https://origsvn.digium.com/svn/asterisk/trunk
................
r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines
Merged revisions 160480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
Jon Bonilla (Manwe) pointed out on the -dev list:
"I guess that having only ip-phones in mind is not a good approach. Since it is
possible to have a sip proxy connected to asterisk we could receive a 407
(unauthorized) or 483 (too many hops) as response and dialog ending would not be
a good behavior."
So modified.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r152969 | tilghman | 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) | 10 lines
Merged revisions 152958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines
Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
(Closes issue #13400)
........
................
r153122 | tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 lines
Merged revisions 153114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines
Turn off qualify on uncached realtime peers.
(Closes issue #13383)
........
................
r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines
Recorded merge of revisions 154263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines
Make the monitor thread non-detached, so it can be joined (suggested by Russell
on -dev list).
........
................
r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines
Merged revisions 154266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines
JIRA ABE-1703
mISDN sets the channel to the wrong state when it receives
the indication AST_CONTROL_RINGING.
........
................
r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines
Merged revisions 154365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines
On busy systems, it's possible for the values checked within a single line
of code to change, unless the structure is locked to ensure a consistent
state.
(closes issue #13717)
Reported by: kowalma
Patches:
20081102__bug13717.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
........
................
r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines
Merged revisions 155398 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines
Clarify error message.
(closes issue #13809)
Reported by: denke
Patches:
20081104__bug13809.diff.txt uploaded by Corydon76 (license 14)
Tested by: denke
........
................
r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines
Merged revisions 155861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines
Channel drivers assume that when their indicate callback
is invoked, that the channel on which the callback was called
is locked. This patch corrects an instance in chan_agent where
a channel's indicate callback is called directly without first
locking the channel.
This was leading to some observed locking issues in chan_local,
but considering that all channel drivers operate under the
same expectations, the generic fix in chan_agent is the right
way to go.
AST-126
........
................
r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines
Merged revisions 156164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines
Move the sanity check that makes sure "always fork" is not set along with the
console option to be after the code that reads options from asterisk.conf.
This resolves a situation where Asterisk can start taking up 100% when
misconfigured.
(Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.)
........
................
r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines
Merged revisions 156294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines
If the SLA thread is not started, then reload causes a memory leak.
(closes issue #13889)
Reported by: eliel
Patches:
app_meetme.c.patch uploaded by eliel (license 64)
........
................
r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines
Merged revisions 156688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines
Provide more space for all the data which can appear in an originating
channel name.
(closes issue #13398)
Reported by: bamby
Patches:
manager.c.diff uploaded by bamby (license 430)
........
................
r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines
Merged revisions 156755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
ast_waitfordigit() requires that the channel be up, for no good logical
reason. This prevents While/EndWhile from working within the "h"
extension.
Reported by: jgalarneau (for ABE C.2)
Fixed by: me
........
................
r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
........
................
r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines
Merged revisions 158071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
........
................
r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines
Merged revisions 158539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines
When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock
........
................
r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines
Merged revisions 158600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines
The passed extension may not be the same in the list as the current entry,
because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
........
................
r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines
Merged revisions 159269 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines
Don't try to send a response on a NULL pvt.
(closes issue #13919)
Reported by: barthpbx
Patches:
chan_iax2.c.patch uploaded by eliel (license 64)
Tested by: barthpbx
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) | 13 lines
Merged revisions 152215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) | 6 lines
Inherit ALL elements of CallerID across a local channel.
(closes issue #13368)
Reported by: Peter Schlaile
Patches:
20080826__bug13368.diff.txt uploaded by Corydon76 (license 14)
........
................
r152287 | jpeeler | 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines
Merged revisions 152286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) | 2 lines
Buffer policy setting for half is not needed.
........
................
r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines
Merged revisions 152368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines
Reset all DIAL variables back to blank, in case Dial is called multiple times
per call (which could otherwise lead to inconsistent status reports).
(closes issue #13216)
Reported by: ruddy
Patches:
20081014__bug13216.diff.txt uploaded by Corydon76 (license 14)
Tested by: ruddy
........
