In the course of this, I also found that the results of ast_gethostbyname
were being used incorrectly in both chan_iax2 and chan_sip, so both have
been fixed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed walking the iflist so it is always done with the iflock locked.
* Simplified iflist walking routines.
* Created chan_dahdi iflist insertion and extraction routines.
* Fixed duplicate_pseudo() malloc fail handling.
* Fixed infinite loop in action_dahdishowchannels() when showing a single channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines
IAX2 encryption regression
The IAX2 Call Token security patch inadvertently broke the use of
encryption due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must first
be decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this, decryption
of full frames is once again done before looking into the frame. This
involves searching for an existing callno, checking the pvt to see if
encryption is turned on, and decrypting the packet before the internal
fields of the full frame are accessed.
(closes issue #15834)
Reported by: karesmakro
Patches:
iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, karesmakro
Review: https://reviewboard.asterisk.org/r/355/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes the problem that would occur if an analog phone was picked up while
the caller id was being sent. The caller id before sent the whole spill even
after pickup and is now corrected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r217735 | oej | 2009-09-10 21:52:19 +0200 (Tor, 10 Sep 2009) | 4 lines
Reinstate muted that was removed by mistake.
muted doesn't compile any more on os/x, so I have to disable it in order to testcompile other code...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r217668 | oej | 2009-09-10 21:07:24 +0200 (Tor, 10 Sep 2009) | 9 lines
Remove harmful code that causes endless loops.
Remove code that causes loops in registrations.
We have agreed that the patch that this code was part of was bad. I am ripping out the code that causes
the issue. putnopvut needs to check the rest of the patch, if it needs to be changed as well.
This solves the issue reported in #15540, but needs more work before we close it (as described above).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Related to #12713
Patch by oej
A big thank you to file for finally fixing the transfer() dialplan application.
I've been waiting for years for this. Great work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
gcc 4.4 has more strict rules for aliasing. It doesn't like a
struct sockaddr_in pointer pointing to a struct sockaddr. So we make it
a union.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In passing, correct the formatting of the Timestamp attribute so that there is a
space after the colon and before the value.
(closes issue #15861)
Reported by: Ivan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines
When MOH is playing on the channel, announcements sent through the conference are not heard.
(closes issue #14588)
Reported by: voipas
Patches:
20090716__issue14588__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, twisted, tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This problem was introduced when the AST_FRIENDLY_OFFSET patch was merged.
I'm surprised that nobody noticed any trouble when testing that patch, but this
fixes the code that fills in the buffer to start filling in after the offset
portion of the buffer.
(closes issue #15850)
Reported by: 99gixxer
Patches:
issue15850.diff1.txt uploaded by russell (license 2)
Tested by: 99gixxer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Don't write asterisk.conf from scratch. Fix the existing one.
* Pass extra 'make' command-line arguments to 'install' and 'samples'.
* Fix some extra typos.
closes issue #15019 .
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.
(closes issue #15839)
Reported by: ebroad
Patches:
blank_cidv2.patch uploaded by ebroad (license 878)
parse_uri_fix.diff uploaded by dvossel (license 671)
Tested by: ebroad, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Remove unused string in sip_registry -- "random"
- Someone added a function in the middle of all forward declarations... Weird. Moved it out of that
section.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Add variable for number of known media streams instead of hardcoding in definition of sip_pvt
- Rename "text" to "codecs" - beacuse it's what it is
- Add documentation for future developers so that we make sure that we define new sdp media types
for SRTP-variants
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
