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connect to a channel.
Before committing to 1.4 i would like some other people to
review and test this fix - thanks.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48573 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48193 | kpfleming | 2006-12-01 17:37:28 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines
if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)
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In general this code needs a deep revision, because the body of
do_forward() deletes/overwrites the output channel without freeing
the resouce in some cases, and without notifying the caller.
Also, on FreeBSD with MALLOC_OPTIONS set i am seeing various panics
(duplicate freee etc.)
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In the original code this would happen in the case of
o->forwards >= AST_MAX_FORWARDS
Likely an 1.2/1.4 isse as well - please someone have a look,
while I am hunting a few more similar panics now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47880 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47850 | file | 2006-11-20 10:51:37 -0500 (Mon, 20 Nov 2006) | 2 lines
Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47782 | qwell | 2006-11-16 17:19:46 -0600 (Thu, 16 Nov 2006) | 2 lines
Fix a couple of typos. Initially pointed out by mrobinson.
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some possibly missing parts in the privacy screening code.
Now that it is more streamlined it is easier to see differences
in handling the various cases.
Have not tested the code in depth.
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another one.
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On passing, avoid two null-pointer string dereference
while printing messages (which are
sometimes not fatal in some platforms, but still wrong).
These two lines at least should be merged to 1.4 once i am
done with all the changes here.
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Mark with XXX a possible bug in previous code which used
the wrong source in case of a forwarded call.
the function do_forward() needs to be split further, as the initial
part is replicated in another places (with some minor differences, most likely
forgotten when updating after the copy).
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as usual a bit at a time to ease locating new bugs or fixes
worth merging into other branches.
In this commit, introduce a macro, S_REPLACE, that replaces
a string possibly freeing the previous value.
In one of these places (see the comment marked XXX) the previous
code might leak memory - if so, this ought to be merged in 1.4
The macro might be worth putting in one of the global headers
(e.g. include/asterisk/strings.h) as the construct is used
in a million places in the asterisk code.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47160 f38db490-d61c-443f-a65b-d21fe96a405b
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asterisk. I include some changes to astman to cover other events that have been added.
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patch provided in bugnote, with minor changes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44253 f38db490-d61c-443f-a65b-d21fe96a405b
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number and CallerIDName for name. (issue #7976 reported by suhler)
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minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
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bug 2617
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hopefully avert any more 7544 type bug reports\!
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38928 | russell | 2006-08-05 02:37:59 -0400 (Sat, 05 Aug 2006) | 3 lines
make sure the priv-callerintros directory exists before trying to create a file
there (issue #7659, patch by hads, with some modifications by me)
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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37988 f38db490-d61c-443f-a65b-d21fe96a405b
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commands, forward hold/unhold in dial, add hold device state
and implement holding in the SLA.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37318 f38db490-d61c-443f-a65b-d21fe96a405b
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(#7362 - twlison / vechers confirming fix)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r33294 | file | 2006-06-09 15:08:00 -0300 (Fri, 09 Jun 2006) | 2 lines
Handle hangup during recording of screened name (issue #7304 reported by kulldominique)
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again :-)
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forward request, and actually set tmp->chan to NULL when we reject the forward request
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31049 f38db490-d61c-443f-a65b-d21fe96a405b
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requests from peers to forward calls elsewhere. #5657 (johnlange w/some minor mods)
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option_priority_jumping not respected) (reported by jkoopmann and branch by murf)
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/trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340.
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#7095 reported by raarts)
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r24567 | bweschke | 2006-05-03 15:58:10 -0500 (Wed, 03 May 2006) | 3 lines
Correct application documentation to make users aware that certain options cannot be used in conjunction with others. #6666 (chotaire)
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in app_dial to support the use of this mode.
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Mark a potentially missing item in managerevent
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for future removal
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(indentation still to be fixed)
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