................
r152467 | tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10 lines
Merged revisions 152463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) | 3 lines
Quoting in the wrong direction
(Fixes AST-107)
........
................
r152569 | russell | 2008-10-29 00:34:26 -0500 (Wed, 29 Oct 2008) | 15 lines
Merged revisions 152539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) | 7 lines
Fix an incorrect usage of sizeof()
(closes issue #13795)
Reported by: andrew53
Patches:
chan_sip_sizeof.patch uploaded by andrew53 (license 519)
........
................
r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) | 22 lines
Merged revisions 152538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines
A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.
I hope this doesn't spoil some vast, eternal plan...
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) | 9 lines
Merged revisions 147517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines
If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8)
........
................
r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines
Merged revisions 147681 via svnmerge from
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)
........
................
r148000 | tilghman | 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines
Merged revisions 147997 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines
When blank, callerid name and number should display "unknown caller" in voicemail
emails.
(Closes issue #13643)
........
................
r148112 | mmichelson | 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines
Merged revisions 146026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines
(closes issue #13579)
Reported by: dwagner
(closes issue #13584)
Reported by: dwagner
Tested by: murf, putnopvut
The thought occurred to me that the res= from the extension spawn
was ending up being returned from the bridge.
"Thou shalt not poison the return value". Made the change
and it appears to allow blind xfers to work as normal.
If I'm wrong, reopen the bugs. But it looks good to me!
Many thanks to putnopvut for helping me reproduce this!
........
................
r148268 | tilghman | 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines
Merged revisions 148257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines
User not notified of temporary greeting, if ODBC storage is in use.
(closes issue #13659)
Reported by: moliveras
Patches:
20081009__bug13659.diff.txt uploaded by Corydon76 (license 14)
Tested by: moliveras
........
................
r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) | 11 lines
Merged revisions 148916 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines
Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used
in headers like 'Subject' and 'To'.
Closes AST-107.
........
................
r148988 | tilghman | 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines
Merged revisions 148987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines
Some compilers warn, some don't. Fixing.
........
................
r149062 | tilghman | 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) | 13 lines
Merged revisions 149061 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines
Check correct values in the return of ast_waitfor(); also, get rid of a
possible memory leak.
(closes issue #13658)
Reported by: explidous
Patch by: me
........
................
r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) | 15 lines
Merged revisions 149130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines
Don't allow reserved characters to be used in register
lines in sip.conf.
(closes issue #13570)
Reported by: putnopvut
........
................
r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines
Merged revisions 149200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines
Update the queue with the correct number of calls and
whether the call was completed within the service level
when a transfer takes place. This way, we do not "break"
the leastrecent and fewestcalls strategies by not logging
a call until after the transferred call has ended.
(closes issue #13395)
Reported by: Marquis
Patches:
app_queue.c.transfer.patch uploaded by Marquis (license 32)
........
................
r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines
Merged revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines
Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
........
................
r149208 | mmichelson | 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines
Merged revisions 149207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines
Call register_peer_exten even in the case that the peer's
IP/port does not change.
(closes issue #13309)
Reported by: dimas
Patches:
v2-13309.patch uploaded by dimas (license 88)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines
Merged revisions 160297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines
When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion
fails, and the resulting integer is garbage. Thus, we must initialize the
integer and check it afterwards for success.
(closes issue #14000)
Reported by: folke
Patches:
asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines
incorporates r159808 from branches/1.4:
------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
------------------------------------------------------------------------
in addition:
move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@159855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines
Use some magic constants to get the right size
for this sscanf statement. Thanks Richard!
........
r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines
Use a more expressive constant for a 64-bit scanned int
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines
Fix the build for 32-bit systems. %lu is only 32-bits
on 32-bit systems, so we need to use %llu instead. Of course
%llu is 128-bits on 64-bit systems, so we have to cast to
unsigned long long. No harm, but it's sure annoying.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines
Change the remote user agent session version variable
from an int to a uint64_t. This prevents potential comparison
problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could
not properly place a call on hold since the version in the
SDP of the re-INVITE to place the call on hold greatly
exceeded INT_MAX.
This also aligns with RFC 2327 better since it recommends
using an NTP timestamp for the version (which is a
64-bit number).
(closes issue #13531)
Reported by: sgofferj
Patches:
13531.patch uploaded by putnopvut (license 60)
Tested by: sgofferj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines
make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines
Merged revisions 157503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines
Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.
(closes issue #13878)
Reported by: nahuelgreco
Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
Tested by: putnopvut
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines
Based on Russell's advice on the asterisk-dev list, I have
changed from using a global lock in update_call_counter to
using the locks within the sip_pvt and sip_peer structures
instead.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines
* Add a lock to be used in the update_call_counter function.
* Revert logic to mirror 1.4's in the sense that it will not allow
the call counter to dip below 0.
These two measures prevent potential races that could cause a SIP peer
to appear to be busy forever.
(closes issue #13668)
Reported by: mjc
Patches:
hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@155468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@151765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@150308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r144025 | mmichelson | 2008-09-23 10:37:00 -0500 (Tue, 23 Sep 2008) | 16 lines
When a promiscuous redirect contained both a user and
host portion in the Contact URI and specifies a
transport, the parsing done in parse_moved_contact
resulted in a malformed URI.
This commit fixes the parsing so that a proper
Dial string may be formed when the forwarded
call is placed.
(closes issue #13523)
Reported by: mattdarnell
Patches:
13523v2.patch uploaded by putnopvut (license 60)
Tested by: mattdarnell
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@144026 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines
Merged revisions 142865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@142867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines
Merged revisions 141565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line
This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@141567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug 2008) | 19 lines
Merged revisions 140299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines
Fix tag checking in get_sip_pvt_byid_locked when
in pedantic mode. The problem was that the wrong
tags would be compared depending on the direction
of the call.
(closes issue #13353)
Reported by: flefoll
Patches:
chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@140302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, 18 Aug 2008) | 1 line
While we're at it, make this machine parseable too.
........
r138779 | seanbright | 2008-08-18 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line
And remove code we don't need anymore.
........
r138780 | seanbright | 2008-08-18 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line
Let it compile now, too (woops)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines
SIP should use the transport type set in the Moved Temporarily for the next
invite.
(closes issue #11843)
Reported by: pestermann
Patches:
20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36)
20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36)
Tested by: pabelanger
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul 2008) | 11 lines
Merged revisions 132777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
Allow Spiraled INVITEs to work correctly within Asterisk.
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
(closes issue #7403)
Reported by: stephen_dredge
Modified:
branches/1.4/channels/chan_sip.c
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 lines
Merged revisions 132645 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
Merging this rev from trunk to 1.6.0 was not
simple. Why? Because we've enhanced trunk to
do a [fast] merge-and-delete operation which
also solved problems with contexts having
entries from different registrars.
Fast as in the amount of time the contexts
are locked down. That *is* fast, but traversing
the entire dialplan looking for priorities to
delete takes more time overall.
This particular fix involved pulling in those
enhancements from trunk, along with all the
various fixes and refinements made along the
way.
Merging all this from trunk into 1.6 involved:
a. mergetrunk6 in the stuff from 130145;
b. revert all but the prop changes
c. catalog all revisions to pbx.c since 1.6.0 was forked
(at rev 105596).
d. catalog all revisions to pbx.c in trunk since 1.6.0
was forked, making special note of all revs that
were not merged into 1.6.0.
e. study each rev in trunk not applied to 1.6.0, and
determine if it was involved in the merge_and_delete
enhancements in trunk. 25 commits were done in 1.6.0,
all but one (106306) was a merge from trunk.
Trunk had 22 additional changes, of which 7 were
involved in the merge_and_delete enhancements:
106757
108894
109169
116461
123358
130145
130297
f. Go to trunk and collect patches, one by one,
of the changes made by each rev across the
entire source tree, using svn diff -c <num> > pfile
g. Apply each patch in order to 1.6.0, and
resolve all failures and compilation problems
before proceding to the next patch.
h. test the stuff.
i. profit!
........
r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines
(closes issue #13041)
Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
