Joshua Colp [Thu, 18 Dec 2008 17:11:42 +0000 (17:11 +0000)]
Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
(closes issue #13545)
Reported by: davidw
Mark Michelson [Tue, 16 Dec 2008 23:04:27 +0000 (23:04 +0000)]
After looking through SIP registration code most of the day, this
is one of the few things I could find that was just plain wrong.
Even though it probably isn't possible for it to happen, it seems weird
to have code that checks if a pointer is NULL and then immediately dereferences
that pointer if it was NULL.
Russell Bryant [Tue, 16 Dec 2008 21:10:44 +0000 (21:10 +0000)]
Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.
This is a bug I noticed while looking at the code for app_macro. This return code
means that another thread has assumed ownership of the channel and it can no longer
be touched. (I hate this return code with a passion, by the way.)
Russell Bryant [Tue, 16 Dec 2008 20:35:25 +0000 (20:35 +0000)]
Add "restart gracefully" to the AMI blacklist of CLI commands.
"module unload" was already identified as a command that can not be used
from the AMI. "restart gracefully" effectively unloads all modules, and will
run in to the same problems.
Russell Bryant [Tue, 16 Dec 2008 17:06:29 +0000 (17:06 +0000)]
Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
One issue was that the ast_mutex_* API was being used within the context of the
thread local data destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all. This led to a memory
leak.
Another issue was an invalid argument being provided to the the object_add
API call.
(closes issue #13678)
Reported by: ys
Tested by: russell
Russell Bryant [Tue, 16 Dec 2008 15:56:37 +0000 (15:56 +0000)]
Fix a memory leak related to the use of the "setvar" configuration option.
The problem was that these variables were being appended to the list of vars
on the sip_pvt every time a re-registration or re-subscription came in.
Since it's just a waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying the vars.
(closes issue #14037)
Reported by: marvinek
Tested by: russell
Steve Murphy [Tue, 16 Dec 2008 15:15:58 +0000 (15:15 +0000)]
I added a sentence to clarify why - and ' ' are ignored in patterns
as per bug 14076. Leif says he'll put some stuff about it in the
extensions.conf sample, etc.
Joshua Colp [Mon, 15 Dec 2008 18:11:21 +0000 (18:11 +0000)]
Do not try to unlock a non-existant channel if the transfer fails.
(closes issue #13800)
Reported by: dwagner
Patches:
asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608)
Russell Bryant [Mon, 15 Dec 2008 14:31:37 +0000 (14:31 +0000)]
Handle a case where a call can be bridged to a channel that is still ringing.
The issue that was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once the parked
call got taken out of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing indication. This patch
fixes this case so that the caller receives ringback once it comes out of
parking until the other side answers.
The fixes are:
- Make sure we remember that a channel was an outgoing channel when doing
a masquerade. This prevents an erroneous ast_answer() call on the channel,
which causes a bogus 200 OK to be sent in the case of SIP.
- Add some additional comments to explain related parts of code.
- Update the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that are events
or commands should be ignored.
- When a bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still ringing.
- Rework ast_indicate_data() a bit for the sake of readability.
Tilghman Lesher [Sat, 13 Dec 2008 23:22:02 +0000 (23:22 +0000)]
Change the default calldurationlimit from the special value 0 to -1, so we
can better detect an exceptional case. This follows on to the changes made
in revision 156386. Related to issue #13851.
(closes issue #13974)
Reported by: paradise
Patches:
20081208__bug13974.diff.txt uploaded by Corydon76 (license 14)
Tested by: file, blitzrage, ZX81
Tilghman Lesher [Fri, 12 Dec 2008 22:03:10 +0000 (22:03 +0000)]
Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a
pointer inside editline to look back to asterisk.c, so others don't spend
as much time as I did looking (in the wrong place) for the appropriate
function.
Reported by: ZX81, via the #asterisk-users channel
Fixed by: me (license 14)
Russell Bryant [Fri, 12 Dec 2008 13:44:08 +0000 (13:44 +0000)]
Resolve issues that could cause DTMF to be processed out of order.
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
Tilghman Lesher [Thu, 11 Dec 2008 23:35:55 +0000 (23:35 +0000)]
When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal
is messed up. By intercepting those events with a signal handler in the remote
console, we can avoid those issues.
