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r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 lines
Correct a copy/pasted error message line for RTCP.
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r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines
When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.
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r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2 lines
Add more frame types to forward in the RTP bridge loops.
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r53040 | russell | 2007-01-31 11:45:05 -0600 (Wed, 31 Jan 2007) | 11 lines
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r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines
Use the proper format string to print unsigned values in the rtp debug output.
(issue #8954 , wmis)
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r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines
Fix a problem with packet-to-packet bridging and DTMF mode translation.  P2P
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions.  Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936 )
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								Only use locking for bridge information if intense P2P bridging is enabled.  
							
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								Change RTP protos list to be read/write. Most of the time it's only going to be read so making it use mutex locks was a waste.  
							
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								39d3580ee4 
								
							
								 
							
						 
						
							
							
								
								Make the RTP stack better conform to coding guidelines. (issue  #8679  reported by johann8384)  
							
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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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								e7c5029d23 
								
							
								 
							
						 
						
							
							
								
								in the interest of portability, avoid using %zd when all  
							
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							we need is to print is an integer that fits in 16 bits.
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r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2 lines
Pass data as well for hold/unhold/vidupdate frames. (issue #8840  reported by mdu113)
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r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2 lines
Return the correct result when directly writing out a packet so that the core doesn't then decide to handle it the regular way again. (issue #8833  reported by rcourtna)
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r51170 | qwell | 2007-01-16 18:20:56 -0600 (Tue, 16 Jan 2007) | 4 lines
Fix issue with dtmf continuation packets when the dtmf digit is 0...
Issue 8831
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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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								Make callback declaration match one used in trunk.  
							
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r50032 | file | 2007-01-08 13:21:31 -0500 (Mon, 08 Jan 2007) | 2 lines
Disable the more intense packet2packet bridging until the bugs can be worked out.
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								68ff3c3575 
								
							
								 
							
						 
						
							
							
								
								Issue  #8663  - Add passthrough support for MPEG4 (neutrino88).  
							
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								e2a50de88f 
								
							
								 
							
						 
						
							
							
								
								Clarify why we are reading in a frame in the Packet2Packet bridge.  
							
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r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2 lines
If the Packet2Packet bridge is being broken because of a masquerade then attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696  reported by kjotte)
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r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006) | 2 lines
since these variables all have static duration, none of them need initializers (they default to zero anyway)
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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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r48506 | file | 2006-12-15 14:55:28 -0500 (Fri, 15 Dec 2006) | 2 lines
Turn payload_lock into bridge_lock and make it encompass all RTP structure contents that may relate to bridge information, including who we are bridged to.
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r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 lines
Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232  reported by tgrman)
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r48461 | file | 2006-12-13 22:33:30 -0500 (Wed, 13 Dec 2006) | 2 lines
Remove direct RTCP bridging. I've come to the conclusion that we should handle this through the core and not just forward it on. Should solve a few bugs.
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r48381 | file | 2006-12-11 00:36:45 -0500 (Mon, 11 Dec 2006) | 2 lines
Merge in my latest RTP changes. Break out RTP and RTCP callback functions so they no longer share a common one.
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								Staticize one, and Constify a bunch of usage strings for CLI commands.  
							
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								fe53552f41 
								
							
								 
							
						 
						
							
							
								
								Doxygen updates  
							
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								00bf07b12e 
								
							
								 
							
						 
						
							
							
								
								Well, yes...  
							
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								b8fcae6d75 
								
							
								 
							
						 
						
							
							
								
								Reserving flags for coming code (currently in the "videocaps" branch)  
							
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							implementing T.140 support in RTP.
T.140/RFC 4351 is TDD over IP - text telephony for hearing impaired.
It defines a realtime text chat, much like the old "talk" application
in Unix. 
T.140 is character by character in real time. It's not 
the same as our current MESSAGE format - that is more like IM, but
can be gatewayed to MESSAGE with a text "codec" if needed.
More patches will follow, as soon as we've separated this code from
the video capabilities functions in the videocaps branch.
Code by John Martin, Aupix (disclaimer on file)
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								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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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448  reported by phsultan)
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								2bee4aba4d 
								
							
								 
							
						 
						
							
							
								
