for more details of this command.
(closes issue #13326)
Reported by: ib2
Patches:
bug13326_trunk_20080822.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
odd that a channel would be named after the originating port.
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
than what was in the invite_branch of a sip_pvt, meaning
that if a CANCEL were sent later, the branch in the CANCEL
would not match the branch in the latest INVITE sent out, leading
to some endpoints responding to the CANCEL with a 481.
(closes issue #13714)
Reported by: fnordian
Patches:
invite_branch.patch uploaded by fnordian (license 110)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- move all setting of 'needdestroy' on dialog structures into the history
- report all tags involved when a pedantic check fails on a REFER
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
necessary to allow for a sip_pvt to maintain a reference
to a sip_peer's outboundproxy even after the peer has
been freed.
(closes issue #13700)
Reported by: fnordian
Patches:
13700.patch uploaded by putnopvut (license 60)
Tested by: fnordian
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We're going to try to get time to fix this and kpfleming believes that there's code in ao2
so that we can solve it...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
in the "kill the user" commit and caused calls relying
on the insecure setting to not work properly. I changed
for finding a peer back to how it was prior to that
commit.
(closes issue #13644)
Reported by: pj
Patches:
13644_trunkv2.patch uploaded by putnopvut (license 60)
Tested by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148376 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines
when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06 Oct 2008) | 9 lines
Check whether an extension exists in the _call method, rather than the _alloc
method, because we need to evaluate the callerid (since that data affects
whether an extension exists).
(closes issue #13343)
Reported by: efutch
Patches:
20080915__bug13343.diff.txt uploaded by Corydon76 (license 14)
Tested by: efutch
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: pj
Tested by: pj
Set transport to SIP_TRANSPORT_UDP mode if not specified which fixes calls to get_transport returning UNKNOWN.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
call-id in the dialog-info event package used with extension state subscriptions
on Snom phones. Then, when the phone sends an INVITE with Replaces for the
special callid, Asterisk will perform a pickup on the extension that was
subscribed to.
The original code on this issue was submitted by xylome. However, contributions
have been made by (at least) mgernoth and pkempgen. The final patch was written
by seanbright, and includes the necessary logic to allow this work in a
technology independent way.
(closes issue #5014)
Reported by: xylome
Patches:
issue5014-trunk.diff uploaded by seanbright (license 71)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: nickpeirson
The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;
chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Michiel van Baak
Mark Michelson
Richard Mudgett
Kevin P. Fleming
Olle Johansson
Tilghman Lesher
Joshua Colp
Steve Murphy
Sean Bright
Terry Wilson
Jason Parker
Jeff Peeler
Russell Bryant
Christian Richter