Commit Graph

1733 Commits

Author SHA1 Message Date
Jason Parker
cee4bd43dc Rename this instance of "busy limit" to "busy level" as well
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-08 17:19:27 +00:00
Kevin P. Fleming
44c6630e4d rename busy-limit to busy-level, since it is not a limit
actually parse the busy-limit option from sip.conf, instead of ignoring it


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-08 16:41:23 +00:00
Olle Johansson
17af1bd4c8 Merged revisions 53143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3 lines

Add some comments on queue system behaviour and how it affects the
SIP channel

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-05 00:30:03 +00:00
Joshua Colp
014feba426 Merged revisions 53138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines

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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-03 21:06:36 +00:00
Joshua Colp
ce8a7c3d9c Add onHold value to sip show inuse as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-02 18:21:46 +00:00
Olle Johansson
cfe66e6b26 Patch based on this patch with small changes for trunk...
Merged revisions 53109 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines

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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-02 00:26:25 +00:00
Joshua Colp
44a9af3576 Merged revisions 53104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines

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/trunk@53105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 22:26:11 +00:00
Joshua Colp
bf66c620c3 Merged revisions 53097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines

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/trunk@53098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 21:56:23 +00:00
Olle Johansson
544f414c0d Merged revisions 53085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 lines

- 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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 21:17:08 +00:00
Olle Johansson
0b84b386b9 Implementing "busy-limit".
If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).

If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled. 

This affects SIP subscriptions, call queues and manager applications.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 20:43:49 +00:00
Olle Johansson
34eaa61700 Merged revisions 53079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2 lines

Cleaning up the devicestate callback function

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 20:33:59 +00:00
Olle Johansson
38b87ec4b7 Signal HOLD status to phones that subscribe for status.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 19:04:47 +00:00
Joshua Colp
e88dda8ca9 Merged revisions 53064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2 lines

Fix silly logic. We really want to write UDPTL frames out when the call is up.

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2007-02-01 17:42:08 +00:00
Russell Bryant
b233892198 Merged revisions 53046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines

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/trunk@53047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-31 21:35:15 +00:00
Russell Bryant
d11f8b7ccd Merged revisions 52952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | 5 lines

Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO.  This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-30 19:36:28 +00:00
Joshua Colp
1fc144435d Use provided variable for name instead of one in the structure since the structure was just allocated and will be NULL. (issue #8938 reported by st41ker)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-30 15:39:09 +00:00
Joshua Colp
300f980223 Use atomic operation functions for use/ringing/hold manipulation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-25 18:10:18 +00:00
Joshua Colp
b8d6cbcd3f Merged revisions 52210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2 lines

Drop out variables I accidentally put in.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-25 17:51:35 +00:00
Joshua Colp
afb9151e19 Merged revisions 52208 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2 lines

Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-25 17:17:56 +00:00
Joshua Colp
2396e24e65 Merged revisions 52016 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2 lines

Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc)

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2007-01-24 18:04:47 +00:00
Olle Johansson
273f1d78c7 Merged revisions 51931 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3 lines

Show capabilities *and* preference in general settings in "sip show settings"
(reported by Clona/Telio - Thanks!)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24 09:42:31 +00:00
Joshua Colp
ee3ab150f6 Merged revisions 51788 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines

Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-23 22:59:55 +00:00
Olle Johansson
ef4db783c6 Issue #8817 - Registry corruption when packet retransmits fail. (tootai, patchy by oej)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-23 15:36:01 +00:00
Joshua Colp
5a25c156c6 Merged revisions 51558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2 lines

Only change audio formats on the channel if we have an audio format to change to. (issue #8535 reported by ivoc)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-23 03:02:09 +00:00
Olle Johansson
1a5dfca2a1 Remove (to quote Rizzo) "useless" variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-22 19:00:25 +00:00
Russell Bryant
dcca8f345f Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines

Merge the changes from the /team/group/vldtmf_fixup branch.

The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged.  So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio.  However,
since there was no audio coming in, the DTMF_END was never generated.  This
caused DTMF based features to no longer work.

To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf).  If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.

Channel drivers also now get passed the length of the digit to their digit_end
callback.  This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.

