Russell Bryant
4ee818eb8f
Merged revisions 48322 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines
Fix the name of the rtignoreregexpire option in the sample configuration file.
(issue #8526 , arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
7945d4ca35
Add missing s from another repository. (thanks jcmoore!)
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20 years ago
Olle Johansson
027096b3a3
Updating sip.conf.sample with information about T38 not working
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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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20 years ago
Jason Parker
56c03478ab
Add documentation to voicemail.conf.sample for ODBC storage.
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Issue 8499 - patch by blitzrage.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 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
20 years ago
Jason Parker
8cbe6025b6
Merged revisions 48183 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines
Fix a small typo - issue 8848, reported by pabelanger
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20 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... :-)
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20 years ago
Joshua Colp
802c3c3ecf
Merged revisions 48142 via svnmerge from
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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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20 years ago
Olle Johansson
a68edf400f
Explain the use device status system implemented in SIP for subscriptions,
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
3fe8e34039
Clarify RTP timers. Sorry, grandma.
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20 years ago
Olle Johansson
7da1a54fe6
Explain properly how videosupport works.
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Committ from Asterisk Video Task Force meeting in Paris!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 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
20 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
20 years ago
Kevin P. Fleming
4fd3b973bf
clean up sample config, and make native file playback the more obvious default choice
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
9ab1cc22a4
Support ;rport when we're supposed to support ;rport. Issue #7473 .
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20 years ago
Christian Richter
6964f148ba
Merged revisions 46176 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line
added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session
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20 years ago
Russell Bryant
8273d95be3
update entry to reboot a snom phone (issue #7850 , pnlarsson)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
590698e583
Adding information about Marks direct-RTP hack to the docs...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
45fc0eaba4
Now, remove all traces of the option that we did not need :-)
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20 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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20 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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20 years ago
Christian Richter
13825dab85
Merged revisions 44334 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) | 1 line
added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible
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20 years ago
Steve Murphy
743097a6c1
Hang on a minute, the install process sticks muted.conf in /etc/asterisk, so that's where muted should look for it, right\?
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Steve Murphy
caa0d129f2
I've been meaning to add some explanation about muted... here it is
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Steve Murphy
7778d017fc
CLI reverbification update to this config file
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Matt O'Gorman
5058b9e13f
updated res_jabber for even better component support, soon will be jep-0100 compliant.
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also removed chan_jingle and infromed info from jingle.txt, chan_gtalk still works and should be used in this version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Paul Cadach
6b37705130
Missed part of userconf functionality for chan_h323
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Russell Bryant
3e2fa16670
Merged revisions 44110 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01 Oct 2006) | 3 lines
Fix the name of the "eventmemberstatus" option in the sample queues.conf
(issue #8065 , adamg)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Steve Murphy
740233659f
This change to extensions.ael was to fix bug 8031; the install scripts are causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43739 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Tilghman Lesher
5ac0ca6af1
Twould help if we actually documented how the new features in res_odbc actually work. (Oops)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Jason Parker
8bd82ebc0d
Add documentation on rtp packetization.
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Jason Parker
3c224654c2
Document member name logging functionality.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Matthew Fredrickson
2e5118cc49
Add the h323 config file. Arrr!!! for international talk like a pirate's day.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Matt O'Gorman
ec4bf7a849
seperate jingle and gtalk so it will be easier to track
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changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Steve Murphy
4ed1578104
Clarified the meaning of the callwaiting variable in the zapata.conf file.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Joshua Colp
956b837a41
Merged revisions 43159 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43159 | file | 2006-09-18 11:05:39 -0400 (Mon, 18 Sep 2006) | 2 lines
Add number unobtainable tone for New Zealand (issue #7969 reported by nic_bellamy)
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20 years ago
Tilghman Lesher
ee27f9efee
Remove the suggestion of realtime hints, since that functionality will not be available until post-1.4
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20 years ago
Mark Spencer
c2d959c2f9
Improve documentation of users.conf items.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Jason Parker
6d5809297b
Skinny hold support.
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Original patch by wedhorn, with modifications by me.
Issue #7588
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20 years ago
Kevin P. Fleming
c2c4f86c72
merge markster's usersconf branch with some slight changes
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20 years ago
Tilghman Lesher
091e1aed8d
Merged revisions 42716 via svnmerge from
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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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20 years ago
Tilghman Lesher
b0666488f3
Merged revisions 42697 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42697 | tilghman | 2006-09-11 09:40:13 -0500 (Mon, 11 Sep 2006) | 2 lines
Two grammar issues (bug 7927)
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20 years ago
Voicetronix Support
e02897acd4
Board numbers and channel numbers are now 0 based, since vpb driver
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version 3.0 (released December 2005)
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20 years ago
Matthew Fredrickson
75822388a4
Make sure we give a little warning about the echotraining option
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20 years ago
Joshua Colp
34eb4f54ba
Use lower case 'x' instead of a UTF-8 character (issue #7888 reported by flefoll)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Mark Spencer
47d8e14871
Comment out default from extensions.ael
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20 years ago
Jason Parker
4ba458e352
Merged revisions 42014 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42014 | qwell | 2006-09-05 09:27:46 -0700 (Tue, 05 Sep 2006) | 4 lines
Small typo in zapata.conf.sample
Reported by ppyy in 7881
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20 years ago
Olle Johansson
3f888b84f3
Use GLOBAL() in dialplan examples
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20 years ago
Joshua Colp
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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20 years ago
Kevin P. Fleming
b281acf0f8
change default setting for autofallthrough
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago