This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines
Have IAX2 optimize the codec translation path just like chan_sip does it. If
the caller's codec is in our codec list, move it to the top to avoid transcoding.
(closes issue #10500)
Reported by: stevedavies
Patches:
iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh
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This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Tested by: jtodd, rjain, loloski
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r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines
Fix a deadlock in chan_local in local_hangup. There was contention because
the local_pvt was held and it was attempting to lock a channel, which is the
incorrect locking order.
(closes issue #11730)
Reported by: UDI-Doug
Patches:
11730.patch uploaded by putnopvut (license 60)
Tested by: UDI-Doug
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r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines
Don't drop the old record route information when dealing with packets related to a reinvite.
(closes issue #11545)
Reported by: kebl0155
Patches:
reinvite-patch.txt uploaded by kebl0155 (license 356)
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r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines
Change a buffer in check_auth() to be a thread local dynamically allocated
buffer, instead of a massive buffer on the stack. This fixes a crash reported
by Qwell due to running out of stack space when building with LOW_MEMORY defined.
On a very related note, the usage of BUFSIZ in various places in chan_sip is
arbitrary and careless. BUFSIZ is a system specific define. On my machine,
it is 8192, but by definition (according to google) could be as small as 256.
So, this buffer in check_auth was 16 kB. We don't even support SIP messages
larger than 4 kB! Further usage of this define should be avoided, unless it
is used in the proper context.
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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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Note: NoAnswer support is currently not implemented, as it would take a
significant amount of work to figure out how to do correctly.
Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself.
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to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines
1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running. Don't just ignore it.
(Closes issue #11698)
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1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
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- support scrolling of message window;
- simplify the code for creating a message window,
and try it using a second one in the top of
the keypad (where we echo the dialed number).
The 'skin' that supports these two windows will be
committed separately.
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r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines
Set the caller id within the gtalk_alloc function.
As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.
Closes issue #11549.
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a number to dial in the 'message' area under the
keypad.
Now you can make calls using the keypad as a regular phone
(or the keyboard for chars not present on the keypad)
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commands so you can start and stop the gui even outside
of a call. This is convenient for testing, and also for
using the keypad to pick up a call, and to dial a number
(the latter not yet implemented, but should be close).
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The main code to implement the textarea is in console_board.c,
and uses a simple png image with the font, blitting characters
on the designated areas of the main screen.
Additionally we provide some annotations in the image used
as a skin to indicate which areas are used for text messages.
(images will be committed separately).
At the moment the dialog area is only used to display a running
counter, just as a proof of concept.
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This way it can contain additional elements (e.g. fonts, buttons,
widgets) without having to use a zillion files to store them.
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revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
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r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines
Make use of the temporary channel pointer while the pvt is unlocked.
(closes issue #11675)
Reported by: flefoll
Patches:
chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244)
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