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Author Commit Message Labels Comments Date
russell
s/1.10/10.0/
Tags
10.0.0-beta1
Branches
10
rmudgett
Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........
Branches
10
rmudgett
Merged revisions 329199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. Was documented as: PickupChan(channel[,channel2[,...][,options]]) Fixed documentation to: PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) This is a continuation of ASTERISK-174…
Branches
10
qwell
Fix version number in UPGRADE.txt.
Branches
10
rmudgett
Merged revisions 329144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked! This appears to be a leftover from when ast_channel was converted to ao2 objects. Simply removed the extraneous unlock. (closes issue ASTERISK-17772) ........
Branches
10
russell
Change Asterisk 2.0 to 2.0 in binary
Branches
10
pabelanger
Merged revisions 329027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for PRI support. ........
Branches
2.0
twilson
Merged revisions 328987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines We can't guarantee an eth0 is present FreeBSD test fails on this case presumably because there is no eth0 on the test machine. Better to just remove this test for now. ........
Branches
2.0
kmoore
Merged revisions 328935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833…
Branches
2.0
kpfleming
Merged revisions 328878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines Revert partial attempt at handling pathnames with spaces. Revision 299794 attempted to improve the build system to be able to handle pathnames (primarily DESTDIR) with spaces in them, since this is common on some platforms (including Mac OSX). Unfortunately, the changes were incomplete and did not …
Branches
2.0
twilson
Fix mistaken version number
Branches
2.0
kmoore
Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Aster…
Branches
1.10
kmoore
Merged revisions 328770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines MeetMe requests a PIN twice in some circumstances If a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference. This behavior was introduced in rev 311616 when adding th…
Branches
1.10
twilson
Merged revisions 328716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This commit also adds linked list unit tests. Review: https://reviewboard.asterisk.org/r/1321/ ........
Branches
1.10
markm
Merged revisions 328663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines app_dial may double free a channel datastore When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it. (closes issue ASTERIS…
Branches
1.10
markm
Merged revisions 328608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines If the sip private structure is null, sip_setoption() will defref the null pointer and crash. Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash. (closes issue ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark Murawski ........
Branches
1.10
markm
Merged revisions 328593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines Fixed invalid read and null pointer deref on asterisk shutdown. In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash. (closes issue ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey…
Branches
1.10
tilghman
Merged revisions 328540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines Typo ........
Branches
1.10
lmadsen
Build app_macro by default because things depend on it.
Branches
1.10
lmadsen
Update UPGRADE.txt and CHANGES files. Update documentation files stating that deprecated modules are no longer built by default.
Branches
1.10
lmadsen
Blocked revisions 328446 via svnmerge ........ r328446 | lmadsen | 2011-07-15 16:41:12 -0400 (Fri, 15 Jul 2011) | 1 line Revert changes to defaultenabled state for modules in Asterisk 1.8 ........
Branches
1.10
wedhorn
Add SLA to skinny. Adds sublines to skinny lines. Each subline can be attached to an SLA station/trunk combo. Includes the following functionality: Callid is persistent for both in/out calls on all skinny devices. Can join, hold, resume. All sublines appear under a single line button. See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc.
Branches
1.10
may
delete unproperly changed svn props
Branches
1.10
may
Merged revisions 328427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines small gk processing fixes: - decrease for 1 second registration ttl for very low expirations (some providers expire few earlier than TTL) - delete rrq and registration expire timers on URQ received as we make new registration. ........
Branches
1.10
rmudgett
Make hint watcher callback take const strings for context and exten parameters.
Branches
1.10
rmudgett
Merged revisions 328302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines Missing SIP pvt and channel unlock in sip_set_rtp_peer(). Regression introduced by -r326144. Add missing SIP pvt and channel unlock in sip_set_rtp_peer(). ........
Branches
1.10
lmadsen
Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the suppo…
Branches
1.10
jrose
Merged revisions 328205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines Monitor application arguments requirements fixed. Monitor was requiring options in spite of no individual option on Monitor being required. Review: https://reviewboard.asterisk.org/r/1320/ ........
Branches
1.10
mnicholson
tune the v21 preamble detector to properly detect the desired sequence of hits and misses
Branches
1.10
dvossel
Preserve sample rate quality of wideband mixmonitor recordings. MixMonitor has the ability to record in any file format Asterisk supports, but the quality of wideband audio is not preserved. This is because regardless of the sample rate the call is being recorded in, the audio is always downsampled to 8khz and then upsampled to whatever wideband format it is being written as. This patch resolves this by requesting the audio from the audiohook in the signed l…
Branches
1.10
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