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r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) | 4 lines
Avoid access of uninitialized memory. This caused a bunch of crashes for me
while doing load testing of development branch where I'm working on some
performance improvements.
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r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines
Fix a deadlock involving channel autoservice and chan_local that was debugged
and fixed by mmichelson and me.
We observed a system that had a bunch of threads stuck in ast_autoservice_stop().
The reason these threads were waiting around is because this function waits to
ensure that the channel list in the autoservice thread gets rebuilt before the
stop() function returns. However, the autoservice thread was also locked, so
the autoservice channel list was never getting rebuilt.
The autoservice thread was stuck waiting for the channel lock on a local channel.
However, the local channel was locked by a thread that was stuck in the autoservice
stop function.
It turned out that the issue came down to the local_queue_frame() function in
chan_local. This function assumed that one of the channels passed in as an
argument was locked when called. However, that was not always the case. There
were multiple cases in which this channel was not locked when the function was
called. We fixed up chan_local to indicate to this function whether this channel
was locked or not. The previous assumption had caused local_queue_frame() to
improperly return with the channel locked, where it would then never get unlocked.
(closes issue #12584)
(related to issue #12603)
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marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
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r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines
Merged revisions 115564 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines
Fix a race condition that bbryant just found while doing some IAX2 testing.
He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes,
however, the audio was extremely choppy. We looked at a packet trace and saw
a storm of INVAL and VNAK frames being sent from one box to another.
It turned out that what had happened was that one box tried to send a CONTROL
frame before the 3 way handshake had completed. So, that frame did not include
the destination call number, because it didn't have it yet. Part of our recent
work for security issues included an additional check to ensure that frames that
are supposed to include the destination call number have the correct one. This
caused the frame to be rejected with an INVAL. The frame would get retransmitted
for forever, rejected every time ...
This race condition exists in all versions that got the security changes,
in theory. However, it is really only likely that this would cause a problem in
Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_
beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing
all versions that could potentially be affected by the introduced race condition.
These changes are what bbryant and I came up with to fix the issue. Instead of
simply dropping control frames that get sent before the handshake is complete,
the code attempts to wait a little while, since in most cases, the handshake
will complete very quickly. If it doesn't complete after yielding for a little
while, then the frame gets dropped.
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r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines
Avoid putting opaque="" in Digest authentication. This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)
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dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified.
(closes issue #12536)
Reported by: bjm
Patches:
12536-dringXrange.diff uploaded by qwell (license 4)
Tested by: bjm
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r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
These changes address a critical performance issue introduced in the latest
release. The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls. On a small embedded platform, it would not be
able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels. Ouch.
These changes address some performance issues of the find_callno() function
that have bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call. This involved a mutex lock and unlock for each call number
checked. So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks. Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.
A second container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number. Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.
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r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines
use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined
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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines
Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.
(closes issue #12513)
Reported by: mneuhauser
Patches:
asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)
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(closes issue #12496)
Reported by: daniele
Patches:
misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
Tested by: daniele
Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision.
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r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line
added fix from #8899
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r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines
If the dial string passed to the call channel callback does not indicate an
extension, then consider the extension on the channel before falling back
to the default.
(closes issue #12479)
Reported by: darren1713
Patches:
exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116)
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Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
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r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines
Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon
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r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines
use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10)
(closes issue #12456)
Reported by: fnordian
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r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines
The call_token on the pvt can occasionally be NULL, causing a crash.
If it is NULL, we can skip this channel, since it can't the one we're looking for.
(closes issue #9299)
Reported by: vazir
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r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines
Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen.
Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed.
(issue #12400)
Reported by: ztel
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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines
We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.
(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann
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r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines
If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario.
(closes issue #12385)
Reported by: viraptor
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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines
If Asterisk receives a 488 on an INVITE (not a reinvite), then
we should not send a BYE.
(closes issue #12392)
Reported by: fnordian
Patches:
chan_sip.patch uploaded by fnordian (license 110) with small modification from me
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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines
Merged revisions 113012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes.
I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short.
Instead of freeing the struct and nulling the pointer, it now just resets it, because this
ast_str is expected by the calling routine to still be there after handle_request_do() returns.
This appears to fix the crash. I assume that it was introduced with ast_str's being adopted. It's a subtle and easy-to-miss sort of problem.
I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well;
no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do...
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r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines
Fix the testing of the "res" variable so that it is more logically correct and
makes the correct warning and debug messages print.
(closes issue #12361)
Reported by: one47
Patches:
chan_zap_deferred_digit.patch uploaded by one47 (license 23)
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that valgrind no longer complains and that calls do complete correctly.