								Change logging for p2p rtp bridge mode  
							
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r48107 | file | 2006-11-29 11:50:33 -0500 (Wed, 29 Nov 2006) | 10 lines
Merged revisions 48106 via svnmerge from 
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r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines
If the frame was duplicated before writing out then we need to free it. (issue #8429  reported by edguy3)
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								7991366506 
								
							
								 
							
						 
						
							
							
								
								- Adding comment on suspicious memory allocation. Seems like it's never freed, but I don't  
							
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							have a clear understanding of the frame allocation/deallocation, so I just mark this
  for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts...
- Doxygen comments on p2p rtp bridge stuff.  I am a bit worried about shortcutting
  rtcp this way, but will need feedback from rtcp gurus. This should work for 
  video calls too, and possibly UDPTL.
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r47944 | file | 2006-11-22 16:47:43 -0500 (Wed, 22 Nov 2006) | 2 lines
Video will never reach Packet2Packet bridging and can do more harm then good.
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r47897 | file | 2006-11-21 12:32:27 -0500 (Tue, 21 Nov 2006) | 2 lines
If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330  reported by voipgate)
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								03a7adf8ce 
								
							
								 
							
						 
						
							
							
								
								Use RTP/RTCP fds on the RTP structure, don't bother storing them.  
							
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r47852 | file | 2006-11-20 10:58:50 -0500 (Mon, 20 Nov 2006) | 2 lines
Only remove/destroy the RTCP I/O item if it exists.
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r47645 | file | 2006-11-14 23:45:24 -0500 (Tue, 14 Nov 2006) | 2 lines
If NAT detection is turned on or already detected then say NAT is active when setting the remote RTP peer when doing early bridging. (issue #8365  reported by marcelbarbulescu)
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								5861048fb6 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 47639 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r47639 | file | 2006-11-14 19:14:07 -0500 (Tue, 14 Nov 2006) | 2 lines
Turn notice about unknown RTCP packet type into a debug message instead.
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								79f75ec09a 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 47053 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006) | 2 lines
More changes making the CLI more consistent with "category verb arguments" (continuation of issue 8236)
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								2cb07fbaa4 
								
							
								 
							
						 
						
							
							
								
								In debug mode, recognize that someone is sending zrtp, even though we  
							
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							can't do anything with it yet. Ideally a first step would be a 
passthrough mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46439  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								52a5d63a2d 
								
							
								 
							
						 
						
							
							
								
								Bind RTCP to the same IP as RTP.  
							
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							I currently don't see this as a bug that needs to be fixed in 1.4/1.2 too,
but feel free to backport if you see it that way. RTCP now binds to
ALL IP addresses on the host, RTP to a specific address.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46409  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0ca6a42d7e 
								
							
								 
							
						 
						
							
							
								
								fix various spelling mistakes in comments (issue  #8237 , jmls)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46339  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
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								88efcea05e 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 46154 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines
add passthrough and file format support for G.722 16KHz audio (issue #5084 , original patch by andrew, updated by mithraen)
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								bb8926d50c 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 45452 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r45452 | file | 2006-10-17 23:02:08 -0400 (Tue, 17 Oct 2006) | 2 lines
Don't segfault if you're using a channel driver that doesn't turn RTCP on
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								62e6417b21 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 44628 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r44628 | file | 2006-10-06 17:08:54 -0400 (Fri, 06 Oct 2006) | 2 lines
Remove the seqno check for RFC2833, the handler is smart enough to not need it.
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								85625f3505 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 44605 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r44605 | file | 2006-10-06 14:46:28 -0400 (Fri, 06 Oct 2006) | 2 lines
When the sequence number rolls over then reset the recorded sequence number for DTMF (issue #8106  reported by bungalow)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44606  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
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								ae8cc3e18b 
								
							
								 
							
						 
						
							
							
								
								bug  #8076  check option_debug before printing to debug channel.  
							
