After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89454 f38db490-d61c-443f-a65b-d21fe96a405b
with the SIPPEER() argument of the same name. The deprecation procedure is not
being used here since this is a trunk-only option.
(closes issue #11307, reported by pj, patched by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89441 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89406 f38db490-d61c-443f-a65b-d21fe96a405b
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 :-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89404 f38db490-d61c-443f-a65b-d21fe96a405b
(this should be handled with the dependencies generated by
configure and menuselect, but will be fixed later)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89366 f38db490-d61c-443f-a65b-d21fe96a405b
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.).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
so it can be reused in the implementation of cli commands when
they have a similar syntax.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89320 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89285 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89278 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89268 f38db490-d61c-443f-a65b-d21fe96a405b
Also fix a common typo I kept seeing (arguement) in various files.
Closes issue #11222, patch by snuffy (with arguement > argument by me).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89202 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89179 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89178 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89176 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89174 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89120 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89108 f38db490-d61c-443f-a65b-d21fe96a405b
- 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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89100 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89098 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89091 f38db490-d61c-443f-a65b-d21fe96a405b
- 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.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines
After seeing crashes related to channel variables, I went looking around at the
ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88934 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines
When traversing the list of channel variables here in transmit_invite(), the
asterisk channel must be locked, as this data may change at any time.
(I have seen numerous reports of crashes related to the handling of channel
variables. There are a couple of issues on the bug tracker related to it,
but it has also been noted on IRC and mailing lists. So, I am finding and
fixing some places where channel variables are handled improperly.)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88769 f38db490-d61c-443f-a65b-d21fe96a405b
details and examples are in include/asterisk/stringfields.h.
Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88454 f38db490-d61c-443f-a65b-d21fe96a405b
This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88368 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 lines
Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context).
(closes issue #10702)
Reported by: slavon
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88367 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines
If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive.
(closes issue #5747)
Reported by: chandi
Patches:
infofix-81430-1.patch uploaded by IgorG (license 20)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88329 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | 11 lines
Merge the changes from team/russell/iax2_poke_fix and iax2-poke-fix-trunk
There was a race condition related to the handling of POKEing peers. Essentially,
a reference to a peer is held by the scheduler when there are pending callbacks,
but the reference count didn't reflect it. So, it was possible for a peer to hit
a reference count of zero and have its destructor begin to be called at the same
time that the scheduler thread ran a POKE related callback. If that happened,
a crash would likely occur.
(closes issue #11082, closes issue #11094)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87687 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 lines
Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason.
(closes issue #9566)
Reported by: atca_pres
Patches:
bug9566.patch uploaded by oej
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87343 f38db490-d61c-443f-a65b-d21fe96a405b
Also fixes a few cli messages and some minor formatting.
(closes issue #11001)
Reported by: seanbright
Patches:
newcli.1.patch uploaded by seanbright (license 71)
newcli.2.patch uploaded by seanbright (license 71)
newcli.4.patch uploaded by seanbright (license 71)
newcli.5.patch uploaded by seanbright (license 71)
newcli.6.patch uploaded by seanbright (license 71)
newcli.7.patch uploaded by seanbright (license 71)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86534 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 lines
Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote.
(closes issue #11027)
Reported by: ramonpeek
Patches:
11027-1.diff uploaded by ramonpeek (license 266)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86472 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines
If Asterisk is in the middle of shutting down, respond to OPTIONS
with 503 Unavailable.
(closes issue #10994)
Reported by: eserra
Patches:
sip-options-503.patch uploaded by eserra (license 45)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86150 f38db490-d61c-443f-a65b-d21fe96a405b
Don't improperly memset() over an ast_str. This was leftover from before it
got changed to use ast_str.
(closes issue #11003, reported by pj)
(closes issue #10770, reported by yehavi)
(patched by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85944 f38db490-d61c-443f-a65b-d21fe96a405b
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Modifications :
- set bridge type to partial ;
- process media candidates from the remote peer properly.
Now we have Jingle audio, at least between two Asterisk Jingle
clients.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85777 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | 6 lines
Make the default for the srvlookup option to be yes. It doesn't really make
sense for it to default to off. The default configuration file has it on, and
proper RFC behavior, as indicated by a comment in the code, is for it to be on.
So, let's have it on by default to make lives easier.
(closes issue #10954, suggested by jtodd)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85605 f38db490-d61c-443f-a65b-d21fe96a405b
in the Dial command. The 'j' option _must_ be used in conjunction with the 'n'
option.
This feature will allow you to use the existing jitterbuffer implementation to
put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by
putting a local channel in the middle.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85097 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r84783 | russell | 2007-10-05 11:44:21 -0500 (Fri, 05 Oct 2007) | 4 lines
Do deadlock avoidance in a couple more places. You can't lock two channels
at the same time without doing extra work to make sure it succeeds.
(closes issue #10895, patch by me)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84784 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) | 6 lines
Use snprintf instead of sprintf in one place. There is no vulnerability here
due to various buffer sizes around the code, but I still didn't like seeing a
non length-limited copy of data coming off of the wire into a stack buffer, as
this would be a problem in the future if buffer sizes elsewhere got changed or
size limitations removed ...
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84371 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6 lines
Add dist-clean support for subdirs.
Change h323 to only remove the Makefile on a dist-clean, rather than a clean.
This fixes a bug I found with trying to run make after a make clean
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84300 f38db490-d61c-443f-a65b-d21fe96a405b
a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string
(closes issue #10843)
Reported by: Corydon76
Patches:
2007092900_10843.diff uploaded by mvanbaak (license 7)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84173 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) | 5 lines
Add a log message that was requested by the masses in the developer tutorial
session at Astricon. chan_sip did not output any message when a call was
rejected because the extension was not found. This adds a verbose message
(at verbose level 3) to note when this happens.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83942 f38db490-d61c-443f-a65b-d21fe96a405b
No real Jingle implementation being available, testing was made using
two Asterisk servers relaying SIP calls over their Jingle channels:
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Thus, it was possible to test the code in both ways, and make the
Jingle channel comply with the latest specifications. No sound available yet.
Main modifications include :
- modified the 'jingle_candidate' structure and the
'jingle_create_candidates' function according to XEP-0176 ;
- modified the 'jingle_action' function in order to properly terminate
a Jingle session, in conformance with XEP-0166 ;
- modified username format used in STUN requests ;
- actually make the bindaddr configuration field useable.
Todo :
- set audio paths up (no native bridging) ;
- make the CLI gtalk functions available to jingle ;
- clean up the storage space used in strings.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83743 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83433 f38db490-d61c-443f-a65b-d21fe96a405b
rizzo