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(closes issue #11304)
Reported by: pj
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Issue 10753
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Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.
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externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
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but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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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.
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when chan_local or chan_agent is involved in the call.
I don't know how big a fix that would be to solve, but this is
the current state of affairs.
(Chan_sip currently checks if the other side of the bridge
has a SIP tech. We could/should implement another check,
possibly for udptl_write or some flag in the ast_channel
structure).
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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.
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with queues and SIP device states
- 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... :-)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines
Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)
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queues and manager a bit better.
Like in 1.2, you will get more detailed information if you set a call
limit for a device. When the call limit is reached, the status system will
report a device as busy.
For queues, setting a call limit per SIP device is propably a requirement.
In most cases, it will work much better if you only use type=peer and not
type=friend. We might decide to backport the new setting from trunk to
apply all call limits to the peer part of a friend only.
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Committ from Asterisk Video Task Force meeting in Paris!
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database lookups
for all those realtime users out there.
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- Update docs on canreinvite. "nonat" is the recommended setting for most users with
phones behind a NAT.
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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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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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Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon.
Issue #7989, patch by DEA, slightly modified.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines
Spelling/grammar fixes (Issue 7929)
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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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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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be seen in the code. Did it exist, was it planned to exist
or was it documentationware only? Ask Dr Asterisk.
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- Fix small issue with SIP history
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add "promises" to the sip.conf.sample...
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r36253 | kpfleming | 2006-06-29 02:19:27 -0500 (Thu, 29 Jun 2006) | 2 lines
add documentation for peer-specific 'outboundproxy' setting
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r36254 | kpfleming | 2006-06-29 02:19:54 -0500 (Thu, 29 Jun 2006) | 2 lines
clarify documentation for 'persistentmembers' setting
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and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257, north, with additional updates and modifications)
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r31321 | kpfleming | 2006-06-01 07:41:47 -0500 (Thu, 01 Jun 2006) | 2 lines
remove a sample entry that never should have been added (code to support it was not merged)
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
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use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response
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media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
also, documented the 'canreinvite=update' option in the sample config file
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contexts when peer registers (issue #6869 reported by and created by Marquis)
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- Optionally send systemname in manager (cool when you have a manager proxy)
- Use systemname in CLI prompt
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- add information about subscriptions and realtime dial plans in sip.conf.sample
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- Improve documentation of pedantic
(related to issue #7016)
From the air above Russia...
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(issue #6390)
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