Steve Murphy
81ef9a4f18
Merged revisions 58115 via svnmerge from
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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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19 years ago
Olle Johansson
275abf4e08
Merged revisions 58052 via svnmerge from
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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
19 years ago
Joshua Colp
53b9bc89c0
Merged revisions 57475 via svnmerge from
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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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19 years ago
Joshua Colp
4565c1483c
Merged revisions 56230 via svnmerge from
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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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19 years ago
Olle Johansson
d67ef52159
Move message from verbose to debug
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19 years ago
Russell Bryant
9a7d1cc182
Restructure a little bit of code to reduce nesting. There is no functionality
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change here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
71921a8329
Merged revisions 56010 via svnmerge from
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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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19 years ago
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
c0f5102378
Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
30d30fef76
Return behavior I removed. I did not remember that you could just add a localnet entry to make it work.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
7a4bed883e
Merged revisions 55073 via svnmerge from
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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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19 years ago
Olle Johansson
3ca445e34c
Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted.
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19 years ago
Russell Bryant
7bcf1d913a
Remove a couple of leftover debug messages
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19 years ago
Russell Bryant
b100b69703
If we fail to create the SIP socket, then return -1 from reload_config() so
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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
19 years ago
Russell Bryant
93fcd4a354
Change some text to properly state "On Hold", which was already done in trunk.
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19 years ago
Russell Bryant
7ee02f585d
Merge team/russell/sla_rewrite
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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
19 years ago
Olle Johansson
e7a0e86756
Formatting
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19 years ago
Olle Johansson
8e07358edf
Add some comments on queue system behaviour and how it affects the
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SIP channel
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19 years ago
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
90a4b844a9
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
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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
19 years ago
Joshua Colp
57fe6882ac
Merged revisions 53103 via svnmerge from
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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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19 years ago
Joshua Colp
09844a7f1a
Merged revisions 53095 via svnmerge from
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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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19 years ago
Olle Johansson
97efd0be22
- Clean INC_COUNT flag when we decrement call counter
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- 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
19 years ago
Olle Johansson
6bb6bba6a3
Cleaning up the devicestate callback function
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
e86275c11c
Fix silly logic. We really want to write UDPTL frames out when the call is up.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
9aab046002
Merged revisions 53045 via svnmerge from
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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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19 years ago
Russell Bryant
29b7393d84
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
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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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19 years ago
Joshua Colp
ed48c69f06
Drop out variables I accidentally put in.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
6b08afd05d
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/branches/1.4@52208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
5ebd1ecf63
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
a207a31a97
Show capabilities *and* preference in general settings in "sip show settings"
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(reported by Clona/Telio - Thanks!)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
8f7ddbef0d
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/branches/1.4@51788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
5a3acb0511
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/branches/1.4@51558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
33235b40d6
Merge the changes from the /team/group/vldtmf_fixup branch.
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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...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
1e3557c636
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
4244459e31
Merged revisions 51197 via svnmerge from
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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/branches/1.4@51198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
915f9315e1
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
240ca25bea
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
9aca2b2a54
Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
1a33c38a15
- handle re-invites properly in sip_hangup()
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- Add some invitestate status changes just to be sure
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
3394598f93
Issue #8677 - Handle failure of T.38 re-invite
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
0f96f73768
Issue #8524 , support multiple via header values (tardieu)
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Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
be32fad9b8
We only need one forward declaration
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19 years ago
Olle Johansson
484add6554
Issue 8735: Terminate state when extension is unavailable for subscription
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Tilghman Lesher
dcbf36432e
Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
444adcb477
reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
fb010e49aa
Merged revisions 49635 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
5edb7fa173
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/branches/1.4@49212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
7db2ca152c
remove incomplete implementation of dnsmgr. Let's fix this in trunk.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
1e9c141c2d
Only include acl.h and lock.h once
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
4f157be79e
Only set rfc2833compensate flag once (handle_common_options)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
f1f8bbaabe
- Remove checking for T38 options twice. Keeping them in handle_common_options
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19 years ago
Kevin P. Fleming
0f5aae9688
make the option actually match the documentation
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
d2b7e8b247
Be a bit more politically correct
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
bfe4bb0f1e
Issue #8575 - Buggy cisco MWI support.
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Normally we try not to change our software for bugs in other devices. But in
this case, the Cisco phones are so widespread so we try to implement a fix while
waiting for a bugfix from Cisco.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
e25756dfac
- Make sure handle_common_options return 1 when we found a common option
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- Move uncommon (only global) option away from handle_common_options
Reported by rizzo. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
4ea530f2dc
Issue 8599 (rizzo) Change invitestate before re-sending invite with auth.
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19 years ago
Olle Johansson
8a42650605
Fix bogus content-length in t38 sdp. (rizzo, #8600 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
9cc04e026d
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
1208869c00
Merged revisions 48939 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
c1f1cf48ac
Issue #8592 - treat 504 as 503 (imported from 1.2)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
a8fe77b95b
Update to latest IANA spec
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
28c36660c7
Issue #8258 - fix handling of 487 being retransmitted to Asterisk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
0fe91149ae
Don't send Contact on MESSAGE
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
694205de93
Merging the invitestate-1.4 branch after successful testing.
