroad, c0rnoTa, samdell3
Review: https://reviewboard.asterisk.org/r/754/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277944 f38db490-d61c-443f-a65b-d21fe96a405b
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disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128
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