Olle Johansson
da7a35a1cc
Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27 16:56:11 +00:00
Russell Bryant
850dd4ea61
Use spaces as a separator for the redirect option to improve readability
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-24 21:01:02 +00:00
Russell Bryant
2c5071a006
- Convert the list of URI handlers to use the linked list macros. While doing
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this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
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2006-12-23 20:13:14 +00:00
Steve Murphy
9327720c37
As per bug 7978, this version introduces the jittertargetextra option in config files
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2006-12-21 00:24:08 +00:00
Luigi Rizzo
437f4288cd
- Generalize the function ssl_setup() so that the certificate info
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are passed as an argument.
- Update the code in main/http.c to use the new interface
(the diff is large but mostly mechanical, due to the name change of
several variables);
- And since now it is trivial, implement "AMI over TLS", and document
the possible options in manager.conf
- And since the test client (openssl s_client -connect host:port )
does not generate \r\n as a line terminator, make get_input()
also accept just a \n as a line terminator (Mac users: do you
also need the \r-only version ?)
The option parsing in manager.conf is not very efficient, and needs
to be cleaned up and made similar to what we have in http.conf
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2006-12-07 16:42:29 +00:00
Russell Bryant
c7efdf6759
Merged revisions 48323 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48323 | russell | 2006-12-06 11:15:45 -0500 (Wed, 06 Dec 2006) | 11 lines
Merged revisions 48322 via svnmerge from
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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2006-12-06 16:19:01 +00:00
Olle Johansson
d1b621c6a5
Adding docs on t.38
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2006-12-05 16:48:15 +00:00
Jason Parker
3e8669595e
Merged revisions 48230 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48230 | qwell | 2006-12-04 11:54:46 -0600 (Mon, 04 Dec 2006) | 4 lines
Add documentation to voicemail.conf.sample for ODBC storage.
Issue 8499 - patch by blitzrage.
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2006-12-04 17:55:38 +00:00
Olle Johansson
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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2006-12-02 12:05:40 +00:00
Jason Parker
97614cb6b4
Merged revisions 48186 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48183 via svnmerge from
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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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2006-12-01 20:26:44 +00:00
Olle Johansson
4ce5b7c080
- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4)
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- Small fixes to limitonpeer
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2006-12-01 18:16:16 +00:00
Joshua Colp
c946e3b3fb
Merged revisions 48143 via svnmerge from
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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
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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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2006-11-30 17:58:53 +00:00
Olle Johansson
7e46275b51
Clarify some settings for status reports in subscriptions, queues and manager
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2006-11-29 20:57:48 +00:00
Olle Johansson
e5145bebe4
Explain RTP timeouts
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2006-11-29 19:47:45 +00:00
Luigi Rizzo
2e7fd7cbdb
add a new http.conf option, sslbindaddr.
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Because https is more secure than http, it usually
makes sense to keep this service more open than the
one on the unencrypted port.
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2006-11-27 20:21:40 +00:00
Olle Johansson
4e47ce525b
Update docs for videosupport
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2006-11-20 11:46:45 +00:00
Jason Parker
54d44e9b00
Add ability to notify an external application/script that the voicemail password was,
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while also still changing the password "internally".
Issue 7371, initial patch by pdunkel, with rewrite/config comments by me.
Additional modifications (yay bitmask) by pdunkel.
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2006-11-17 21:51:42 +00:00
Jason Parker
bfd630682e
Add ability to modify range for dring matching.
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Issue #8369 , patch by ssuehring, modified slightly by me.
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2006-11-16 22:32:23 +00:00
Olle Johansson
a6f5adefa1
Make it possible to enable/disable onhold tracking, in order to make life easier
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for realtime users.
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2006-11-16 19:29:28 +00:00
Olle Johansson
a427a2a89a
- CANCEL never uses authentication
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- Add docs on canreinvite
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2006-11-16 15:12:30 +00:00
Joshua Colp
64d5316a53
Add 'loose' option to joinempty and leavewhenempty which is almost exactly like 'strict' except it does not count paused queue members as unavailable. (issue #8263 reported by gnarf)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-13 18:23:55 +00:00
Tilghman Lesher
f2bc05d1d4
Feature: allow the sanity SQL to be customized per connection class (Issue 6453)
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2006-11-13 05:58:14 +00:00
Kevin P. Fleming
8b8d8d0b24
Merged revisions 47279 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006) | 2 lines
clean up sample config, and make native file playback the more obvious default choice
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2006-11-07 18:56:56 +00:00
Russell Bryant
729620dd3f
List ss7 with the rest of the valid signalling types. Group SS7 options
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together and comment them out by default.
