https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines
We previously attempted to use the ESCAPE clause to set the escape delimiter to
a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89561 f38db490-d61c-443f-a65b-d21fe96a405b
and we now have the groupcount system to implement call-limits in the dialplan. You
can use the "setvar" option in realtime/sip.conf to set limits per device.
- Implement "callcounter" as a new option to enable the call counting we need to
report device status to queue, manager and SIP subscriptions.
The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b
- Fix typo in chan_sip
- Remove changes to caller ID structure, moving it to branch (russellb)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89551 f38db490-d61c-443f-a65b-d21fe96a405b
If needed, the code to extract this information should be implemented
in some generic header or library and the function called here.
(closed bug #11362)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89543 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) | 9 lines
Currently, zero-length voicemail messages cause a hangup in VoicemailMain.
This change fixes the problem, with a multi-faceted approach. First, we
do our best to avoid these messages from being created in the first place,
and second, if that fails, we detect when the voicemail message is
zero-length and avoid exiting at that point.
Reported by: dtyoo
Patch by: gkloepfer,tilghman
(Closes issue #11083)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007) | 10 lines
Up until this point, the XML output of the manager has been technically
invalid, due to the repetition of certain parameters in a single event.
This caused various issues for XML parsers, some of which refused to parse
at all, given the invalidity of the rendered XML. So this commit fixes
the XML output, ensuring that each entity parameter has a unique name, thus
ensuring valid XML.
Reported by: msetim
Patch by: tilghman
(Closes issue #10220)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89537 f38db490-d61c-443f-a65b-d21fe96a405b
This is a NOP as far as the current code is concerned,
but there is already support in ./configure and the
Makefiles for the various libraries used by console_video.c
(not yet in the tree) so addition is trivial.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89533 f38db490-d61c-443f-a65b-d21fe96a405b
make the format explicit in a debug message;
print the audio instead of aggregated peer capability in a debugging msg.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89532 f38db490-d61c-443f-a65b-d21fe96a405b
so we can handle multiple formats properly. This is not carved in stone,
but a proposal to start with.
We need to add support for transliterations as well as UTF8 handling,
propably with libiconv. Murf is looking into that for the dialplan.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89531 f38db490-d61c-443f-a65b-d21fe96a405b
In this commit:
- move the ast_register/unregister_app functions to module.h
to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
dependency of app.h on linkedlists.h
Note, this is a long process that I am doing in small steps.
The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).
This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.
The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89522 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov 2007) | 5 lines
Changing an inaccurate debug message to be less inaccurate. Under the circumstances, this
message would always report that there were 0 members available, even though that may not be true.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) | 4 lines
If a channel gets masqueraded in the middle of a park, don't play the
announcement to the masqueraded channel, and dial back to the original
channel on timeout.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89492 f38db490-d61c-443f-a65b-d21fe96a405b
only do the calculations if fax detection is enabled on the dsp.
(closes issue #11331)
Reported by: dimas
Patches:
dsp.patch uploaded by dimas (license 88)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89490 f38db490-d61c-443f-a65b-d21fe96a405b