The fix is along the same lines as before: Make sure the final null terminator gets copied
into the new sip_request's data pointer. Without it, parse_request will read and potentially
write past the end of the string, causing potential crashes.
(closes issue #12284...for real this time!)
reported by falves11
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for the string to be copied into. This resulted in parse_request reading invalid
memory beyond the end of the string, and in some cases led to crashes. Thanks
to falves11 for providing the valgrind output which led to the closure of this issue.
(closes issue #12284)
Reported by: falves11
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Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.
2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.
3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.
4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.
5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.
(closes issue #11968)
Reported by: dimas
Patches:
v2-11968-dsp.patch uploaded by dimas (license 88)
v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell
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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
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r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines
When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.
So this is a revert of a revert...sort of.
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SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.
(closes issue #12279)
Reported by: rjain
Patches:
chan_sip.c.diff uploaded by rjain (license 226)
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r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines
This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk
so that all scheduler functions are fixed at once.
I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.
(closes issue #12272)
Reported by: qq12345
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r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines
Merged revisions 110335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines
Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address
structure that a background thread continuously updates. However, in these cases,
a stack variable was passed. That means that the dnsmgr thread would be continuously
writing to bogus memory.
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by Juggie while attempting to load chan_zap. Apparently this would happen
if an error were encountered while trying to process zapata.conf.
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actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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when dialing a channel that does not provide progress (analog ZAP lines)
The phone does handle the double update on calls to channels that do
provide progress and wont insert duplicate items
(closes issue #12239)
Reported by: DEA
Patches:
chan_skinny-call-log.txt uploaded by DEA (license 3)
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r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines
Fix a race condition in the SIP packet scheduler which could cause a crash.
chan_sip uses the scheduler API in order to schedule retransmission of reliable
packets (such as INVITES). If a retransmission of a packet is occurring, then the
packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if
a response is received from the packet as previously transmitted, then when we
ACK the response, we will remove the packet from the scheduler and free the packet.
The problem is that both the ACK function and retrans_pkt attempt to acquire the
same lock at the beginning of the function call. This means that if the ACK function
acquires the lock first, then it will free the packet which retrans_pkt is about to
read from and write to. The result is a crash.
The solution:
1. If the ACK function fails to remove the packet from the scheduler and the retransmit
id of the packet is not -1 (meaning that we have not reached the maximum number of
retransmissions) then release the lock and yield so that retrans_pkt may acquire the
lock and operate.
2. Make absolutely certain that the ACK function does not recursively lock the lock in
question. If it does, then releasing the lock will do no good, since retrans_pkt will
still be unable to acquire the lock.
(closes issue #12098)
Reported by: wegbert
(closes issue #12089)
Reported by: PTorres
Patches:
12098-putnopvutv3.patch uploaded by putnopvut (license 60)
Tested by: jvandal
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r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines
Make a tweak that gets the LEDs on polycom phones to blink when an extension that
has been subscribed to goes on hold. Otherwise, they just stay on like it does
when an extension is in use.
(closes issue #11263)
Reported by: russell
Patches:
notify_hold.rev1.txt uploaded by russell (license 2)
Tested by: russell
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This set of changes removes the hard coded maximum packet size of 4kB from chan_sip.
It now starts by allocating 1kB, and growing the buffer as needed to accommodate large
packets.
(closes issue #8556, reported by mikma, patch by jamesgolovich)
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r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines
Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip.
The scheduler callback will always return 0. This means that this id
is never rescheduled, so it makes no sense to loop trying to delete
the id from the scheduler queue. If we fail to remove the item from the
queue once, it will fail every single time.
(Yes I realize that in this case, the macro would exit early because the
id is set to -1 in the callback, but it still makes no sense to use
that macro in favor of calling ast_sched_del once and being done with it)
This is the first of potentially several such fixes.
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(closes issue #9503)
Reported by: tzafrir
Patches:
fix_cleanups uploaded by tzafrir (license 46)
zapata_sections uploaded by tzafrir (license 46)
skipchannel_options uploaded by tzafrir (license 46)
conf_sample uploaded by tzafrir (license 46)
patches updated by me to better conform to coding guidelines and fix some problems
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r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines
if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP
closes issue #11475
Reported by: andrebarbosa
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r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines
don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"
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Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
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r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines
when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
(closes issue #11917)
Reported by: mavetju
Tested by: mavetju
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r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines
Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
(closes issue #12140)
Reported by: slavon
Patches:
sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)
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- fix a spot where a lock wouldn't get unlocked in an error condition
- call ast_mutex_destroy() on the lock before freeing its memory
(related to issue #11972)
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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
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automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6 lines
Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device.
(closes issue #9044)
Reported by: queuetue
Patches:
sip-gui-friend.diff uploaded by qwell (license 4)
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r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6 lines
If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog.
