channel list calling a caller-defined callback. The callback returns non-zero
if a match is found. This should speed up some of the code that I committed
earlier today in chan_sip (which is also updated by this commit).
Reviewed by russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
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This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
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had no effect
* Updated dialing API documentation to indicate that timeouts
are specified in milliseconds
* Added a new timeout argument to the Page application. If time
expires, any endpoints which have not answered will be hung up.
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the low default value of featuredigittimeout, I decided it
was high time to change it. I have changed the default to
2000 ms based on a suggestion from Leif Madsen.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
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r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) | 8 lines
Move AMI initialization to occur after loading modules. This prevents a
deadlock when someone tries to initiate a module reload from the AMI just
as Asterisk is starting.
(closes issue #13778)
Reported by: hotsblanc
Fix suggested by hotsblanc
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1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines
2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines)
3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines)
4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied
5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address
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We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing.
(closes issue #13715)
reported by: makoto
patch by: bweschke
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Prodded on IRC by Russell and fixed by eliel
(closes issue #13730)
Reported by: eliel
Patches:
main_cli.patch uploaded by eliel (license 64)
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r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct 2008) | 6 lines
Reverting changes from commits 150298 and 150301 since
I was mistakenly under the assumption that dialplan functions
*always* required that a channel be present. I need to go
home earlier, I think :)
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r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct 2008) | 10 lines
Don't try to call a dialplan function's read callback from
the manager's GetVar handler if an invalid channel has
been specified. Several dialplan functions, including
CHANNEL and SIP_HEADER, do not check for NULL-ness of
the channel being passed in.
(closes issue #13715)
Reported by: makoto
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r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct 2008) | 3 lines
And don't forget to return on the error condition
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r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines
Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
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'update2', which permits updates which match across multiple columns, instead
of requiring all tables to have a single unique identifier. All of the other
API calls with the exception of 'update' already had the ability to match on
multiple fields, so it was a missing and very desireable feature that an API
call implementing an update should have this, too.
This does not change any outward performance of Asterisk, but it should make
life easier for application developers who use the RealTime framework.
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the trie info when they do 'dialplan show ...'
(even with debug set to non-zero); so I set up a
'dialplan debug [context]' cli command instead,
to explicitly show just the trie info. I even
added an extension_exists() call to make sure the
trie info is built. I moved the explanatory header
to above the extension loop to ensure it only prints
once. And it will do this now, whether debug is set
or not.
I removed the trie printing from the 'dialplan show'
command entirely.
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app_voicemail. Instead, include it where it is needed. This turned out to be a
relatively minor issue because other headers include logger.h as well.
Need to test -addons before merging this back to 1.6.0.
(closes issue #13605)
Reported by: tomo1657
Patches:
13605_seanbright.diff uploaded by seanbright (license 71)
Tested by: mmichelson
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been removed. Furthermore, now we actually use the Context argument
passed to set the transfer context and don't error out if no
context is specified.
This addresses the actual problems outlined in issue 12158. Regarding
the other points brought up, regarding the inability to not transfer
to extensions which cannot be represented by DTMF, it is not enough of
a constraint that it is worth attempting to rework the feature.
(closes issue #12158)
Reported by: davidw
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r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines
(closes issue #13579)
Reported by: dwagner
(closes issue #13584)
Reported by: dwagner
Tested by: murf, putnopvut
The thought occurred to me that the res= from the extension spawn
was ending up being returned from the bridge.
"Thou shalt not poison the return value". Made the change
and it appears to allow blind xfers to work as normal.
If I'm wrong, reopen the bugs. But it looks good to me!
Many thanks to putnopvut for helping me reproduce this!
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Reported by: krisk84
Tested by: krisk84
This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur.
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Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
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1) r143204:
The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel.
2) r143270:
Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced.
3) r143475: (the one liner)
compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread
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is no garbage attempted to be used. Also, do not set args to unknown value again if there are
no options passed in.
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read from the .asterisk_history file (and subsequently being duplicated
when written). We weren't checking the result of fgets() which meant
that we read the same line twice before feof() actually returned non-
zero.
Also, stop writing out an extra blank line between each item in the
history file, fix a minor off-by-one error, and use symbolic constants
rather than a hardcoded integer.
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the first item in the CLI entry list. This was preventing '!' from
showing up in either 'help' or in tab completion.
(closes issue #13578)
Reported by: mvanbaak
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problem within strptime(3), which we are correcting here with ast_strptime().
(closes issue #11040)
Reported by: DEA
Patches:
20080910__bug11040.diff.txt uploaded by Corydon76 (license 14)
Tested by: DEA
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to fix blind transfers with my last commit here, I also
caused an unwanted side-effect. That is, only the first
priority of the 'h' extension would be executed when
a blind transfer occurred instead of all priorities.
Essentially, my last commit corrected the return value
of ast_bridge_call. However, the implementation still
was not 100% correct. Now it is.
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within ast_bridge_call was reversed.
This problem was observed when a blind transfer placed from
the callee channel of a test call failed.
While the problem I am solving here is exactly the same
as what was reported in issue #13584, the difference is
that this fix I am applying is trunk-only. Issue #13584
was reported against the 1.4 branch, and my tests
of 1.4's blind transfers appear to work fine.
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This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
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