This patch removes code that was duplicated from pbx.c to manager.c
in order to prevent API change in released versions of Asterisk.
There are propably also other places that would benefit from reading the
return code and react if a function returns error codes on writing a value into it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242919 f38db490-d61c-443f-a65b-d21fe96a405b
Initialize the calendars container before calling load_config and return FAILURE
on allocation failure. Also, use the AST_MODULE_LOAD_* values for return values.
Thanks to rmudgett for pointing out the error and the need to use the defined
values for return
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242812 f38db490-d61c-443f-a65b-d21fe96a405b
incorrect q.931 message order filtered on incoming calls (first msg must be setup,
next must be not setup)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242645 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242521 f38db490-d61c-443f-a65b-d21fe96a405b
When this code was developed, we came up with our own XML format for the test
output. I have since started looking at integration with other tools, namely
continuous integration frameworks, and this format seems to be supported
across a number of applications. With these changes in place, I was able
to get Atlassian Bamboo to interpret the test results.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241855 f38db490-d61c-443f-a65b-d21fe96a405b
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call,
by executing ast_cdr_update() early in __ast_pbx run.
Reverts sig_pri changes in trunk that are specific to isdn technology only.
(closes issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241416 f38db490-d61c-443f-a65b-d21fe96a405b
passdata was only being set in pbx_substitue_variables when arguments were
passed.
(closes issue #16406)
(closes issue #16586)
Reported by: DLNoah
Patches:
bug16586v2.patch uploaded by jpeeler (license 325)
Tested by: DLNoah
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241366 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 Jan 2010) | 13 lines
Fix deadlock in agent_read by removing call to agent_logoff.
One must always lock the agents list lock before the agent private. agent_read
locks the private immediately, so locking the agents list lock is not an
option (which is what agent_logoff requires). Because agent_read already
has access to the agent private all that is necessary is to do the required
hanging up that agent_logoff performed.
(closes issue #16321)
Reported by: valon24
Patches:
bug16321.patch uploaded by jpeeler (license 325)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241314 f38db490-d61c-443f-a65b-d21fe96a405b
After some back and forth with the reporter, we came up with the necessary changes.
(closes issue #16489)
Reported by: Chainsaw
Patches:
asterisk-1.6.2.1-parallel-make-minimal.patch uploaded by Chainsaw (license 723)
Tested by: Chainsaw, qwell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241230 f38db490-d61c-443f-a65b-d21fe96a405b
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.
(issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241187 f38db490-d61c-443f-a65b-d21fe96a405b
If the limit was set past MAX_INT upon answering, the call was immediately
hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup).
The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been
changed to return an int64_t to prevent overflow. Also the reporter suggested
adding a message indicating the reason for the call hanging up. Given that the
new limit is so much higher, the message (which would only really be useful in
the overflow scenario) has been made a debug message only.
(closes issue #16006)
Reported by: viraptor
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241143 f38db490-d61c-443f-a65b-d21fe96a405b
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.
(closes issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff.txt uploaded by alecdavis (license 585)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241097 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r241015 | seanbright | 2010-01-18 14:54:19 -0500 (Mon, 18 Jan 2010) | 12 lines
Plug a memory leak when reading configs with their comments.
While reading through configuration files with the intent of returning their
full contents (comments specifically) we allocated some memory and then forgot
to free it. This doesn't fix 16554 but clears up a leak I had in the lab.
(issue #16554)
Reported by: mav3rick
Patches:
issue16554_20100118.patch uploaded by seanbright (license 71)
Tested by: seanbright
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241016 f38db490-d61c-443f-a65b-d21fe96a405b
Rewrote a large portion of the existing documentation
and added information about the TCP/IP socket interface
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240973 f38db490-d61c-443f-a65b-d21fe96a405b
In asterisk.conf, transmit_silence_during_record has been removed
in favor of using only the transmit_silence option. The
transmit_silence_during_record option remains a valid option in
asterisk.conf, but has been removed from the sample config and
noted in CHANGES.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240971 f38db490-d61c-443f-a65b-d21fe96a405b
Add file information to data element of T event so
the file information is sent to the client when it is
interrupted. Previously only notification of pending
files that were dropped was sent
(closes issue #16147)
Reported by: thedavidfactor
Tested by: thedavidfactor
Review: https://reviewboard.asterisk.org/r/449/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240969 f38db490-d61c-443f-a65b-d21fe96a405b
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r240891 | dvossel | 2010-01-18 10:51:35 -0600 (Mon, 18 Jan 2010) | 7 lines
updated transmit_silence option documentation in asterisk.conf
This patch updates the transmit_silence option to better document
why the option exists, and what it affects. Thanks to russell
for providing the verbage for this update.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240892 f38db490-d61c-443f-a65b-d21fe96a405b
This patch updates the transmit_silence option to better document
why the option exists, and what it affects. Thanks to russell
for providing the verbage for this update.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240887 f38db490-d61c-443f-a65b-d21fe96a405b
russell