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2006-11-04 22:41:12 +00:00
Olle Johansson
d900b47ccf
Adding new config option "limitpeersonly" to only apply call limits
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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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2006-11-04 19:13:30 +00:00
Luigi Rizzo
0b8669b87e
document the "debug" parameter, and the change
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manager list -> manager show
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2006-11-04 01:20:38 +00:00
Olle Johansson
b136baaff4
Fix rport handling.
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...where did the 1.2 properties come from, really? they're back.
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2006-10-31 10:29:24 +00:00
Olle Johansson
f98f457727
Change name of "contact" setting to "callback" which better reflects what it
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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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2006-10-30 19:56:14 +00:00
BJ Weschke
95a4fc7af2
* Added option to run macro when a queue member is connected to a caller,
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see queues.conf.sample for details.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
(#8216 , jmls reported and submitted)
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2006-10-27 18:59:16 +00:00
Russell Bryant
4a523b1b2d
Add the ability to customize some of the prompts used within the voicemail
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application by configuring them in voicemail.conf (issue #7415 , patch by
fkasumovic, with some fixes and documentation updates by myself)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-27 16:47:44 +00:00
Christian Richter
f19300635f
Merged revisions 46351-46353 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46176 via svnmerge from
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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r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line
fixed not compile issue, which was just introduced
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r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
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2006-10-27 11:18:32 +00:00
Luigi Rizzo
e85d8e98d1
document the match_auth_username option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-26 07:32:00 +00:00
Matthew Fredrickson
67926b9ac4
Update changes to do US style point code parsing/formatting (xxx.xxx.xxx)
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2006-10-25 19:14:23 +00:00
Olle Johansson
c30f1d12c5
Ok, second attempt...
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2006-10-25 14:26:22 +00:00
Olle Johansson
25b8f577b8
On the other hand, don't use 1.4 patches for trunk... Sorry.
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2006-10-25 14:14:53 +00:00
Olle Johansson
13ea5fc0d0
Add ability to adapt the IAX trunk packets to the MTU size, to avoid bad audio
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when the number of channels fill the MTU on a given link.
In the future, this needs to be configurable per peer with trunking enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-25 14:06:13 +00:00
Luigi Rizzo
c15f7953c8
Fix a few issues in the previous (disabled) HTTPS code,
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and support linux as well (using fopencookie(), which should
be available in glibc).
Update configure.ac to check for funopen (BSD) and fopencookie(glibc),
and while we are at it also for gethostbyname_r
(the generated files need to be updated, or you need
to run bootstrap.sh yourself).
Document the new options in http.conf.sample
(names are only tentative, better ones are welcome).
At this point we can safely enable the option.
Anyone willing to try this on Sun and Apple platforms ?
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2006-10-22 19:09:25 +00:00
Luigi Rizzo
d171c3d864
remove unused fields and unimplemented options.
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2006-10-18 11:59:08 +00:00
Russell Bryant
d8e688ece9
Merged revisions 45439 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45439 | russell | 2006-10-17 22:19:07 -0400 (Tue, 17 Oct 2006) | 2 lines
update entry to reboot a snom phone (issue #7850 , pnlarsson)
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2006-10-18 02:19:55 +00:00
Olle Johansson
a8a26ad389
Update of docs
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2006-10-17 17:51:34 +00:00
Joshua Colp
c62784c10d
In the course of a data this has been turned into an option to ignore 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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2006-10-16 20:26:56 +00:00
Joshua Colp
da330feb60
Merged revisions 45280 via svnmerge from
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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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2006-10-16 20:08:23 +00:00
Joshua Colp
b58cc9e1bd
Merged revisions 45262 via svnmerge from
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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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2006-10-16 19:43:33 +00:00
Christian Richter
e09ad744af
Merged revisions 44561 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44561 | crichter | 2006-10-06 14:50:25 +0200 (Fr, 06 Okt 2006) | 9 lines
Merged revisions 44334 via svnmerge from
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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2006-10-11 08:34:03 +00:00
Olle Johansson
77c69dc4ef
Recommend using "sip reload" since it's much easier to learn and
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remember.
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2006-10-07 16:26:11 +00:00
Luigi Rizzo
b19b4b9764
document a bit the use of templates.
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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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2006-10-06 16:43:36 +00:00
Luigi Rizzo
f94849ca2a
document the "contact" option a bit better.