(issue #8597, maybe others...)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 18:06:03 +00:00
Joshua Colp
1a06a58250 Merged revisions 51243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2 lines

Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113)

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2007-01-18 18:39:21 +00:00
Russell Bryant
c56f9184e8 Merged revisions 51198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51198 | russell | 2007-01-17 15:18:35 -0600 (Wed, 17 Jan 2007) | 11 lines

Merged revisions 51197 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | 3 lines

Move the check for a failure of ast_channel_alloc() to before locking the
pvt structure again.  Otherwise, on a failure, this will cause a deadlock.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-17 21:20:22 +00:00
Joshua Colp
91dfb27494 Get rid of unneeded code, fix a spelling mistake, and use registry state a bit more. (issue #8402 reported by rizzo)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-13 04:56:25 +00:00
Joshua Colp
0d781f9553 Ensure error variable is set to 0 or else we might get false error messages. (issue #8798 reported by tootai, fix by anthonyl)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-13 03:37:10 +00:00
Joshua Colp
469cf8ab2a Merged revisions 50468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2 lines

Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing.

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2007-01-11 05:54:14 +00:00
Joshua Colp
4942fd94d2 Merged revisions 50466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines

Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson)

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2007-01-11 05:21:03 +00:00
Joshua Colp
47c515768c Merged revisions 50377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2 lines

Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker)

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2007-01-10 18:34:45 +00:00
Olle Johansson
3246ad09a4 Based on the following patch, changed for trunk...
Merged revisions 50124 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3 lines

- handle re-invites properly in sip_hangup()
- Add some invitestate status changes just to be sure

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2007-01-09 12:25:33 +00:00
Olle Johansson
5d4f272a90 Merged revisions 50006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines

Issue #8677 - Handle failure of T.38 re-invite

This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).

To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.

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2007-01-08 14:31:16 +00:00
Olle Johansson
835e2ed684 Merged revisions 49983 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3 lines

Issue #8524, support multiple via header values (tardieu)
	Thanks!

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2007-01-08 13:34:56 +00:00
Olle Johansson
975dba04bc Merged revisions 49945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2 lines

We only need one forward declaration

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2007-01-08 09:22:53 +00:00
Olle Johansson
7158f48a79 Merged revisions 49925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2 lines

Issue 8735: Terminate state when extension is unavailable for subscription

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2007-01-08 08:57:35 +00:00
Tilghman Lesher
4866a44535 Merged revisions 49831 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007) | 2 lines

Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747)

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2007-01-07 21:26:43 +00:00
Kevin P. Fleming
cd73a483f1 const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way here
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2007-01-05 23:32:42 +00:00
Kevin P. Fleming
87b9abc892 Merged revisions 49676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007) | 2 lines

reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases

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2007-01-05 22:43:18 +00:00
Kevin P. Fleming
a8dc50812f Merged revisions 49636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r49636 | kpfleming | 2007-01-05 11:09:00 -0600 (Fri, 05 Jan 2007) | 10 lines

Merged revisions 49635 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines

ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly

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2007-01-05 17:10:59 +00:00
Olle Johansson
f7ee0abcd2 Removing propably accidentally added debug messages sent to verbose channel
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2007-01-02 20:02:37 +00:00
Olle Johansson
0d340b7d02 Merged revisions 49212 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r49212 | oej | 2007-01-02 20:58:45 +0100 (Tue, 02 Jan 2007) | 2 lines

Small cleanup of add_t38sdp - it's always enabled at that point in the code

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-02 20:01:34 +00:00
Olle Johansson
9074555e37 - Add error handling to ast_parse_allow_disallow
- Use this in chan_sip configuration parsing


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-01 19:48:31 +00:00
Olle Johansson
f83b845f08 - Implement error handling in ast_append_ha
- Use this in chan_sip
- Document ha functions in acl.c


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-01 19:20:46 +00:00
Olle Johansson
7674260e75 Add version number to useragent string - Issue #8700, blanchet - THANKS!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-31 09:31:23 +00:00
Olle Johansson
fc86ea2eed Removing extra output
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-29 11:02:28 +00:00
Olle Johansson
b54feaffec Only include include files once (imported from 1.4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 21:27:57 +00:00
Olle Johansson
906a51a1af Only set "rfc2833compensate" option once
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 20:30:03 +00:00