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r160555 | mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11 lines
When investigating issue #13548, I found that gosub
handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.
With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.
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r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines
Merged revisions 160480 via svnmerge from
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r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
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.
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r160447 | eliel | 2008-12-03 09:01:23 -0200 (Wed, 03 Dec 2008) | 4 lines
- Avoid setting .synopsis and .syntax if we are using XML documentation (or the
xml documentation wont be loaded).
- Use <variable></variable> to refer to a dialplan variable.
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r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines
Merged revisions 160297 via svnmerge from
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r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines
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)
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r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec 2008) | 1 line
Pay attention to the return value of system(), even if we basically ignore it.
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r160171 | seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 line
Silence a build warning. (chan_phone.c:810: warning: value computed is not used)
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r160172 | seanbright | 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines
Merged revisions 159976 via svnmerge from
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r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines
Get rid of the useless format string and argument in the Bogus/ manager channelname.
Noted by kpfleming and name Bogus/manager suggested by eliel
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r160062 | eliel | 2008-12-01 16:52:14 -0200 (Mon, 01 Dec 2008) | 13 lines
Introduce CLI permissions.
Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.
(Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/
(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak
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r160004 | russell | 2008-12-01 11:34:31 -0600 (Mon, 01 Dec 2008) | 14 lines
Merged revisions 160003 via svnmerge from
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r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines
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.
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r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines
incorporates r159808 from branches/1.4:
------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
------------------------------------------------------------------------
in addition:
move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
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r159774 | kpfleming | 2008-11-29 09:29:33 -0600 (Sat, 29 Nov 2008) | 5 lines
we can now build with -Wformat=2, which found a couple of real bugs
because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking
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r159734 | mvanbaak | 2008-11-28 15:20:11 +0100 (Fri, 28 Nov 2008) | 8 lines
Make res_config_ldap compile with the official OpenLDAP 2.3.X versions.
They removed the LDAP_DEPRECATED define from their source and since we are using a couple
of deprecated function calls we should define it with a CFLAG.
Tested by me on OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps compiling.
It shouldn't break, we only define the LDAP_DEPRECATED with this which is what
all 2.2.X and older versions of OpenLDAP did in their own tree.
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r159631 | kpfleming | 2008-11-26 15:20:50 -0600 (Wed, 26 Nov 2008) | 5 lines
improve handling of API calls provided by loaded modules through use of some GCC features; this makes app_stack's usage of AGI APIs even cleaner, and will allow it to work 'as expected' either with or without res_agi being loaded
reviewed at http://reviewboard.digium.com/r/62
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r159554 | mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 lines
Add some necessary hangup commands in the case that forwarding
a call fails
1) Hang up the original destination if the local channel cannot
be requested.
2) Hang up the local channel (in addition to the original destination)
if ast_call fails when calling the newly created local channel.
This prevents channels from sticking around forever in the
case of a botched call forward (e.g. to an extension which does not
exist).
(closes issue #13764)
Reported by: davidw
Patches:
13764_v2.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, davidw
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r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, 25 Nov 2008) | 23 lines
Merged revisions 159316 via svnmerge from
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r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines
(closes issue #12694)
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
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.
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r159250 | mmichelson | 2008-11-25 15:49:42 -0600 (Tue, 25 Nov 2008) | 6 lines
Make the options for the general and profiles more consistent
for the "pls_hold_prompt" option. This does not affect any released
version of Asterisk, so there is no need to update the CHANGES
file for this.
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r158959 | seanbright | 2008-11-24 20:01:49 -0500 (Mon, 24 Nov 2008) | 8 lines
This is basically a complete rollback of r155401, as it was determined that
it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
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r158808 | twilson | 2008-11-24 12:11:08 -0600 (Mon, 24 Nov 2008) | 8 lines
This patch adds a new application for sending MWI to phones via Asterisk's event subsystem. Also, the minivm documentation is all converted to use xmldocs.
(closes issue #13946)
Reported by: Marquis
Patches:
minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32)
Tested by: otherwiseguy, Marquis
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r158756 | seanbright | 2008-11-22 22:36:52 -0500 (Sat, 22 Nov 2008) | 6 lines
If you enabled 'notifycid' one of the limitations is that the calling channel
is only found if it dialed the extension that was subscribed to. You can now
specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as
it's value implies, ignore the current context of the caller when doing the
lookup.
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r158754 | seanbright | 2008-11-22 22:30:46 -0500 (Sat, 22 Nov 2008) | 3 lines
No need to use a separate structure for this since we can just pass
our sip_pvt pointer in directly.
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r158723 | mvanbaak | 2008-11-22 18:17:33 +0100 (Sat, 22 Nov 2008) | 4 lines
last commit worked on OpenBSD but still generated warning on Ubuntu.
Initialise a variable so --enable-dev-mode does not complain
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r158694 | mvanbaak | 2008-11-22 17:57:11 +0100 (Sat, 22 Nov 2008) | 8 lines
dont send reorder tone after a device is hungup if a dialout is abandoned or failed.
Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box
(closes issue #13948)
Reported by: wedhorn
Patches:
switch.diff uploaded by wedhorn (license 30)
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r158606 | murf | 2008-11-21 16:40:46 -0700 (Fri, 21 Nov 2008) | 19 lines
Merged revisions 158603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) | 11 lines
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.
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