........
r280227 | qwell | 2010-07-28 14:54:54 -0500 (Wed, 28 Jul 2010) | 1 line
Add sha1sum-sh in case there is no util on the system.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The version of libedit that is bundled with asterisk is old and has some bugs.
This patch updates the bundled version of libedit within asterisk, and also
updates asterisk to use the system libedit instead if one is available (and
pkg-config is available). This review integrates several patches from other
users specifically kkm and tzafrir.
(closes issue #15929)
Reported by: kkm
Patches:
015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
(issue #16858)
Reported by: jw-asterisk
(closes issue #17039)
Reported by: tzafrir
Patches:
0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/807/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines
Merged revisions 279945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a regression from the sig_pri split from chan_dahdi. When a call is
first initiated, the inband DTMF detector is not enabled if it's an outgoing
ISDN call. However, it needs to be turned on once the media path starts up.
This handling was put back in the open_media() callback of chan_dahdi. In
sig_pri, open_media() calls were added to a few places where it was needed,
including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code was written in a way that did a bad job of
parsing the port out of a URI. Specifically, it would
do badly when dealing with an IPv6 address. In this
particular scenario, there was no value from parsing
the port out, so I just removed that logic. And while
I was messing around in the function, I changed some
variable names to be more descriptive.
(closes issue #17661)
Reported by: oej
Patches:
17661.diff uploaded by mmichelson (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r279883 | qwell | 2010-07-27 12:54:54 -0500 (Tue, 27 Jul 2010) | 1 line
Add SHA1SUM to configure, since we require it for sounds/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines
Fix bad behavior of dynamic_exclude_static option in sip.conf.
We were attempting to create a contactdeny rule based on the peer's
IP address before the peer's IP address had been set. By moving the
processing further down in the function, we can ensure stuff works
as we expect for it to.
(closes issue #17717)
Reported by: mmichelson
Patches:
17717.patch uploaded by mmichelson (license 60)
Tested by: DennisD
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies the way chan_sip.c does transaction to dialog
matching. Asterisk now stores information in the top most Via header
of the initial incoming request and compares that against other Requests
that have the same call-id. This results in Asterisk being able to
detect a forked call in which it has received multiple legs of the
fork. I completely stripped out the previous matching code and made
the comparisons a little more explicit and easier to understand. My
comments in the code should offer all the details involving this patch.
This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns (CMP_MATCH)
on a match allowing for multiple items to be returned.
Review: https://reviewboard.asterisk.org/r/776/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines
Merged revisions 279206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines
SIP promiscuous redirect could fail to dial the redirect.
The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable. The two variables are not equivalent if the call_forward string
included a channel technology specifier. e.g., SIP/200
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