(closes issue #10727)
Reported by: s0l4rb03
Patches:
10727-2.diff uploaded by file (license 11)
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r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6 lines
Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank.
(closes issue #12061)
Reported by: flefoll
Patches:
chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244)
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r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19 Feb 2008) | 8 lines
Account for the fact that the "other" channel can disappear while the local pvt
is not locked.
(fixes a problem introduced in rev 100581)
(closes issue #12012)
Reported by: stevedavies
Patch by me
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Reported by: julianjm
Patches:
chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm (license 99)
Patch fixes problem of device state incorrectly reporting idle before PBX answers incoming call on FXO channel. Device status is updated now during new channel creation.
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r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines
When a SIP channel is being auto-destroyed, it's possible for it to still be
in bridge code. When that happens, we crash. Delay the RTP destruction until
the bridge is ended.
(closes issue #11960)
Reported by: norman
Patches:
20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: norman
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r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb 2008) | 10 lines
Fix a linked list corruption that under the right circumstances
could lead to a looped list, meaning it will traverse forever.
(closes issue #11818)
Reported by: michael-fig
Patches:
11818.patch uploaded by putnopvut (license 60)
Tested by: michael-fig
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(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
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r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8 lines
Handle ACK and CANCEL in an invite transaction - even if we get INFO transactions during the actual call setup.
(closes issue #10567)
Reported by: jacksch
Tested by: oej
Patch by: oej inspired by suggestions from neutrino88 in the bug tracker
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r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) | 5 lines
Change the SDP_SAMPLE_RATE macro. It turns out that even though G.722 is 16 kHz,
it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed
to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but
people follow it anyway, because it's the spec ...)
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r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31 Jan 2008) | 8 lines
Add some more sanity checking on IAX2 dial strings for the case that no peer
or hostname was provided, which is the one part of the dial string that is
absolutely required. If it's not there, bail out.
(closes issue #11897)
Reported by sokhapkin
Patch by me
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r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008) | 2 lines
Add missing locking to the find_agent() function.
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r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31 Jan 2008) | 3 lines
Move the locking from find_agent() into the agent dialplan function handler to
ensure that the agent doesn't disappear while we're looking at it.
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broke realtime configurations which still used the "username" field. This was taken
care of properly when reading from realtime but was not handled properly when updating
a realtime peer. This change also adds a deprecation NOTICE CLI message that will print
if using the deprecated "username" field.
(closes issue #11880)
Reported by: cabal95
Patches:
11880.patch uploaded by putnopvut (license 60)
Tested by: cabal95
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r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008) | 5 lines
For some reason, the use of this strdupa() is leading to memory corruption on
freebsd sparc64. This trivial workaround fixes it.
(closes issue #10300, closes issue #11857, reported by mattias04 and Home-of-the-Brave)
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r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines
Make some deadlock related fixes. These bugs were discovered and reported
internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
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that was just merged from 1.4, so this is a changeover to those APIs to use the
macro versions, so that we properly detect errors from ast_sched_del, instead
of simply ignoring the return values.
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r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines
When deleting a task from the scheduler, ignoring the return value could
possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9 lines
Make sure we don't cancel destruction on calls in CANCEL state, even if we
get 183 while waiting for answer on our CANCEL.
(issue #11736)
Reported by: MVF
Patches:
bug11736.txt uploaded by oej (license 306)
Tested by: MVF
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r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 lines
Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language
over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old
head to avoid too heavy memory allocations on some systems.
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as a channel variable BRIDGEPVTCALLID
This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.
The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.
Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.
Inspired by (issue #11816)
Reported by: ctooley
Patch by oej
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r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 Jan 2008) | 12 lines
Fixing an issue wherein monitoring local channels was not possible. During a channel
masquerade, the monitors on the two channels involved are swapped. In 99% of the cases
this results in the desired effect. However, if monitoring a local channel, this caused
the monitor which was on the local channel to get moved onto a channel which is immediately
hung up after the masquerade has completed. By swapping the monitors prior to the masquerade,
we avoid the problem by tricking the masquerade into placing the monitor back onto the channel
where we want it.
During the investigation of the issue, the channel's monitor was the only thing that was swapped
in such a manner which did not make sense to have done. All other variable swapping made sense.
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is currently active for the Asterisk CLI, or to set it. Also, knock multiple device
support off of the to-do list.
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- Add support for multiple devices. All devices are configured in console.conf.
- Add "console list devices" CLI command to show configured devices. Also, changed
the old "list devices" to be "list available", which queries PortAudio for all
audio devices that are available for use.
