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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
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
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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
1a5dfca2a1
Remove (to quote Rizzo) "useless" variable.
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2007-01-22 19:00:25 +00:00
dcca8f345f
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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
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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
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
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
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
469cf8ab2a
Merged revisions 50468 via svnmerge from
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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
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Merged revisions 50466 via svnmerge from
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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
47c515768c
Merged revisions 50377 via svnmerge from
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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
3246ad09a4
Based on the following patch, changed for trunk...
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Merged revisions 50124 via svnmerge from
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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
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
835e2ed684
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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
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
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
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
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
87b9abc892
Merged revisions 49676 via svnmerge from
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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
a8dc50812f
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
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
f7ee0abcd2
Removing propably accidentally added debug messages sent to verbose channel
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2007-01-02 20:02:37 +00:00
0d340b7d02
Merged revisions 49212 via svnmerge from
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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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2007-01-02 20:01:34 +00:00
9074555e37
- Add error handling to ast_parse_allow_disallow
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- Use this in chan_sip configuration parsing
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2007-01-01 19:48:31 +00:00
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
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
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
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
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
8794577966
Only handle T38 options once
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 20:26:30 +00:00
820d82470a
-Remove "localmask" setting (deprecated in earlier version)
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- Remove "musiconhold" and "musicclass" settings (also deprecated earlier)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 19:57:06 +00:00
85b6138d83
Merged revisions 48988 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006) | 2 lines
make the option actually match the documentation
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2006-12-27 18:34:24 +00:00
29ed493b40
Be politically correct
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 18:02:10 +00:00
da7a35a1cc
Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 16:56:11 +00:00
3713ce7ed1
Cleanup of handle_common_options
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 16:35:41 +00:00
a12dc35eb3
Reset invitestate when sending new invite
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 16:26:02 +00:00
d69e9086cc
Issue #8600 - bogus SDP Content Length in T.38 re-invite
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2006-12-27 16:01:07 +00:00
7f61b822c1
Merged revisions 48964 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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2006-12-26 04:34:07 +00:00
e0f55d2385
Merged revisions 48940 via svnmerge from
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r48940 | russell | 2006-12-24 01:49:31 -0500 (Sun, 24 Dec 2006) | 11 lines
Merged revisions 48939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines
Remove a couple of misplaced dots in log messages. This was reported by
Andrea Spadaccini on the asterisk-dev mailing list.
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2006-12-24 06:51:37 +00:00
d5eef300d8
define a mask SIP_INSECURE sam as for other sets of flags.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-17 12:47:08 +00:00
168deb2366
Issue #8592 - treat 504 as congestion (imported from 1.2/1.4)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-15 10:58:36 +00:00
bf2172faac
Update to latest IANA specs
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2006-12-15 10:54:06 +00:00
666d526aad
Fix various spelling mistakes in comments.
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2006-12-12 22:32:20 +00:00
2b42422940
Handle multiple 487's correctly
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2006-12-06 16:54:27 +00:00
c39f887ac8
Don't send Contact in SIP Messages (imported from 1.2/1.4).
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Reported by Gunnar at Omnitor.
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2006-12-06 12:34:58 +00:00
a44e55a3f3
Constify a bunch of the usage strings for CLI commands.
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2006-12-06 07:23:32 +00:00
22142e1aa0
Cleaning up handle_response a bit. (Imported from 1.4)
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2006-12-02 22:03:14 +00:00
df8c773783
Send CANCEL to call with early media (PROGRESS INBAND).
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This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
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2006-12-02 20:30:58 +00:00
fc6de9e982
Invitestate updates
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2006-12-02 19:11:02 +00:00
c23bc46089
- Disable RTP timeouts during T.38 transmission
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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2006-12-02 12:05:40 +00:00
Joshua Colp
Olle Johansson
Russell Bryant
Tilghman Lesher
Kevin P. Fleming
Luigi Rizzo