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2006-10-06 16:20:42 +00:00
Luigi Rizzo
ccca5843fd
Two things:
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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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2006-10-06 15:41:12 +00:00
Luigi Rizzo
2a7ac3f735
update example commands to match current syntax
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(does not apply to 1.4)
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2006-10-06 06:43:49 +00:00
Steve Murphy
8135d5016a
I've been meaning to add some explanation about muted... here it is
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2006-10-04 17:10:53 +00:00
Steve Murphy
6e42aa676c
CLI reverbification update to this config file
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-04 16:48:07 +00:00
Joshua Colp
8ff3dd273a
Expand setinterfacevar option to also set a variable, MEMBERNAME, which contains the member's name. (issue #8046 reported by jmls)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-02 20:58:48 +00:00
Paul Cadach
b4ef9599de
Merged revisions 44186 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44186 | pcadach | 2006-10-03 00:52:56 +0600 (Втр, 03 Окт 2006) | 1 line
Missed part of userconf functionality for chan_h323
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-02 19:01:10 +00:00
Joshua Colp
e5203bb283
Add option to logger to rename log files with timestamp (issue #8020 reported by jmls)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-02 17:54:21 +00:00
Joshua Colp
cc1945ce1b
Add option 'keepstats' which will keep queue statistics during a reload. (issue #7908 reported by jmls)
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2006-10-02 15:40:38 +00:00
Russell Bryant
8a5cf10121
Merged revisions 44111 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44111 | russell | 2006-10-01 11:20:12 -0400 (Sun, 01 Oct 2006) | 11 lines
Merged revisions 44110 via svnmerge from
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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2006-10-01 15:20:50 +00:00
Paul Cadach
9cf1f14ed5
Handle HOLD/RETRIEVE notifications
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-28 10:41:38 +00:00
Steve Murphy
2b7debf368
Merged revisions 43739 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43739 | murf | 2006-09-26 20:32:47 -0600 (Tue, 26 Sep 2006) | 1 line
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/trunk@43740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-27 02:46:50 +00:00
Jason Parker
fe1a3b8877
Add optional queue_log_name config option for logger.conf, to change the
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name of the queue_log file.
Issue #7363 , patch by Steve Davies, slightly modified by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-26 17:09:01 +00:00
Paul Cadach
c479c25182
Support for negotiation and receiption of Cisco's RTP DTMF
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-25 09:03:14 +00:00
Paul Cadach
1af96a0b21
Specify RFC2833 payload on dtmfmode option rather than dtmfcodec option (deprecated)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-24 18:59:38 +00:00
Tilghman Lesher
859bc68383
Merged revisions 43464 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43464 | tilghman | 2006-09-21 18:24:41 -0500 (Thu, 21 Sep 2006) | 2 lines
Twould help if we actually documented how the new features in res_odbc actually work. (Oops)
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2006-09-21 23:29:32 +00:00
Matthew Fredrickson
bd76cda68d
Merge in SS7 changes.... need to still cleanup zapata.conf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-21 18:41:33 +00:00
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
2006-09-20 17:39:59 +00:00
Jason Parker
3c224654c2
Document member name logging functionality.
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2006-09-20 14:47:59 +00:00
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
2006-09-19 19:25:18 +00:00
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
2006-09-18 16:36:14 +00:00
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
2006-09-18 15:44:18 +00:00
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
2006-09-18 15:07:59 +00:00
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
2006-09-18 14:53:54 +00:00
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
2006-09-18 05:40:17 +00:00
Jason Parker
6d5809297b
Skinny hold support.
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Original patch by wedhorn, with modifications by me.
Issue #7588
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2006-09-17 22:24:27 +00:00
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
2006-09-16 23:53:58 +00:00
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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2006-09-11 16:41:49 +00:00
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
2006-09-11 14:41:35 +00:00
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
2006-09-11 03:29:08 +00:00
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
2006-09-07 20:05:37 +00:00
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
2006-09-07 17:09:55 +00:00
Mark Spencer
47d8e14871
Comment out default from extensions.ael
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2006-09-07 15:35:52 +00:00
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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2006-09-05 16:38:22 +00:00
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
2006-08-31 10:40:24 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31 01:59:02 +00:00
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
2006-08-30 21:44:05 +00:00
Jason Parker
0850bb72ed
Kevins last commit made me spot a typo.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-29 21:52:40 +00:00
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
2006-08-29 21:20:43 +00:00
Kevin P. Fleming
7b65cdc0c7
remove documentation of 'global' section in modules.conf, since it is no longer needed or supported
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-25 14:14:27 +00:00
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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2006-08-24 17:13:57 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-24 16:46:13 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-18 19:11:34 +00:00
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
2006-08-14 01:32:46 +00:00
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
2006-08-08 17:02:45 +00:00
Joshua Colp
0f0323dbab
Clarify volgain option a bit, it needs sox to work.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 15:53:51 +00:00
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
2006-08-08 15:37:53 +00:00
Kevin P. Fleming
9d26f46fc7
