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1908 Commits

Author SHA1 Message Date
Steve Murphy
7d5a79a0b9 This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-09 18:32:07 +00:00
Joshua Colp
a85156a737 Merged revisions 60213 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines

Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-05 12:55:02 +00:00
Russell Bryant
0b94c7c27c Add a Content-Length of 0 to the response built by transmit_response_with_unsupported(). 
(issue #9454, reported by makoto, fixed by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-04 16:49:45 +00:00
Russell Bryant
3e058dab84 Merged revisions 60083 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines

Fix the return value of handle_common_options() so that it always properly
indicates whether it handled the option or not.  
(issue #9455, reported by Netview, fixed by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-04 16:39:04 +00:00
Russell Bryant
dff65e38f4 Merged revisions 59938 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines

Don't attempt to report configuration errors in build_user().  oej pointed out
that for a "friend" entry, this won't work, because all user options are valid
for peers, but not the other way around.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-03 19:16:53 +00:00
Russell Bryant
f8773a1e80 Merged revisions 59916 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines

Make chan_sip report when it encounters an unknown option.
(issue #9440, reported by nightcrawler)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-03 18:55:57 +00:00
Russell Bryant
a2fc7b9e36 Fix the check that ensures that the CHANNEL function's first argument is "rtpqos". 
Thanks, Corydon.  :)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-27 18:17:47 +00:00
Russell Bryant
c0aca26f30 Convert the RTPQOS function to just be additional parameter of the CHANNEL 
function.  This way, it will be possible for other RTP based channel drivers
to expose this information in the future.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-27 16:20:53 +00:00
Tilghman Lesher
0220349ca4 Oops, this should be case insensitive 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-26 21:45:44 +00:00
Tilghman Lesher
ec028bd1b2 Change this to a single dp function to make oej happy. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-26 21:37:41 +00:00
Russell Bryant
cd3238d543 Rename the new dialplan functions to match the variable name 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-26 17:53:07 +00:00
Russell Bryant
fa97f6c381 The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some 
because they get set in sip_hangup.  So, there are common situations where
the variables will not be available in the dialplan at all.  So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-26 17:45:55 +00:00
Joshua Colp
3b6f376425 Merged revisions 59194 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines

Only try to handle a response if it has a response code. (ASA-2007-011)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-24 01:39:44 +00:00
Kevin P. Fleming
cf58b2ba46 don't allow string input to overrun the buffer to hold it (ASA-2007-010) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-22 23:40:01 +00:00
Joshua Colp
067876afbc Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-21 03:25:48 +00:00
Olle Johansson
156c178143 Issue #9313, Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-18 20:37:06 +00:00
Russell Bryant
4fc7d081d3 Some people like to put "limitonpeer" instead of "limitonpeers" in their 
configuration.  While we're at it, support "limitonpeerz" and 
"limitonpeerssssss".  (inspired by issue #9172)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-14 19:18:08 +00:00
Olle Johansson
0eda92609d Merged revisions 58847 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines

Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-13 11:49:35 +00:00
Olle Johansson
f546453573 Don't hangup the call on OK or errors on MESSAGE and INFO 
inside of a dialog (like video update requests).


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-13 10:03:03 +00:00
Olle Johansson
83bb6718fc Issue #9251 - Clear From URI from user attributes (tgrman) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-13 09:12:16 +00:00
Joshua Colp
fa866efb5c Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-12 00:51:16 +00:00
Joshua Colp
b9124234e9 Merged revisions 58579 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines

If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-09 20:49:47 +00:00
Joshua Colp
240355d78b Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-07 17:52:58 +00:00
Steve Murphy
81ef9a4f18 Merged revisions 58115 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line

Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-06 23:10:14 +00:00
Olle Johansson
275abf4e08 Merged revisions 58052 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines

Change error message to proper message

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-06 20:37:07 +00:00
Joshua Colp
53b9bc89c0 Merged revisions 57475 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines

If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-02 17:06:52 +00:00
Joshua Colp
4565c1483c Merged revisions 56230 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines

Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-22 18:49:39 +00:00
Olle Johansson
d67ef52159 Move message from verbose to debug 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-22 10:33:55 +00:00
Russell Bryant
9a7d1cc182 Restructure a little bit of code to reduce nesting. There is no functionality 
change here.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-22 01:24:10 +00:00
Russell Bryant
71921a8329 Merged revisions 56010 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines

If we receive a frame that is not in any of the negotiated formats, then drop
it.  (potentially issue #8781 and SPD-12)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-22 00:57:36 +00:00
Joshua Colp
3d95841eb3 Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-21 17:18:19 +00:00
Olle Johansson
c0f5102378 Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-21 08:32:34 +00:00
Joshua Colp
30d30fef76 Return behavior I removed. I did not remember that you could just add a localnet entry to make it work. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-20 23:57:03 +00:00
Joshua Colp
49749e44a2 Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-20 23:08:45 +00:00
Joshua Colp
7a4bed883e Merged revisions 55073 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines

Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-17 01:16:59 +00:00
Olle Johansson
3ca445e34c Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-16 12:06:23 +00:00
Russell Bryant
7bcf1d913a Remove a couple of leftover debug messages 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-13 21:31:22 +00:00
Russell Bryant
b100b69703 If we fail to create the SIP socket, then return -1 from reload_config() so 
that load_module() will return AST_MODULE_LOAD_DECLINE.  Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-13 19:42:00 +00:00
Russell Bryant
93fcd4a354 Change some text to properly state "On Hold", which was already done in trunk. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-10 00:41:57 +00:00
Russell Bryant
7ee02f585d Merge team/russell/sla_rewrite 
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4.  It is now functional and ready for testing.  However, I will be
adding some additional features over the next week, as well.

For information on how to set this up, see configs/sla.conf.sample 
and doc/sla.txt.

In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:

chan_sip:
 - Add the ability to indicate HOLD state in NOTIFY messages.
 - Queue HOLD and UNHOLD control frames even if the channel is not bridged to
   another channel.

linkedlists.h:
 - Add support for rwlock based linked lists.

dial.c:
 - Add the ability to run ast_dial_start() without a reference channel to
   inherit information from.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-10 00:35:09 +00:00
Olle Johansson
e7a0e86756 Formatting 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-05 23:43:59 +00:00
Olle Johansson
8e07358edf Add some comments on queue system behaviour and how it affects the 
SIP channel


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-05 00:18:34 +00:00
Joshua Colp
910898b7be Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-03 21:05:02 +00:00
Olle Johansson
90a4b844a9 Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now 
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-02 00:24:03 +00:00
Joshua Colp
57fe6882ac Merged revisions 53103 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines

Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 22:24:32 +00:00
Joshua Colp
09844a7f1a Merged revisions 53095 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines

Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113) 

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 21:54:28 +00:00
Olle Johansson
97efd0be22 - Clean INC_COUNT flag when we decrement call counter 
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
  before we destroy the dialog


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 21:05:34 +00:00
Olle Johansson
6bb6bba6a3 Cleaning up the devicestate callback function 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 20:28:54 +00:00
Joshua Colp
e86275c11c Fix silly logic. We really want to write UDPTL frames out when the call is up. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 17:37:44 +00:00
Russell Bryant
9aab046002 Merged revisions 53045 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines

Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-31 21:32:08 +00:00