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Will check if I can solve this with less changes in 1.2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
9636bccb7c
- Removing one of two pieces of code to handle 481 response on INVITE
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- Move handling of REFER response to handle_response_refer()
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
f89143bd13
- Disable RTP hold timers while T.38 fax transmission happens
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- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
something that video phones support in the RTP stream.
I now this is a big architectual change at this stage for 1.4, but decided it was needed
to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample
Issue 7679 in the bug tracker. Please test.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
98d3fb64ed
- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states
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- Remove support for T.38 early media, since it's impossible.
(Two patches in one - extra friday evening offer due to being off line from svn today... :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
76e35bac11
Issue 8319 - change noncecount before using it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
073e937757
Clean up bad dialogs properly. Caused by GS 487 adapter without CSEQ on separate line
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in the REGISTER request. Imported from 1.2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
9436ecaad7
Change logging message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
5a3dcaff3c
Not having a HINT is not an ERROR. In 1.4 and future releases, you can disable
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subscription support totally or per peer in sip.conf with allowsubscribe = yes | no
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
c3a7e9d7c2
Don't allocate unused variable.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
5c0be47e69
Apparently Exosip sends a 101 after a 100 provisional response. Let's not treat that as
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early media. (discovered at the AVTF meeting in Paris).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
a7c1ac60b7
Make sure we destroy scheduled items and not use them ever again after destruction (rizzo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
d20e81d829
fix bug 7450 - Parsing fails if From header contains angle brackets
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(the bug was only in a corner case where the < was right after the
opening quote, and the fix is trivial).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
f373cad528
Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
e1e6a1b2a8
Make the HOLD notification optional, in order to avoid a lot of extra database lookups
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for all those realtime users out there.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
d1ee29fc20
Don't fixup if there's nothing to fixup
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
5bd53e3588
- CANCEL is never authenticated (according to the RFC)
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- Update docs on canreinvite. "nonat" is the recommended setting for most users with
phones behind a NAT.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
c78840b900
- Improve SIP history
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- Never send reply to ACK (again...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
63f87d112d
Send error message when we can't allocate SIP dialog, possibly due to limitation of file descriptors.
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(imported from 1.2)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
25b5cf18b7
Only keep the video RTP structure around if 1. Video support is enabled and 2. A video codec is enabled on the dialog
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
4ecb09f6e8
Update documentation to fit the implementation...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
cb82979700
Issue #8272 - Don't destroy dialog in retransmission system if it's an OPTION packet from peerpoke
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Tilghman Lesher
e944561518
Merged revisions 47580 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) | 2 lines
Having more than 255 old messages caused corruption in the new/old count
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Tilghman Lesher
6e4d0e50c8
Re-enable old deprecated commands
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
55c8a5ded8
- Don't reply to INVITE already replied to when we get BYE
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- Declare errmsg as int. Oops.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
0ca5dae550
Only produce error message about sip history once
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
153cb1309f
Issue 8314 - Restore auto-framing (Thanks DEA!!!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
fc5f9c78d1
Part of issue 8078 - parse even if udptl is UDPTL in sdp...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
a471f0a87c
- Don't destroy SIP dialog because of a failed T.38 re-invite. Wait for a bye.
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Final response to a re-invite does not mean that the session dies, only that
the re-invite fails.
- Keep RTP active during processing of T.38 re-invite. If the re-invite fails,
RTP needs to remain as before the re-invite.
Issue 8338 - darren1713. Please test.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
853958e863
-Remove blocking of ptime: parsing in sdp
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-Add some comments to t.38 code
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
2a321d80f5
...and make sure that the dialog is destroyed, even if we don't get any answer on the bye...
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This is the channel that remains dead after the SIP transfer
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
ea257647a2
Add debug output while trying to trace bug in bug report
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
ba58cf9ecd
Make sure we destroy dialog...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
1f2a4eb8c2
Small cleanup of handle_request_invite() - imported from 1.2 with changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
d83a1f1356
Issue 8010 - Fix support for multipart SDP (alphaque)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Tilghman Lesher
f11ea0549d
Discussion of these CLI changes resulted in more consistency (Bug 8236)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
54994dfe04
Rip out half implementation of 491 response support, since it wasn't implemented
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properly and caused memory leaks in the case of us getting 491's, which Asterisk
actually sends...
Since it is a bit too complicated to fix this, I'll rip it out of 1.4 and put
it on the to-do-list for future releases. Now, we handle this as congestion,
which it really is.
