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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48142 via svnmerge from
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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for realtime users.
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- Add docs on canreinvite
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to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
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...where did the 1.2 properties come from, really? they're back.
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is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
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replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45280 | file | 2006-10-16 16:06:18 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45265 via svnmerge from
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/1.4
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r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45260 via svnmerge from
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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remember.
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They are highly convenient for writing configuration files, especially
if you have many similar entries, or want to switch quickly between
different configurations without having to comment/uncomment large
sections of the files.
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1. slightly rearrange/simplify the parsing of the argument in sip_register.
This brings in a patch that has been in Mantis (5834) for ages,
and is the larger part of the commit;
2. implement the "contact" option for peers, similar to the one in users.conf:
If you put a "contact" option with a non-empty argument (e.g. contact=123)
in a peer section, asterisk will register with the provider as if you had a
register= username:secret@host/contact
line in the general section.
The latter is a very small is a new feature so i am not putting it
in the 1.4 branch, although the "contact" option in user.conf is
already in the 1.4 branch and so it wouldn't be too strange to
merge it.
Note that the implementation of "contact" is much simpler than
the one in 5834, and limited to a few lines in build_peer().
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(does not apply to 1.4)
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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modifications)
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- Enable videosupport per device
- Implement maxcallbitrate setting for video calls
Patch by John Martin, thanks!
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- Implement option for allow/disallow subscriptions
- Implement option for allow/disallow overlap dialling
- Set default to disable overlap dialling in sip.conf.sample for new installations
- Remove overlap dialling from subscription logic
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