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines
Have IAX2 optimize the codec translation path just like chan_sip does it. If
the caller's codec is in our codec list, move it to the top to avoid transcoding.
(closes issue #10500)
Reported by: stevedavies
Patches:
iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh
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This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Tested by: jtodd, rjain, loloski
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines
Fix a deadlock in chan_local in local_hangup. There was contention because
the local_pvt was held and it was attempting to lock a channel, which is the
incorrect locking order.
(closes issue #11730)
Reported by: UDI-Doug
Patches:
11730.patch uploaded by putnopvut (license 60)
Tested by: UDI-Doug
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r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines
Don't drop the old record route information when dealing with packets related to a reinvite.
(closes issue #11545)
Reported by: kebl0155
Patches:
reinvite-patch.txt uploaded by kebl0155 (license 356)
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r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines
Change a buffer in check_auth() to be a thread local dynamically allocated
buffer, instead of a massive buffer on the stack. This fixes a crash reported
by Qwell due to running out of stack space when building with LOW_MEMORY defined.
On a very related note, the usage of BUFSIZ in various places in chan_sip is
arbitrary and careless. BUFSIZ is a system specific define. On my machine,
it is 8192, but by definition (according to google) could be as small as 256.
So, this buffer in check_auth was 16 kB. We don't even support SIP messages
larger than 4 kB! Further usage of this define should be avoided, unless it
is used in the proper context.
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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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Note: NoAnswer support is currently not implemented, as it would take a
significant amount of work to figure out how to do correctly.
Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines
1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running. Don't just ignore it.
(Closes issue #11698)
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1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
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- support scrolling of message window;
- simplify the code for creating a message window,
and try it using a second one in the top of
the keypad (where we echo the dialed number).
The 'skin' that supports these two windows will be
committed separately.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines
Set the caller id within the gtalk_alloc function.
As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.
Closes issue #11549.
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a number to dial in the 'message' area under the
keypad.
Now you can make calls using the keypad as a regular phone
(or the keyboard for chars not present on the keypad)
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commands so you can start and stop the gui even outside
of a call. This is convenient for testing, and also for
using the keypad to pick up a call, and to dial a number
(the latter not yet implemented, but should be close).
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The main code to implement the textarea is in console_board.c,
and uses a simple png image with the font, blitting characters
on the designated areas of the main screen.
Additionally we provide some annotations in the image used
as a skin to indicate which areas are used for text messages.
(images will be committed separately).
At the moment the dialog area is only used to display a running
counter, just as a proof of concept.
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This way it can contain additional elements (e.g. fonts, buttons,
widgets) without having to use a zillion files to store them.
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revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines
Make use of the temporary channel pointer while the pvt is unlocked.
(closes issue #11675)
Reported by: flefoll
Patches:
chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244)
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remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
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of a hack. It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call. But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.
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(this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working)
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to an 8 kHz endpoint, then codec_resample will automatically be used to properly
resample the audio before sending it to/from chan_console.
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Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface. It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.
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to add more entries. This required moving struct grab_desc to the common
header, and adding an entry in the Makefile.
On passing, cleanup some comments and file headers (some are still missing).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | 6 lines
Remove duplicate increment of the header count in the add_header() function.
(closes issue #11648)
Reported by: makoto
Patch provided by sergee, committed patch by me, inspired by comments from putnopvut
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r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 lines
Include types.h in chan_h323 as without it it can not be compiled on some operating systems like FreeBSD to name one.
(closes issue #11585)
Reported by: sobomax
Patches:
chan_h323.c.diff uploaded by sobomax (license 359)
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SDL is also detected at runtime).
Now we should be able to stream video even without a rendering device
(useful for remote monitoring).
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in the peers container after a reload. Somehow, this bug doesn't exist in 1.4 ...
(closes issue #11626)
(reported by pnlarsson, additional info from mvanbaak, fixed by me)
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are in separate files (still #include'd because of tangling in the data
structures, but this is going to be cleaned up).
The video grabbing functions still need to be moved to a separate file.
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With this change we can do
setenv SDL_VIDEODRIVER aalib
and output to an ascii window (which is still in an X11 window).
If you also do
unsetenv DISPLAY
then the output goes into the main asterisk window, unfortunately
it interferes with the normal output so you don't see much.
In any case, i don't think we are very far away from having a working
xterm videophone!
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Russell Bryant
Joshua Colp
Matthew Fredrickson
Sean Bright
Jim Dixon
Jeff Peeler
Olle Johansson
Mark Michelson
Brett Bryant
Tilghman Lesher
Jason Parker
Kevin P. Fleming
Michiel van Baak
Steve Murphy
Terry Wilson
Philippe Sultan
Doug Bailey
James Golovich
Christian Richter
Luigi Rizzo