Jason Parker
cee4bd43dc
Rename this instance of "busy limit" to "busy level" as well
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2007-02-08 17:19:27 +00:00
Kevin P. Fleming
44c6630e4d
rename busy-limit to busy-level, since it is not a limit
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actually parse the busy-limit option from sip.conf, instead of ignoring it
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2007-02-08 16:41:23 +00:00
Olle Johansson
17af1bd4c8
Merged revisions 53143 via svnmerge from
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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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2007-02-05 00:30:03 +00:00
Joshua Colp
014feba426
Merged revisions 53138 via svnmerge from
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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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2007-02-03 21:06:36 +00:00
Joshua Colp
ce8a7c3d9c
Add onHold value to sip show inuse as well.
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2007-02-02 18:21:46 +00:00
Olle Johansson
cfe66e6b26
Patch based on this patch with small changes for trunk...
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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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2007-02-02 00:26:25 +00:00
Joshua Colp
44a9af3576
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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
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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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2007-02-01 22:26:11 +00:00
Joshua Colp
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Merged revisions 53097 via svnmerge from
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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
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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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2007-02-01 21:56:23 +00:00
Olle Johansson
544f414c0d
Merged revisions 53085 via svnmerge from
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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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2007-02-01 21:17:08 +00:00
Olle Johansson
0b84b386b9
Implementing "busy-limit".
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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.
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2007-02-01 20:43:49 +00:00
Olle Johansson
34eaa61700
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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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2007-02-01 20:33:59 +00:00
Olle Johansson
38b87ec4b7
Signal HOLD status to phones that subscribe for status.
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2007-02-01 19:04:47 +00:00
Joshua Colp
e88dda8ca9
Merged revisions 53064 via svnmerge from
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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
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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
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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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2007-01-31 21:35:15 +00:00
Russell Bryant
d11f8b7ccd
Merged revisions 52952 via svnmerge from
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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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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)
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2007-01-30 15:39:09 +00:00
Joshua Colp
300f980223
Use atomic operation functions for use/ringing/hold manipulation.
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2007-01-25 18:10:18 +00:00
Joshua Colp
b8d6cbcd3f
Merged revisions 52210 via svnmerge from
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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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2007-01-25 17:51:35 +00:00
Joshua Colp
afb9151e19
Merged revisions 52208 via svnmerge from
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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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2007-01-25 17:17:56 +00:00
Joshua Colp
2396e24e65
Merged revisions 52016 via svnmerge from
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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
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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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2007-01-24 09:42:31 +00:00
Joshua Colp
ee3ab150f6
Merged revisions 51788 via svnmerge from
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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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2007-01-23 22:59:55 +00:00
Olle Johansson
ef4db783c6
Issue #8817 - Registry corruption when packet retransmits fail. (tootai, patchy by oej)
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2007-01-23 15:36:01 +00:00
Joshua Colp
5a25c156c6
Merged revisions 51558 via svnmerge from
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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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2007-01-23 03:02:09 +00:00
Olle Johansson
1a5dfca2a1
Remove (to quote Rizzo) "useless" variable.
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2007-01-22 19:00:25 +00:00
Russell Bryant
dcca8f345f
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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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2007-01-19 18:06:03 +00:00
Joshua Colp
1a06a58250
Merged revisions 51243 via svnmerge from
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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
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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
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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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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)
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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)
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2007-01-13 03:37:10 +00:00
Joshua Colp
469cf8ab2a
Merged revisions 50468 via svnmerge from
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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
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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
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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...
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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
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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
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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
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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
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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
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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
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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
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Merged revisions 49636 via svnmerge from
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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
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-02 20:02:37 +00:00
Olle Johansson
0d340b7d02
Merged revisions 49212 via svnmerge from
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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
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- 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
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- 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!
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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
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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)
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 20:30:03 +00:00