(closes issue #13464)
Reported by: tzafrir
Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
Russell Bryant [Thu, 11 Dec 2008 16:54:51 +0000 (16:54 +0000)]
Fix an issue that made it so you could only have a single caller executing
a custom feature at a time. This was especially problematic when custom
features ran for any appreciable amount of time.
The fix turned out to be quite simple. The dynamic features are now stored
in a read/write list instead of a list using a mutex.
(closes issue #13478)
Reported by: neutrino88
Fix suggested by file
Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams.
Joshua Colp [Wed, 10 Dec 2008 19:01:17 +0000 (19:01 +0000)]
Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
(closes issue #12560)
Reported by: vsauer
Patches:
patch001.diff uploaded by ramonpeek (license 266)
Joshua Colp [Wed, 10 Dec 2008 17:50:43 +0000 (17:50 +0000)]
When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
(closes issue #13599)
Reported by: hjourdain
Patches:
chan_sip.c.diff uploaded by hjourdain (license 583)
I checked, and I added a mod to the trunk version
of Asterisk that would make it 8-bit transparent
on 27 Nov 2007, but I made no such updates to
1.4. My best guess is that 1.4 was released, and
it was not appropriate to commit an enhancement.
But I'm going to add the same fixes to 1.4 now,
for the following reasons:
1. wetwired is correct; 1.4 is **mostly** 8-bit
transparent now. This is because the lexical
token forming rules use . in most 'word'
state continuances. It's just the beginning
of a 'word' that is picky.
2. Accepting 8-bit chars in some places and
not others leads to bug reports like this.
This patch was originally uploaded to issue #13783 by
jamessan. While the issue was closed for other reasons, this
patch is valid and fixes a separate problem, and is thus
being committed.
Joshua Colp [Wed, 10 Dec 2008 16:05:29 +0000 (16:05 +0000)]
Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
(closes issue #12983)
Reported by: vt
Patches:
dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
Russell Bryant [Tue, 9 Dec 2008 22:17:39 +0000 (22:17 +0000)]
Remove the test_for_thread_safety() function completely.
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
Tilghman Lesher [Tue, 9 Dec 2008 21:53:25 +0000 (21:53 +0000)]
We appear to have documented tz= in the [general] section of voicemail.conf,
without actually having implemented it. Oops.
(Reported by Olivier on the -users list)
Joshua Colp [Tue, 9 Dec 2008 21:14:29 +0000 (21:14 +0000)]
Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing.
(closes issue #14005)
Reported by: ddl
Russell Bryant [Tue, 9 Dec 2008 20:57:35 +0000 (20:57 +0000)]
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.
Steve Murphy [Tue, 9 Dec 2008 20:20:54 +0000 (20:20 +0000)]
In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this.
Joshua Colp [Tue, 9 Dec 2008 19:47:07 +0000 (19:47 +0000)]
Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
(closes issue #13209)
Reported by: ip-rob
Patches:
13209.diff uploaded by file (license 11)
Tested by: ip-rob, bujones
This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.
32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.
Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.
I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.
And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.
Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.
Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!
Russell Bryant [Tue, 9 Dec 2008 14:52:25 +0000 (14:52 +0000)]
Fix a problem with GROUP() settings on a masquerade.
The previous code carried over group settings from the old channel to the new
one. However, it did nothing with the group settings that were already on the
new channel. This patch removes all group settings that already existed on the
new channel.
I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category. However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Mark Michelson [Thu, 4 Dec 2008 16:44:18 +0000 (16:44 +0000)]
Fix a callerid parsing issue. If someone formatted callerid like the
following: "name <number>" (including the quotation marks), then the parts
would be parsed as
name: "name
number: number
This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.
Tilghman Lesher [Wed, 3 Dec 2008 17:55:12 +0000 (17:55 +0000)]
During bridge code, the channel bridge may return a retry code, if a transfer
was initiated but not yet completed. If the bridge is immediately retried,
then we may send a storm of TXREQ packets, even though the first set is sent
reliably (retransmitted). Fixes AST-137.
Tilghman Lesher [Wed, 3 Dec 2008 17:34:34 +0000 (17:34 +0000)]
If an entry is added to the directory during a scan when another entry expires,
then that new entry will not be processed promptly, but must wait for either a
future entry to start or a current entry's retry to occur. If no other entries
exist in the directory (other than the new entries) when a bunch expire, then
the new entries must wait until another new entry is added to be processed.