Issue #8331
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
27e6ea10fb
Fix bit definition for SIP_PAG2_CALL_ONHOLD. Thanks fenlander!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
10978d0390
add simple fix for SDP to report proper sample rate for G.722 media sessions
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
764ba93e05
Destroy dialog properly at unload (rizzo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Steve Murphy
517978fd5f
These mods are to solve the problem in bug 7506. It's a lot of rework to solve a fairly small problem... such is life.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
647fb52e31
Issue 8303 (lrizzo) - break instead of continue.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
eac4efcf22
Fixing the attack shield so it doesn't produce attacks... Issue 8265 - never reply to an ACK
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
d5edfd432a
Issue #8284 : Fixes to Invite/replaces and transfer from "john"
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Thank you!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Steve Murphy
b6079499e4
This fix introduced via bug 8233
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Tilghman Lesher
e05a2752e8
Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
ea69dd0cbc
Move check for codec translation to sip_call() instead of in add_sdp. No one bothers
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with the result of add_sdp anyway... Yet...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
bfb1d68414
Disable code for T38 over TCP and RTP since there's no trace of actual functionality for it :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
efdffc0cf8
don't send INVITE when we have determined that we can't offer any audio formats due to lack of trancoding support (or incorrect configuration)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
eb25ec6ed5
Merged revisions 46920 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 lines
Repeat after me oej: I will at least make sure my code compiles before I commit it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
5667c1c71d
Dont overwrite pkt->flags (from 1.2)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
9ab1cc22a4
Support ;rport when we're supposed to support ;rport. Issue #7473 .
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
56f2cab81c
If peer fails ACL check, fail peer at REGISTER
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
cdb0c64271
Fix T38 too. Thanks, tgrman !
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
86c973f71f
Issue #8246 - Doxygen fixes from kshumard.
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An extra big thankyou is given to everyone that contributes to doxygen!
THANK YOU!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
9213317979
Issue #7869 - Stop retransmission of 302 redirects (imported from 1.2)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
4456e58811
Issue #7608 - Notifications sent with wrong content-type (imported from 1.2, modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
e8350b836b
Backport of patch for #7828 that was reported for trunk, but obviously
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exists in 1.4 too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
0ff30203f9
Restoring the old logic, since working around it and fixing it seemed too complicated.
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- The SIP_OUTGOING flag indicates the direction of the last transaction in the dialog.
- The initreq stores the last request in the dialog, the request that opened the
latest transaction.
Please now retry all the 1.4 bug reports with mixed to/from headers, tags etc
in ACK, BYE, CANCEL. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
b1b2177079
Accepting a message twice may be misinterpreted...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
5c59f2f5f5
- 183 is not reliable message...
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- Error should not have SDP
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
77e12e971c
Old todo note - Don't add Contact header on BYE and Cancel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
b36a4fad1b
Somewhat ugly code to try to fix issue #7608 .
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Since the problem was not very well defined, the fix is a bit fuzzy too...
Thanks to Luigi for accidentally spotting the possible problem!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
03b4e17d23
show settings doesn't produce a list
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of similar objects, it should stay a "show"
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
ce4b0afb73
apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
05eb71c699
code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
50fda8522d
merge 45966 refer_to_domain potentially containing options
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
d7b0db6530
merge 46026 improper checks on get_header() return values
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
fa4d6f23a6
merge 46045 prevent NULL args to ast_strdupa() in chan_sip.c
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
2681aea6b2
Let's understand SIP:
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- REFER can create dialog, Asterisk does not support it yet
- NOTIFY can create dialog in Asterisk's implementation (voicemail) even though
we don't support the server side of it. In this case, the standard is a side issue ;-)
- Added extened functionality for unsupported methods (PING, PUBLISH) so we don't
create PVT's for those either.
Russellb needs to judge what to do with this in 1.2, but I think the current implementation
n 1.2 is a bug since we're sending bad replies to NOTIFY and REFER outside of dialogs
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
67c180956c
If the jitterbuffer is forced on then we can't partially bridge (reported by wangster on #asterisk-dev)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
a37a95c0d8
Cleaning up
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
87cb020143
Don't leak the actual thread-specific sip_pvt struct
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
2f6a555fbd
don't leak memory when a chan_sip thread is destroyed that has a thread-local temp_pvt allocated
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
9bd68dd32c
remove duplicate declarations
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
227d415709
optimize the 'quick response' code a bit more... no more malloc() or memset() for each response
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expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
0c4dfe3aed
Don't create a "real" pvt structure for requests that shouldn't be able to create one. Instead use a temporary pvt and fill it with enough information so we can send a reply.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
1e2935d447
remove the option that was not needed, really.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
d28fd24747
Merged revisions 45265 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 lines
Use responses rather then replies even though they mean the same thing.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
3f24dceeca
Merged revisions 45260 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines
Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
eaaddb90bf
Don't destroy dialog for unexpected REFER response...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
1bfbc1419a
Complete merging in RPID screen changes (issue #8101 reported by hristo, patch by oej in revision 44757)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
0e230d79d8
merge formatting and minor code simplifications from trunk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
ec4bf13915
we can only send one 'a=ptime' attribute per media session, not one for each format
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
09f6a6a167
Merged revisions 44955 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44955 | kpfleming | 2006-10-12 13:31:26 -0500 (Thu, 12 Oct 2006) | 2 lines
ensure that IAX2 and SIP sockets allow UDP fragmentation when running on Linux (thanks to Brian Candler on the asterisk-dev list for the tip)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
84a6a67b15
Do not set default/global values in the variable declaration, set it in reload_config()
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago