There was a problem in the function responsible for doing peer matching by
IP address and port number such that during the second pass for checking for
a peer configured with insecure=port, it would end up treating every peer as
if it had been configured that way. These changes fix the logic in the peer
IP and port comparison callback to handle insecure=port checking properly.
This problem was introduced when SIP peers were converted to astobj2. Many
thanks to dvossel for noticing this while working on another peer matching
issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For new readers: The janitor list is a list of tasks we need help with in the Asterisk project. Taking up
one of these is often a good way to get into Asterisk development and getting a lot of developers in
the project to be grateful. It's stuff we could spend time on when the bug tracker is empty, when our
employers hasn't filled our task lists and our servers is running bugfree and happily without any issues.
If you want to start working on one of these small projects, feel free to ask for help in the #asterisk-dev
channel on IRC or asterisk-dev mailing list. We'll be more than happy to help you to start and reach
goal.
Thank you for your help.
</end of long commit message>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi
monitoring loop so DTMF CID can be detected without the need of a polarity
change precursor.
(closes issue #9096)
Reported by: fleed
Patches:
9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a document describing the language prompt submission process,
licensing terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language codes with
any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts
can be named with gender, customer/company, etc. suffices as well.
(closes issue #15771)
Reported by: jtodd
Patches:
language-criteria.txt uploaded by jtodd
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This did not function in the way that was intended, causing more compatibility
issues than it solved. It is best, therefore, that it be simply removed.
(Discussed with kpfleming; agreement to remove was reached.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines
Re-send non-100 provisional responses to prevent cancellation
From section 13.3.1.1 of RFC 3261:
If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction. A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction. To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.
(closes issue #11157)
Reported by: rjain
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/315/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are several instances where a port is parsed
from a uri or some other source and converted to
an int value using atoi(), if for some reason the
port string is empty, then a standard port is used.
This logic is used over and over, so I created a function
to handle it in a safer way using sscanf().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If we had this from the start, debugging the 'parking not using configured parkinglot'
bug would have been easier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Init the parkings list member of struct parkinglot.
Thanks Sean for the explanation why this should be here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
made analog_set_linear_mode return back the mode that was being overwritten
so it could be restored later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Now, the scheme passed to parse_uri can either be a
single scheme, or a list of schemes ',' delimited.
This gets rid of the whole problem of having to create
two buffers and calling parse_uri twice to check for
separate schemes.
Review: https://reviewboard.asterisk.org/r/343/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
keep-alive events are used by Sipura/Linksys for NAT keepalive.
There currently don't appear to be any problems with NAT, but
everytime a keep-alive event is received, Asterisk responds with a
"489 Bad event". This error may indicate to a user that NAT
problems exist just because this even is not supported. Now,
rather than respond with an error, the packet is consumed and
a "200 ok" is sent just to indicate we received the packet.
(issue #15084)
Patches:
chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer
into the newly created channel.
(closes issue #15538)
Reported by: gracedman
Patches:
2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7)
With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call)
Tested by: gracedman, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines
Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names
In general channel names are in the form Foo/Bar-Z, but the channel name
could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to
truncate the channel name at the last hyphen.
(closes issue #15810)
Reported by: dhubbard
Patches:
dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames returned by
read(). However, it lied. This means that other parts of the code that
attempted to make use of the offset buffer would end up corrupting the fields
in the ast_filestream structure. This resulted in quite a few crashes due to
unexpected values for fields in ast_filestream.
This patch closes out quite a few bugs. However, some of these bugs have been
open for a while and have been an area where more than one bug has been
discussed. So with that said, anyone that is following one of the issues
closed here, if you still have a problem, please open a new bug report for the
specific problem you are still having. If you do, please ensure that the bug
report is based on the newest version of Asterisk, and that this patch is
applied if format_mp3 is in use. Thanks!
(closes issue #15109)
Reported by: jvandal
Tested by: aragon, russell, zerohalo, marhbere, rgj
(closes issue #14958)
Reported by: aragon
(closes issue #15123)
Reported by: axisinternet
(closes issue #15041)
Reported by: maxnuv
(closes issue #15396)
Reported by: aragon
(closes issue #15195)
Reported by: amorsen
Tested by: amorsen
(closes issue #15781)
Reported by: jensvb
(closes issue #15735)
Reported by: thom4fun
(closes issue #15460)
Reported by: marhbere
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
One note on defining _POSIX_C_SOURCE: while this feature test macro
works to require certain behaviors on Linux, it works differently on *BSD
platforms to REMOVE certain API calls that are not in the POSIX specification,
such as vasprintf(3). Thus, defining it while depending upon vasprintf (and
other extensions to the POSIX standard) to be defined is a recipe to ensure
that Asterisk is only buildable on Linux.
Hence, this define which was meant to INCREASE portability, effectively
ensures the opposite.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines
Modify comment to be a bit more accurate.
We have kept this comment around long enough, that it's pretty clear that we're
keeping the code, because changing the code would require a pretty fundamental
architectural shift. We've also taken criticism in some quarters, because it
was believed that it was referring to the code being nasty. No, the code isn't
nasty, just the operation itself is rather odd. Fixed for eternity (probably
not).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Cross-compilation environments want to provide 'defaults' for compiler and
linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script. This patch
modifies the configure script and Makefile to preserve these settings and
ensure they are used in the build process.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #15786)
Reported by: a_villacis
Patches:
asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch uploaded by a villacis (license 660)
(Plus a few of my own, to catch the remaining places within manager.c where it could have been a problem)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
