Olle Johansson
7945d4ca35
Add missing s from another repository. (thanks jcmoore!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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
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
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48186 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
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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
19 years ago
Olle Johansson
3fe8e34039
Clarify RTP timers. Sorry, grandma.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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
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
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
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
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
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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
19 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
19 years ago
Olle Johansson
45fc0eaba4
Now, remove all traces of the option that we did not need :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45314 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
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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
19 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
19 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
19 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
19 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
19 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
19 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
19 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
19 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
19 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
19 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
19 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
19 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
19 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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
19 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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19 years ago
Kevin P. Fleming
c2c4f86c72
merge markster's usersconf branch with some slight changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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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19 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Matthew Fredrickson
75822388a4
Make sure we give a little warning about the echotraining option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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
19 years ago
Mark Spencer
47d8e14871
Comment out default from extensions.ael
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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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19 years ago
Olle Johansson
3f888b84f3
Use GLOBAL() in dialplan examples
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 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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19 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
19 years ago
Jason Parker
0850bb72ed
Kevins last commit made me spot a typo.
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19 years ago
Kevin P. Fleming
ece7018515
add one remaining bit of functionality to the features.conf applicationmap (from Matt Nicholson in Digium Express Services)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
7b65cdc0c7
remove documentation of 'global' section in modules.conf, since it is no longer needed or supported
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19 years ago
Joshua Colp
236c23269d
Merged revisions 40979 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40979 | file | 2006-08-24 13:13:04 -0400 (Thu, 24 Aug 2006) | 2 lines
I can type english. Honest. Thanks Kenny.
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19 years ago
Joshua Colp
e5c0665caf
Merged revisions 40971 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40971 | file | 2006-08-24 12:44:41 -0400 (Thu, 24 Aug 2006) | 2 lines
Minor documentation fix to add the 'dynamic' dialplan option from angler
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19 years ago
Kevin P. Fleming
749cd217c3
Merged revisions 40392 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40392 | kpfleming | 2006-08-18 14:10:51 -0500 (Fri, 18 Aug 2006) | 2 lines
make a feeble attempt to avoid the 'how do I enable my hardware echo canceler' questions
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19 years ago
Russell Bryant
87ac16847e
- unregister SLA apps on module unload and add sample config (issue #7701 , junky)
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- rename SLAS to SLAStation, and SLAT to SLATrunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
a0bd41f79b
Add support for Sigma Designs cards. These basically allow you to offload dialtone generation to the board. If you're using a quicknet board where this might work, give it a try as well. (issue #6092 reported by ywalther - minor mods by moi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Joshua Colp
0f0323dbab
Clarify volgain option a bit, it needs sox to work.
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20 years ago
Tilghman Lesher
f37a4e3e12
Bug 6237 - add volgain parameter, such that voicemail messages may be amplified after recording
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Kevin P. Fleming
9d26f46fc7
remove some extraneous 'followme' in prompt names
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20 years ago
Olle Johansson
2f69bec40e
Add placeholder for sla.conf sample in configs/. Please update with
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correct information not to overload the switchboard with calls to sales.
