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
................
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
........
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
........
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
................
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
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/1.6.1@157429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r156962 | mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7 lines
Revision 155513 of chan_sip.c in trunk inadvertently
removed a very important line to set the "len" field
for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk
could do no meaningful processing of anything SIP-related
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@156963 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/1.6.1@155469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r155241 | russell | 2008-11-07 08:50:30 -0600 (Fri, 07 Nov 2008) | 4 lines
Fix some code in chan_sip that was intended to unlink multiple objects from a
container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r152877 | russell | 2008-10-30 14:21:53 -0500 (Thu, 30 Oct 2008) | 9 lines
Modify the documentation of the sip_registry struct
- Remove a comment that says that the monitor thread is the only one that
ever touches these objects. This is no longer the case with TCP. Also,
I would eventually like to get the scheduler in its own thread, so this
is just a poor assumption to make.
- Note that reference counting of these objects with respect to scheduler
entries is not complete. There are some leaked references when deleting
scheduler entries.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines
Make the sip_standard_port function more granular by allowing separate
type and port arguments. This is necessary because when building our From
and Contact headers, we need to be absolutely sure that we are placing our
source port there and not the peer's source port.
(closes issue #12761)
Reported by: asbestoshead
Patches:
patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@151513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
