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With this, you can control the peer in the dialplan, so you avoid placing outbound
calls when the device has reached busy-level.
Reported by pj.
Closes bug #11180
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working with
the bug tracker. Since we don't authenticate devices (peers/users) on OPTIONS we don't
have the proper context set for the user/peer.
However, we might not want to process an authentication for every OPTIONS, so we could
have a config option for this, "optionsforceok" to always answer 200 OK on the request
and not check device or destination, nor add a SDP. If Asterisk sends the OPTIONs request,
it doesn't care about the reply. Some devices use OPTIONs to discover capabilities,
since we should answer like an INVITE from the device and we need to support that properly
too, which we don't today.
So much to do :-)
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(this should be handled with the dependencies generated by
configure and menuselect, but will be fixed later)
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Reproduce with
grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr
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but make sure that asterisk/compiler.h is included everywhere
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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so it can be reused in the implementation of cli commands when
they have a similar syntax.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15 Nov 2007) | 2 lines
Fix an uninitialized memory read found by valgrind
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007) | 5 lines
Yet another memory corruption issue.
Reported by: atis
Patch by: tilghman
Fixes issue #10923
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constification, and marking a function in chan_sip as purposely unused until it is fixed up.
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solution, it breaks communication.
Rizzo - you need to implement a configuration option for this
code. It's good, but maybe should be off by default.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 lines
Don't send re-invites during pending INVITE transactions.
Patch by one47 - thanks!
Closes issue #9305
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 lines
Improve support for multipart messages. Code by gasparz, changes
by me (mostly formatting). Thanks, gasparz!
Closes issue #10947
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way of handling DTMF in SIP. Totally undocumented, but implemented
in enough devices so we have to support it.
Code by sergee, small changes by oej.
Closes issue #11049
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ast_variable_new parameters
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This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
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- Fix a minor spelling error in a comment
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) | 2 lines
If we set a value for qualify, we should actually pay attention to it, instead of overriding the value
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Also fix a common typo I kept seeing (arguement) in various files.
Closes issue #11222, patch by snuffy (with arguement > argument by me).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines
Fix two cases of memory corruption caused by background threads.
Reported by: atis
Patch by: tilghman
Fixes issue #10923
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line
if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | 1 line
added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it.
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(closes issue #11218)
Reported by: ys
Patches:
trunk89169.diff uploaded by ys (license 281)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | 1 line
fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12 Nov 2007) | 1 line
fixed the support for CW and therefore for the reject_cause option.
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r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line
aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer.
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r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov 2007) | 7 lines
Rework of the commit I made yesterday to use the already built-in
ast_uri_decode function as opposed to my home-rolled one. Also added
comments.
Thanks to oej for pointing me in the right direction
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The type of warnings emitted depends on the optimization level,
at the lower levels the compiler doesn't always understand what the
programmer has in mind. In this case I could not understand it either.
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines
Do not add a sip: to the beginning of the To URI unless needed.
(closes issue #10756)
Reported by: goestelecom
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines
Improve the devicestate logic for multiple devices. If any are available then the extension is considered available.
(closes issue #10164)
Reported by: nic_bellamy
Patches:
sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)
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r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines
Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support.
(closes issue #10946)
Reported by: flefoll
(closes issue #10915)
Reported by: ramonpeek
(closes issue #9567)
Reported by: atca_pres
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r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines
If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan.
(closes issue #11185)
Reported by: spditner
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r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov 2007) | 6 lines
This patch makes it possible for SIP phones to dial extensions defined with '#' characters
in extensions.conf AND maintain their escaped characters when forming URI's
(closes issue #10681, reported by cahen, patched by me, code review by file)
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Reported by: thetatag
Patch by: thetatag/stevens/tilghman
Closes issue #5331
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- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
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(closes issue #11171, reported and patched by blitzrage)
Many thanks!
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Closes issue #9972
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines
Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable.
(closes issue #11006)
Reported by: pj
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