This was a rather weird race condition, really. Fixes AST-147.
Tilghman Lesher [Wed, 3 Dec 2008 16:58:34 +0000 (16:58 +0000)]
Don't start scanning the directory until all modules are loaded, because some
required modules (channels, apps, functions) may not yet be in memory yet.
Fixes AST-149.
Tilghman Lesher [Wed, 3 Dec 2008 14:09:35 +0000 (14:09 +0000)]
Jon Bonilla (Manwe) pointed out on the -dev list:
"I guess that having only ip-phones in mind is not a good approach. Since it is
possible to have a sip proxy connected to asterisk we could receive a 407
(unauthorized) or 483 (too many hops) as response and dialog ending would not be
a good behavior."
So modified.
Tilghman Lesher [Tue, 2 Dec 2008 17:42:09 +0000 (17:42 +0000)]
When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion
fails, and the resulting integer is garbage. Thus, we must initialize the
integer and check it afterwards for success.
(closes issue #14000)
Reported by: folke
Patches:
asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
Russell Bryant [Mon, 1 Dec 2008 17:27:30 +0000 (17:27 +0000)]
Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they
both have the potential to send control frames in the middle of call setup. We
have to wait until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will consider it
invalid, and we'll get stuck in an INVAL/VNAK storm.
Kevin P. Fleming [Wed, 26 Nov 2008 18:36:24 +0000 (18:36 +0000)]
simplify (and slightly bug-fix) the recent developer-oriented COMPILE_DOUBLE mode
add channels/busy.h and channels/ringtone.h to the repository instead of generating them repeatedtly; most users do not change the settings to build them, but the Makefile rules are still there if they wish to do so
ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
Jeff Peeler [Sat, 22 Nov 2008 00:04:36 +0000 (00:04 +0000)]
(closes issue #13786)
Reported by: tzafrir
When compiling against Zaptel dahdi_compat will now only define all the DAHDI defines if the Zaptel define is present. Also, there is no such thing as DAHDI_PRI.
Steve Murphy [Fri, 21 Nov 2008 23:14:50 +0000 (23:14 +0000)]
In reference to the fix made for 13871, I was
merging the fix into 1.6.0 and realized I missed
the code in the h-exten block, and didn't catch it
because my test case had the h-exten commented out.
So, this corrects the code I missed, as a
preventative against another crash report.
Tested with the h-exten defined, all is well.
Tilghman Lesher [Fri, 21 Nov 2008 23:07:46 +0000 (23:07 +0000)]
The passed extension may not be the same in the list as the current entry,
because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
Steve Murphy [Fri, 21 Nov 2008 21:19:47 +0000 (21:19 +0000)]
(closes issue #13871)
Reported by: mdu113
This one is totally my fault. The code doesn't even
create a bridge if the channel CDR has POST_DISABLED.
I didn't check for that at the end of the bridge.
Fixed with a few small insertions. Tested. Looks
good. No cdr generated, no crash, no unnecc. data
objects created either.
Mark Michelson [Thu, 20 Nov 2008 18:06:48 +0000 (18:06 +0000)]
There was an issue when attempting to reference an embedded
frame in a freed ast_filestream. This patch makes use of the
ao2 functions to make sure that we do not free an ast_filestream
structure until the embedded ast_frame has been "freed" as well.
Mark Michelson [Thu, 20 Nov 2008 17:48:42 +0000 (17:48 +0000)]
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
Mark Michelson [Thu, 20 Nov 2008 17:33:06 +0000 (17:33 +0000)]
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
Kevin P. Fleming [Wed, 19 Nov 2008 21:34:47 +0000 (21:34 +0000)]
the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
Mark Michelson [Tue, 18 Nov 2008 22:47:57 +0000 (22:47 +0000)]
Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.
Jeff Peeler [Tue, 18 Nov 2008 19:13:33 +0000 (19:13 +0000)]
(closes issue #13899)
Reported by: akkornel
This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer.
Mark Michelson [Tue, 18 Nov 2008 18:25:55 +0000 (18:25 +0000)]
Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
projects / thirdparty / asterisk.git / log
