This does not change any behavior, but moves the frames from the local frame
list back to the channel read queue using an O(n) algorithm instead of O(n^2).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182553 f38db490-d61c-443f-a65b-d21fe96a405b
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182525 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines
Fix race in astdb
The underlying db1 implementation does not fully isolate the pages retrieved
from astdb, so the lock protecting accesses needs to be extended until the
copy from the shared memory structure is done.
(closes issue #14682)
Reported by: makoto
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182450 f38db490-d61c-443f-a65b-d21fe96a405b
* Fixed OPENR2 using an incorrect string value if the extension
delimiter is not present in the Dial() function. This was fixed for
SS7 and PRI in trunk -r172400.
* Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others.
* Removed trailing whitespace that appeared with OPENR2.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182408 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
Randomize IAX2 encryption padding
The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
Review: http://reviewboard.digium.com/r/193/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182282 f38db490-d61c-443f-a65b-d21fe96a405b
Previously, FILE() returned one less character than specified, due to the
terminating NULL. Both the offset and length parameters now behave
identically to the way variable substitution offsets and lengths also work.
(closes issue #14670)
Reported by: BMC
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182278 f38db490-d61c-443f-a65b-d21fe96a405b
For a channel that is not yet answered this API call will wait
until a voice frame is received on the channel before returning.
It does this by waiting for frames on the channel and reading them
in. The frames read in were not freed when they should have been.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182171 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines
Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF.
Dynamic features defined in the applicationmap section of features.conf allow
one to specify whether the caller, callee, or both have the ability to use the
feature. The documentation in the features.conf.sample file could be interpreted
to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the
calling channel in order to allow for the callee to be able to use the features
which he should have permission to use. However, the DYNAMIC_FEATURES variable
would only be read from the channel of the participant that pressed the DTMF
sequence to activate the feature. The result of this was that the callee was
unable to use dynamic features unless the dialplan writer had taken measures
to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel.
This commit changes the behavior of ast_feature_interpret to concatenate the
values of DYNAMIC_FEATURES from both parties involved in the bridge. The features
themselves determine who has permission to use them, so there is no reason to believe
that one side of the bridge could gain the ability to perform an action that they
should not have the ability to perform.
Kevin Fleming pointed out on the asterisk-users list that the typical way that this
was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel
so that the value would be inherited by the called channel. While this works, the
documentation alone is not enough to figure out why this is necessary for the callee
to be able to use dynamic features. In this particular case, changing the code to match
the documentation is safe, easy, and will generally make things easier for people for
future installations.
This bug was originally reported on the asterisk-users list by David Ruggles.
(closes issue #14657)
Reported by: mmichelson
Patches:
14657.patch uploaded by mmichelson (license 60)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182029 f38db490-d61c-443f-a65b-d21fe96a405b
The code responsible for sending the T38 reinvite did not check if an INVITE was
already being handled. This caused things to get confused and the call to fail.
The code now defers sending the T38 reinvite until the current INVITE is done being
handled.
(issue AST-191)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182022 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
Just recording the v1.4 change in trunk since it originally came from here.
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r181898 | rmudgett | 2009-03-12 20:19:29 -0500 (Thu, 12 Mar 2009) | 4 lines
Use the correct branch integrated property when generating the version string.
Copied the make_version file from Asterisk trunk.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181899 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines
Properly send a 487 on an INVITE we have not responded to if we receive a BYE.
If we receive an INVITE from an endpoint and then later receive a BYE from that
same endpoint before we have sent a final response for the INVITE, then we need
to respond to the INVITE with a 487.
There was logic in the code prior to this commit which seemed to exist solely to
handle this situation, but there was one condition in an if statement which
was incorrect. The only way we would send a 487 was if the sip_pvt had no owner
channel. This made no sense since we created the owner channel when we received
the INVITE, meaning that the majority of the time we would never send the 487.
The 487 being sent should not rely on whether we have created a channel. Its
delivery should be dependent on the current state of the initial INVITE transaction.
With this commit, that logic is now correctly in place.
(closes issue #14149)
Reported by: legranjl
Patches:
14149.patch uploaded by mmichelson (license 60)
Tested by: legranjl
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181769 f38db490-d61c-443f-a65b-d21fe96a405b
Previously, only 5 columns were displayed, and if a translation time exceeded
99,999 useconds, it would be displayed as 0, instead of its actual time.
(closes issue #14532)
Reported by: pj
Patches:
20090311__bug14532.diff.txt uploaded by tilghman (license 14)
Tested by: pj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181731 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines
Fix another scenario where depending on configuration the stream would not get read.
For custom commands we don't know whether the audio is coming from a stream or not
so we are going to have to read the data despite no channels.
(closes issue #14416)
Reported by: caspy
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r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines
Fix logic flaw in previous commit.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181661 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines
Fix issue with streaming MOH failing if nobody is listening.
When a music class is setup to actually provide music on hold
from a stream we need to constantly read audio from it since it
will constantly be providing audio. This is now done despite there
being no channels listening to it.
(closes issue #14416)
Reported by: caspy
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181656 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines
Make code that updates BRIDGEPEER variable thread-safe.
It is not safe to read the name field of an ast_channel without the channel
locked. This patch fixes some places in channel.c where this was being done,
and lead to crashes related to masquerades.
(closes issue #14623)
Reported by: guillecabeza
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181424 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
(closes issue #14607)
Reported by: stevenla
Tested by: dvossel
Review: http://reviewboard.digium.com/r/192/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181371 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines
Fix issue where an attended transfer could not be completed under a rare scenario.
When completing an attended transfer chan_sip does a check to make sure the extension
in the URI portion of the Refer-To header is a local valid extension. We don't actually
need to check this since we know for sure the other channel is already up and talking to
the extension. Some devices do not put the extension in the Refer-To header either, which
can cause the extension check to fail. We now no longer do this check if it is an attended
transfer.
(closes issue #14628)
Reported by: sverre
Patches:
14628.diff uploaded by file (license 11)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181345 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines
Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
When dtmfmode was set to auto the inband DTMF detector was not setup
on outgoing SIP calls. This caused inband DTMF detection to fail.
The inband DTMF detector is now setup for both dtmfmode inband and auto.
(closes issue #13713)
Reported by: makoto
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181296 f38db490-d61c-443f-a65b-d21fe96a405b
If trying to dial a non-existent queue, there would
be a segfault when attempting to access q->weight, even
though q was NULL. This problem was introduced during
the queue-reset merge and thus only affects trunk.
(closes issue #14643)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181244 f38db490-d61c-443f-a65b-d21fe96a405b
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue.
A few other issues were addressed:
- There were a few instances of functions improperly passing ast_free instead
of ast_free_ptr.
- Some clean up was done to avoid the debug macros intentionally being redefined.
(copied below from Kevin's commit, appreciate the help)
- disable astmm.h from doing anything when STANDALONE is defined, which is used
by the tools in the utils/ directory that use parts of Asterisk header files in
hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined.
(closes issue #13593)
Reported by: pj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181135 f38db490-d61c-443f-a65b-d21fe96a405b
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r181133 | jpeeler | 2009-03-10 22:25:04 -0500 (Tue, 10 Mar 2009) | 13 lines
Fix malloc debug macros to work properly with h323.
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue. Because using the ast prefix calls are
a better choice, ast_free_ptr is the new wrapper for free to pass to functions.
Also, a little bit of clean up was done to avoid the debug macros intentionally
being redefined.
(closes issue #13593)
Reported by: pj
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181134 f38db490-d61c-443f-a65b-d21fe96a405b
russell