Thank you! ;-)
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20 years ago
Russell Bryant
4d7c67fc72
Merge my applicationmap_fixup branch to address the issues described in this
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post to the asterisk-dev mailing list:
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022174.html
This implements full control over both which channel(s) can activate a dynamic
feature, as well as which channel to run the application on. I also updated
the documentation on the applicationmap in features.conf.sample in hopes that
the configuration is more clear.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Tilghman Lesher
0d902b3033
Update documentation on realtime; add a workaround for lack of realtime hints by using func_odbc
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Kevin P. Fleming
6d0742fc16
merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Mark Spencer
837910062b
First pass at in-place file manipulation via manager
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
North Antara
e69e056012
config sample for the previous, regarding ADSI
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Kevin P. Fleming
4376af0080
actually make the non-standard G726-32 behavior available for SIP clients
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Christian Richter
54ce0f0a22
added even more statefulness for sending out disconnect/release/release_complete messages. added support for incoming presentation/screening. fixed a bug that we generate TONE_EVENTS on hanguptone_indicatem, which caused asterisk to write blocking thread messages. added nodialtone option to prevent dialtone for always_immediate
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
3b5a2aafa4
- Change filename to current file name
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Joshua Colp
ba092c1244
And now the trunk version! Add an option for IAX2 users that allows you to set how many outstanding AUTHREQs chan_iax2 will wait for replies on.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
0e0059c0f3
Remove configuration option "restrictcid" that is nowhere to
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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
20 years ago
Matthew Fredrickson
de03118578
Asterisk portion of the T309 patch. ( #7271 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Christian Richter
bd0b801a0d
* removed tone_indicate, we genrate only the dialtone by ourself (and the hanguptone of course)
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* removed the state handling from release_chan, and simplified the ast_hangup/ast_queue_hangup stuff
* added pp_l2_check option, for pp lines where the pbx does not initially gets the L2 up
* simplified and fixed a bug in the pid generation code
* fixed a bug in empty_chan, which might cause segfaults and memorry corruptions
* added prepare_bc function, which is sort of the opposite of empty_bc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
ec9d4711d7
- Add notes about voicemail depending on res_adsi
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- Remove obsolete modules from modules.conf.sample
(make install will warn if those exist on the machine)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
b971f65978
- Make use of system name in realtime SIP peers optional
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- Fix small issue with SIP history
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
f3594bd1a0
Removing configuration options that does not do anything yet. No need to
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add "promises" to the sip.conf.sample...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Christian Richter
f5c0cd2ddc
added better L2 handling for ptp, if it's down we don't try to call on that port in groupdial anymore, also we try to get it up then. Additionally added the configoptions ntdebugflags and ntdebugfile to debug the mISDNuser NT Stack (should have done that ages before..). isdn_lib.c compiles again.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Kevin P. Fleming
dec3d7d4c0
Merged revisions 36253-36254 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
4177596e8d
reformatting sip.conf.sample a bit, adding dumphistory that was not documented
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
cc43f0bdc7
Speling error. Avoid swenglish :-) (thanks, jtodd!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
6399ac438d
Add explanation and warning about the "s" extension. (Hi Mike :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
f8311adcda
METERMAIDS:
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-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
e2b0c5b558
Add example of permit/deny to sip.conf.sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Tilghman Lesher
89f6ffe1e5
Bug 6589 - option to display channel variables in queue events
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Joshua Colp
4c066de7cf
Merged revisions 35334 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r35334 | file | 2006-06-21 16:25:34 -0300 (Wed, 21 Jun 2006) | 2 lines
Add Venezuelan indications (issue #7402 reported by palillo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
North Antara
a5d6979fac
Finally merge chan_skinny fixes into trunk.
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ready...set...GO!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Russell Bryant
035a8b4278
Merged revisions 34627 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r34627 | russell | 2006-06-18 16:15:15 -0400 (Sun, 18 Jun 2006) | 5 lines
don't store multiple secrets delimited with semicolons for peers because this
is only valid for users. Instead, only keep the last specified secret for a
peer entry. Also, document how multiple secrets are handled in the sample
config. (Reported by PCadach on #asterisk-bugs)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Joshua Colp
0e5e744fb2
Add bulgarian indications (issue #7314 reported by KNK)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Joshua Colp
5456f425c6
Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough support
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Matt O'Gorman
1e530787f3
solves some bugs with memory allocation, and adds
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a message stack.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Kevin P. Fleming
bc49d5bfb3
moh files will now be distributed in native format, not mp3, so...
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update sounds/Makefile to download/unpack the proper files
remove all evidence of mpg123 from the main Makefile
change the sample musiconhold.conf file to use native mode instead of mp3
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
BJ Weschke
3d973a0686
Introducing app_followme into /trunk!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Kevin P. Fleming
dd6de5ee4e
it's time... only enable global priority jumping if the config file says to do so
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
4e74b8fb75
Issue #7231 - Missing indications from libtonezone (tzafrir)
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- "tuttter" changed to "stutter" ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Olle Johansson
2dc6947144
Issue #2863 - Improved RTCP support (John Martin, Fredrik Olsson)
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Thanks to everyone involved in working with this!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Matt O'Gorman
18b135f215
oops my config file was out of date not the sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Matt O'Gorman
58cd5f2440
oops fixing example config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Russell Bryant
4c76028de9
- add the ability to configure forced jitterbuffers on h323, jingle,
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Christian Richter
4be235a974
added bearer capability reject support. we send release instead of disconnect in case we have no real channel yet. added support for Restarting channels added support for sending complete decoding. changed some log levels.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Kevin P. Fleming
6bce269454
Merged revisions 31321 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago