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r156817 | mmichelson | 2008-11-14 09:20:03 -0600 (Fri, 14 Nov 2008) | 18 lines
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r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines
If the prompt to reenter a voicemail password timed out, it
resulted in the password not being saved, even if the input matched
what you gave when first prompted to enter a new password. This is
because the return value of ast_readstring was checked, but not checked
properly.
This bug was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!
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r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | 26 lines
Merged revisions 156297 via svnmerge from
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r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines
It turns out that the 0x0XX00 codes being returned for
N, X, and Z are off by one, as per conversation with
jsmith on #asterisk-dev; he was teaching a class
and disconcerted that this published rule was not
being followed, with patterns _NXX, _[1-8]22 and
_[2-9]22... and NXX was winning, but [1-8] should
have been.
This change, tested on these 3 patterns now
picks the proper one.
However, this change may surprise users who
set up dialplans based on previous behavior,
which has been there for what, 2 and half
years or so now.
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r156243 | tilghman | 2008-11-12 12:55:18 -0600 (Wed, 12 Nov 2008) | 18 lines
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r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines
Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not
to be sent, and instead, schedule a task to destroy the iax2 pvt structure
10 seconds later. This allows the IAX2 HANGUP packet to be queued,
transmitted, and ACKed before the pvt is destroyed.
(closes issue #13645)
Reported by: dzajro
Patches:
20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: vazir
Reviewed: http://reviewboard.digium.com/r/51/
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r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) | 15 lines
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r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines
When doing some tests, I was having a crash at the end of every call
if an attended transfer occurred during the call. I traced the cause to
the CDR on one of the channels being NULL. murf suggested a check in
the end bridge callback to be sure the CDR is non-NULL before proceeding,
so that's what I'm adding.
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r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines
Set the invite state to INV_CANCELLED in a place that
makes more sense. Where it was set before, it was impossible
to actually delay sending a CANCEL if we had not yet received
a provisional response to an INVITE.
(closes issue #13626)
Reported by: atis
Patches:
13626.patch uploaded by putnopvut (license 60)
Tested by: atis
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r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, 07 Nov 2008) | 8 lines
Remove one more instance of the sample configuration
lying about what's possible. The tz cannot be set in a
context like this. It can only be set in the general
section or per-mailbox.
Thanks to sasargen on #asterisk-dev for pointing this out
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r155012 | mmichelson | 2008-11-06 13:46:53 -0600 (Thu, 06 Nov 2008) | 16 lines
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r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines
The documentation listed the ability to set 'maxmsg' per
context. The truth is that you can only set this in the general section
or per mailbox. Thus I am updating the sample config file to be more
accurate.
Thanks to sasargen on IRC for bringing up this issue.
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r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) | 3 lines
instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it.
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r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines
Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call
(closes issue #13793)
Reported by: greenfieldtech
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r152646 | mmichelson | 2008-10-29 15:53:53 -0500 (Wed, 29 Oct 2008) | 9 lines
If there was no named defined in a voicemail.conf mailbox
entry, then app_directory would crash when attempting to
read that entry from the file. We now check for the NULL
or empty string properly so that there will be no crash.
(closes issue #13804)
Reported by: bluecrow76
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r152060 | seanbright | 2008-10-26 16:25:08 -0400 (Sun, 26 Oct 2008) | 15 lines
Merged revisions 152059 via svnmerge from
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r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct 2008) | 7 lines
Since passing \0 as the second argument to strchr is valid (and will
match the trailing \0 of a string) we need to check that first, otherwise
we end up with incorrect results. Fix suggested by reporter.
(closes issue #13787)
Reported by: meitinger
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- Freeing the peer got accidentally removed from the peer's destructor. It is
still needed for astobj, but not for astobj2.
- Fix some places that called find_user or find_peer, but did not release the
reference that was returned.
(closes issue #13331)
Reported by: sergee
Patches:
chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138)
Tested by: sergee
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r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct 2008) | 9 lines
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r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct 2008) | 3 lines
break up acinclude.m4 into individual files, which will make it easier to maintain, easier to add new macros (less patching) and will ease maintenance of these macros across Asterisk branches
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r151243 | kpfleming | 2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines
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r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct 2008) | 2 lines
rename this macro to properly reflect what it does
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r150817 | bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines
Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766.
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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r150307 | mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14 lines
After a long discussion on #asterisk-bugs, it seems kind of
odd that a channel would be named after the port on which it
came in on. For endpoints that always include ":5060" as part
of the From: header, it will mean that you have a ton of
channels with names like "SIP/5060-3ea38a8b."
I am boldly moving forward with this change in trunk, but I'm
not touching other branches with this one since this definitely
would qualify as a behavior change. If there is a problem with
this commit, and I haven't seen the obvious reason why you'd want
to name the channel after the port from which the call originated,
then please feel free to revert this
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r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct 2008) | 9 lines
Merged revisions 148736 via svnmerge from
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r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct 2008) | 3 lines
on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly
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r148519 | murf | 2008-10-13 11:14:38 -0600 (Mon, 13 Oct 2008) | 15 lines
Hmmm. Nobody (but me) is interested in seeing
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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r148373 | mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 lines
Make sure that the inUse and inRinging fields for
a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well.
(closes issue #13668)
Reported by: mjc
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r148200 | seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12 lines
Don't include logger.h in asterisk.h by default as it is causing problems building
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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r148144 | mmichelson | 2008-10-09 18:30:47 -0500 (Thu, 09 Oct 2008) | 10 lines
Read the callerid in the correct order and make sure to
read the Urgent flag value from the IMAP headers.
(closes issue #13652)
Reported by: jaroth
Patches:
imapheaders.patch uploaded by jaroth (license 50)
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r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines
(closes issue #13557)
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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