This patch implements a temporary storage in the pvt and use that instead.
The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!
(closes issue #14294)
related to issue #13385
Reported by: klaus3000 and adomjan
Patches:
bug14294b.diff uploaded by oej (license 306)
Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems.
Thanks Fredrik for pointing out where the bug in the SIP messaging was.
(closes issue #14346)
Reported by: oej
Patches:
bug14346.diff uploaded by oej (license 306)
Tested by: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash
This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER
(closes issue #14211)
Reported by: aborghi
Patches:
14211.diff uploaded by putnopvut (license 60)
Tested by: aborghi
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests
were not checking the topmost Via to determine where
to send the response. Adding check_via calls to those
request handlers solves this.
(closes issue #13071)
Reported by: baron
Patches:
check_via.patch uploaded by baron (license 531)
Tested by: baron
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are some boolean attributes for T.38 such
as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we
should treat these as a "true" value. The current
code, however, was requiring a 1 or 0 as the value
of the attribute in order to parse it. This is due
to the fact that there are some T.38 endpoints and
gateways that also transmit this information
incorrectly. This patch follows the "be liberal in
what you accept and strict in what you send"
philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending
information as it is supposed to be sent.
It was also discovered that a particular type of
T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram
and T38MaxBitRate, it used T38MaxDatagram and
T38FaxMaxRate respectively. We now will properly
accept these attributes as well.
Note that there are a lot of patches cited in
the below commit message template. This is
because the person who submitted these patches is
an awesome person and wrote 1.4, 1.6.0, and 1.6.1
variants.
(closes issue #13976)
Reported by: linulin
Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648)
Tested by: arcivanov
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@164977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@164672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"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/1.4@160480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@160297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@158071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@158053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@157503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/branches/1.4@146799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: MVF
Tested by: neutrino88, urzedo, murf, thiagofernandes
Many thanks to neutrino88 for this patch, which
solves a problem whereby channels get a CANCEL
request, respond to it properly, but end up
in a hung state, infinitely being rescheduled.
This fix is a bit crude, in that catches the
problem at a rather late phase, but it may
prevent infinite rescheduling problems that
might still arise.
It might have been better to find out why,
in the course of protocol handling, the channel
was not destroyed, but we leave that to
future generations.
Many thanks to urzedo and thiagofernandes for
their work in verifying that the patch code
indeed is being executing, and averting the
problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@142865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the media to be natively bridged, use the jointcapability
instead of the peercapability.
It seems that the intent of using the peercapability was to
expand the choice of codecs for the call to increase the
chances of being able to native bridge the channels. The
problem is that if a codec were settled on for the native
bridge and that wasn't a codec that was configured to be used
by Asterisk for that peer, then Asterisk would send a
REINVITE with no codecs in the SDP which is a bug no matter
how you slice it.
(closes issue #13076)
Reported by: ramonpeek
Patches:
13076.patch uploaded by putnopvut (license 60)
Tested by: tbelder
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@142079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
remote tag of an endpoint once a dialog has
been confirmed. Up until that point, it is possible
and legal for the far-end to send provisional
responses with a different To: tag each time. With
this patch applied, these provisional messages
will not cause a matching problem.
(closes issue #11536)
Reported by: ibc
Patches:
11536v2.patch uploaded by putnopvut (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@141809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
caused port 5060 to always be used when dialing
a peer if no explicit port was specified. This
broke the behavior of implicitly using the port
from which the peer registered if no port is
specified. This commit fixes the logic flaw.
(closes issue #13424)
Reported by: mdu113
Patches:
13424.patch uploaded by putnopvut (license 60)
Tested by: mdu113
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@141217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
in a string to Dial(), it is not ignored.
(closes issue #13355)
Reported by: acunningham
Patches:
13355v2.patch uploaded by putnopvut (license 60)
Tested by: acunningham
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@140417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/1.4@140299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
of a completely unrelated function to determine whether the scheduler should
be run or not. This would have caused the scheduler to not run in cases where
it should have. Also, leave a note about another scheduler issue that needs
to be addressed at some point.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@140060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