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							patch provided in bugnote, with minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
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								3cea4702a3 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 44090 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r44090 | pcadach | 2006-10-01 01:20:38 +0600 (Вск, 01 Окт 2006) | 1 line
Allow one-way RTP streams (device->Asterisk)
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								6df7c274d8 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 43798 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r43798 | file | 2006-09-27 15:10:59 -0400 (Wed, 27 Sep 2006) | 2 lines
Compensate for out of order packets better if RFC2833 compensation is turned on.
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								04cf782862 
								
							
								 
							
						 
						
							
							
								
								Small Cisco's RTP DTMF update  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43546  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								1d50a8e881 
								
							
								 
							
						 
						
							
							
								
								Correct behavior on Cisco's DTMF  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43539  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								2b55678e1f 
								
							
								 
							
						 
						
							
							
								
								Remove deprecated CLI apps from the core  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43449  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								1c764935f2 
								
							
								 
							
						 
						
							
							
								
								SS7 marked the start of an open season for trunk again but here's something minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43437  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								70af28270d 
								
							
								 
							
						 
						
							
							
								
								Constify the result of a config retrieval function, to avoid mutilation (issue 7983).  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43364  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								3c6d5053ba 
								
							
								 
							
						 
						
							
							
								
								Totally break a P2P bridge upon going on hold, and re-establish it upon going off hold.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43343  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								659d467720 
								
							
								 
							
						 
						
							
							
								
								Expand codec check so that raw formats must be equal for a Packet2Packet bridge to occur  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43340  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								465adf2bf1 
								
							
								 
							
						 
						
							
							
								
								allow for packetization on rtp channel drivers, need to add  
							
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							option for setting our own packetization as apposed to just doing 
what is asked.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43243  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								fcb999c01c 
								
							
								 
							
						 
						
							
							
								
								merge qwell's CLI verbification work  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43212  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								d2a359e57f 
								
							
								 
							
						 
						
							
							
								
								Optimize a bit  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42583  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								10e361763c 
								
							
								 
							
						 
						
							
							
								
								It's another round of RTP updates!  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42569  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								f912c9e69f 
								
							
								 
							
						 
						
							
							
								
								Unbridge the RTP streams at the correct place  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41735  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0be2884d80 
								
							
								 
							
						 
						
							
							
								
								If we are doing video and we can't reinvite, then resort to generic bridging instead of Packet2Packet since video isn't supported there yet. (reported by PCadach in #asterisk-bugs)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41718  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								f2b836ff4f 
								
							
								 
							
						 
						
							
							
								
								Tweak the DTMF muting stuff a bit to take into account VLDTMF and compensation.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41632  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								29ee02bfce 
								
							
								 
							
						 
						
							
							
								
								Only write a received packet out if we are actually bridged to something  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41574  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								c6977b9983 
								
							
								 
							
						 
						
							
							
								
								Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0855df6a5a 
								
							
								 
							
						 
						
							
							
								
								Only feed a DTMF frame into the core if we need to  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41338  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								245aa1a62d 
								
							
								 
							
						 
						
							
							
								
								Only switch the second alert fd (which is RTCP) to callback mode if it is in use  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41317  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								12b6ec4e11 
								
							
								 
							
						 
						
							
							
								
								Use an API call (ast_rtp_get_bridged) to return the RTP stream we are bridged to, and also use it in chan_sip so we know to ignore the no RTP activity checking  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41316  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								da7d969ae1 
								
							
								 
							
						 
						
							
							
								
								If the RTP stack is already being operated in callback mode, then suspend it upon switching to P2P callback bridging. Once P2P callback bridging has ended, then restore callback mode.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41303  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								ca33d2ecc6 
								
							
								 
							
						 
						
							
							
								
								This is the last round of RTP bridge optimizations. Basically it introduces a way that under a straight bridge (ie: no transcoding and no DTMF detection) the core is not touched at all and no frames pass through (not even null frames). This is accomplished by stealing the file descriptors from the channel and using the provided IO context with a custom callback. When a channel is placed on hold this bridge is broken so audio can flow from the core to the other side. When a channel is off hold this bridge is re-established.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41302  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								a2e2a51d8d 
								
							
								 
							
						 
						
							
							
								
								Move the direct bridge write to after the NAT handling code  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41285  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								c70ed7614a 
								
							
								 
							
						 
						
							
							
								
								Merge in RTP-level packet bridging. Packet comes in, packet goes out - that's what RTP-level packet bridging is all about!  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41235  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0a27d8bfe5 
								
							
								 
							
						 
						
							
							
								
								merge new_loader_completion branch, including (at least):  
							
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							- restructured build tree and makefiles to eliminate recursion problems
  - support for embedded modules
  - support for static builds
  - simpler cross-compilation support
  - simpler module/loader interface (no exported symbols)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago