https://origsvn.digium.com/svn/asterisk/trunk
................
r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec 2008) | 15 lines
Merged revisions 164977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines
After looking through SIP registration code most of the day, this
is one of the few things I could find that was just plain wrong.
Even though it probably isn't possible for it to happen, it seems weird
to have code that checks if a pointer is NULL and then immediately dereferences
that pointer if it was NULL.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines
Merged revisions 164672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines
Fix a memory leak related to the use of the "setvar" configuration option.
The problem was that these variables were being appended to the list of vars
on the sip_pvt every time a re-registration or re-subscription came in.
Since it's just a waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying the vars.
(closes issue #14037)
Reported by: marvinek
Tested by: russell
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines
Rename a number of tcptls_session variables. There are no functional changes here.
The name "ser" was used in a lot of places. However, it is a relic from when
the struct was a server_instance, not a session_instance. It was renamed since
it represents both a server or client connection.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r163629 | file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines
When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found.
(closes issue #13811)
Reported by: pj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines
Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven.
(closes issue #13525)
Reported by: pj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) | 13 lines
Merged revisions 162804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines
Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
(closes issue #12560)
Reported by: vsauer
Patches:
patch001.diff uploaded by ramonpeek (license 266)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) | 13 lines
Merged revisions 162738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines
When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
(closes issue #13599)
Reported by: hjourdain
Patches:
chan_sip.c.diff uploaded by hjourdain (license 583)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r160663 | eliel | 2008-12-03 17:25:30 -0200 (Wed, 03 Dec 2008) | 13 lines
- iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module.
- Move the code to start using the LIST macros.
Review: http://reviewboard.digium.com/r/72
(closes issue #13232)
Reported by: eliel
Patches:
iax2-provision.patch.txt uploaded by eliel (license 64)
(with minor changes pointed by Mark Michelson on review board)
Tested by: eliel
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@160665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.6.1@160483 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/1.6.1@160325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec 2008) | 1 line
Pay attention to the return value of system(), even if we basically ignore it.
................
r160171 | seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 line
Silence a build warning. (chan_phone.c:810: warning: value computed is not used)
................
r160172 | seanbright | 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines
Merged revisions 159976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines
Get rid of the useless format string and argument in the Bogus/ manager channelname.
Noted by kpfleming and name Bogus/manager suggested by eliel
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@160175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r160004 | russell | 2008-12-01 11:34:31 -0600 (Mon, 01 Dec 2008) | 14 lines
Merged revisions 160003 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines
Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they
both have the potential to send control frames in the middle of call setup. We
have to wait until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will consider it
invalid, and we'll get stuck in an INVAL/VNAK storm.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@160006 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/1.6.1@159851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, 25 Nov 2008) | 23 lines
Merged revisions 159316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines
(closes issue #12694)
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@159375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r158694 | mvanbaak | 2008-11-22 17:57:11 +0100 (Sat, 22 Nov 2008) | 8 lines
dont send reorder tone after a device is hungup if a dialout is abandoned or failed.
Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box
(closes issue #13948)
Reported by: wedhorn
Patches:
switch.diff uploaded by wedhorn (license 30)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158701 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/1.6.1@158268 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/1.6.1@158264 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/1.6.1@158232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov 2008) | 13 lines
Merged revisions 157859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157975 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/1.6.1@157531 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/1.6.1@157498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
