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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.37-rc1@287633 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.37-rc1@287632 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.37-rc1@287631 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.37-rc1@287630 f38db490-d61c-443f-a65b-d21fe96a405b
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(related to issue #17928)
Reported by: mdu113
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287555 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17891)
reported, solved and tested by oej
Review: https://reviewboard.asterisk.org/r/869/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287469 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16893)
Reported by: haakon
Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287386 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_hint_state_changed().
(related to issue #17928)
Reported by: mdu113
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287307 f38db490-d61c-443f-a65b-d21fe96a405b
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isn't there.
Headers were taken from trunk.
(closes issue #17958)
Reported by: javyer
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287197 f38db490-d61c-443f-a65b-d21fe96a405b
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length strings.
(closes issue #17928)
Reported by: mdu113
Patches:
20100831__issue17928.diff.txt uploaded by tilghman (license 14)
Tested by: mdu113
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287118 f38db490-d61c-443f-a65b-d21fe96a405b
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value.
(closes issue #17900)
Reported by: under
Patches:
core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@287114 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286956 f38db490-d61c-443f-a65b-d21fe96a405b
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Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286941 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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286756 f38db490-d61c-443f-a65b-d21fe96a405b
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Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework.
ABE-2473
(related to issue #2342)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286679 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17983)
Reported by: oej
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286381 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/911/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286267 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286222 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/914/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286114 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286070 f38db490-d61c-443f-a65b-d21fe96a405b
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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/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286059 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286023 f38db490-d61c-443f-a65b-d21fe96a405b
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This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@285889 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17688)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347)
Tested by: mkeuter, pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@285817 f38db490-d61c-443f-a65b-d21fe96a405b
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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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A translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed up.
(closes issue #17630)
Reported by: manvirr
Patches:
encoder_fix.diff uploaded by dvossel (license 671)
Tested by: manvirr, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@280448 f38db490-d61c-443f-a65b-d21fe96a405b
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conference, then masquaraded away.
ABE-2422
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@280341 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@280088 f38db490-d61c-443f-a65b-d21fe96a405b
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If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279945 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279346 f38db490-d61c-443f-a65b-d21fe96a405b
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menuselect doesn't get confused:
Unknown value '' found in build_tools/menuselect-deps for DAHDI_TRANSCODE
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279344 f38db490-d61c-443f-a65b-d21fe96a405b
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The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable. The two variables are not equivalent if the call_forward string
included a channel technology specifier. e.g., SIP/200
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279206 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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will fail to load, even if it links.
(issue #17679)
Reported by: am
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278984 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17080)
Reported by: sybasesql
Patches:
20100721__issue17080.diff.txt uploaded by tilghman (license 14)
Tested by: sybasesql
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278981 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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If a channel involved in a bridge was using SLIN audio, then translation
paths were not guaranteed to be set up properly since in all likelihood
the number of translation steps was only 1.
This patch enforces the transcode_via_slin behavior if transcode_via_slin
or generic_plc is enabled and one of the formats to make compatible is
SLIN.
AST-352
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278618 f38db490-d61c-443f-a65b-d21fe96a405b
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expunge does not cause deletion of the wrong message.
(closes issue #16350)
Reported by: noahisaac
Patches:
20100623__issue16350.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278261 f38db490-d61c-443f-a65b-d21fe96a405b
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(Fixes ABE-2110)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278167 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16506)
Reported by: nik600
Patches:
20100629__issue16506.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278023 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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completed.
A crash could occur if the extension is picked up while the parking extension is
being announced. Testing pu->notquiteyet while searching for a parked extension
resolves this crash.
(ABE-2418)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277906 f38db490-d61c-443f-a65b-d21fe96a405b
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only uclinux that does not.
(closes issue #17616)
Reported by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277738 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended
transfer, ast_bridge_call() is called for a second bridge on the same channel,
and it clears that flag, which still needs to get set for when the original
ast_bridge_call() gets control back and checks it.
Review: https://reviewboard.asterisk.org/r/741
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277625 f38db490-d61c-443f-a65b-d21fe96a405b
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semicolon as an encoded value.
(closes issue #17369)
Reported by: gkservice
Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277568 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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277497 f38db490-d61c-443f-a65b-d21fe96a405b
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During a reload, the priexclusive and outsignalling parameters are not
read in from the config file as intended. Unfortunately, they get set to
defaults as a result. This patch makes sure that they do not get set to
defaults during a reload.
(closes issue #17441)
Reported by: mtryfoss
Patches:
issue17441_v1.4.patch uploaded by rmudgett (license 664)
issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
issue17441_trunk.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277419 f38db490-d61c-443f-a65b-d21fe96a405b
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AST_EXTENSION_NOT_INUSE.
(closes issue #16035)
Reported by: francesco_r
Patches:
pbx.c.patch uploaded by viniciusfontes (license 978)
Tested by: francesco_r, agx, lawbar
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277327 f38db490-d61c-443f-a65b-d21fe96a405b
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