remove some extraneous 'followme' in prompt names
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-07 18:47:33 +00:00
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! ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-07 17:01:40 +00:00
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
2006-08-07 04:15:52 +00:00
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
2006-08-03 19:05:06 +00:00
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
2006-07-19 20:44:39 +00:00
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
2006-07-19 13:28:38 +00:00
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
2006-07-14 21:49:40 +00:00
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
2006-07-13 20:35:41 +00:00
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
2006-07-13 14:13:24 +00:00
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
2006-07-12 20:56:49 +00:00
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
2006-07-11 19:03:56 +00:00
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
2006-07-10 11:20:49 +00:00
Matthew Fredrickson
de03118578
Asterisk portion of the T309 patch. ( #7271 )
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2006-07-06 21:51:51 +00:00
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
2006-07-06 15:11:40 +00:00
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
2006-07-05 06:06:23 +00:00
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
2006-07-02 12:00:36 +00:00
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
2006-06-30 07:18:30 +00:00
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
2006-06-29 20:12:19 +00:00
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
2006-06-29 08:01:08 +00:00
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
2006-06-29 07:04:43 +00:00
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
2006-06-26 18:34:29 +00:00
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
2006-06-26 17:31:51 +00:00
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
2006-06-26 16:43:21 +00:00
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
2006-06-26 16:24:43 +00:00
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
2006-06-22 15:43:02 +00:00
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
2006-06-21 19:26:52 +00:00
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
2006-06-18 22:36:39 +00:00
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
2006-06-18 20:18:41 +00:00
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
2006-06-16 16:23:13 +00:00
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
2006-06-13 19:38:41 +00:00
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
2006-06-07 22:43:20 +00:00
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
2006-06-07 22:37:31 +00:00
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
2006-06-06 07:29:57 +00:00
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
2006-06-05 14:00:13 +00:00
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
2006-06-05 09:27:38 +00:00
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
2006-06-05 08:39:42 +00:00
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
2006-06-04 17:08:18 +00:00
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
2006-06-04 17:07:09 +00:00
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
2006-06-01 16:47:28 +00:00
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
2006-06-01 12:51:41 +00:00
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
2006-06-01 12:43:01 +00:00
Matt O'Gorman
8cfb992c1e
adds statusmessage customization from Julian Lyndon-Smith
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and fixes bug with pruneregister
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 07:49:47 +00:00
North Antara
e25c4621b4
Nobody saw this coming, I bet.
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jitterbuffer!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 04:35:00 +00:00
Russell Bryant
bb7dd96cfe
Add support for using a jitterbuffer for RTP on bridged calls. This includes
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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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 16:56:50 +00:00
Kevin P. Fleming
18606233da
fix various typos and other bits (from Ian Kinner)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-30 16:01:50 +00:00
Joshua Colp
18248f092f
Merged revisions 30239 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r30239 | file | 2006-05-25 12:27:44 -0300 (Thu, 25 May 2006) | 2 lines
Get rid of an incorrect SIP dial string in the sample extensions.conf - I even tried variations... no go (issue #7222 reported by arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-25 15:29:30 +00:00
Christian Richter
8122c35675
fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 07:58:52 +00:00
Kevin P. Fleming
3e99be68d1
add a new option for 'obscuring' SIP user/peer names from fishers
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 03:28:49 +00:00
Matt O'Gorman
45107ed763
allows for configurable answer timeout on attended transfer
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patch 0006763 with minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23 18:23:05 +00:00
Matt O'Gorman
7aa1a77e75
asterisk-xmpp merge in
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 21:12:30 +00:00
BJ Weschke
5235890be4
This is part 2/2 of the patches for #7090 . Adds one-step call parking to /trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340 .
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 16:43:43 +00:00
Christian Richter
19d46333bf
added callcounters for incoming and outgoing calls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 15:02:03 +00:00
Tilghman Lesher
9e81cc3e0c
Escaping commas within fields isn't always desireable.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 03:40:31 +00:00
Russell Bryant
1fcc86d905
Add support for logging CDR recrods to a radius server (issue #6639 , phsultan)
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- with contributions from miconda, jcollie, and sb
- branch maintained by oej
Thanks everyone!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-20 22:30:05 +00:00
Kevin P. Fleming
42cf0b0a8f
add another media path reinvite 'flavor', where we will only redirect our media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
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also, documented the 'canreinvite=update' option in the sample config file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18 16:57:59 +00:00
Joshua Colp
6d603ec09c
Allow contexts in regexten so that extensions can be added to multiple contexts when peer registers (issue #6869 reported by and created by Marquis)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18 14:07:46 +00:00
Joshua Colp
15358932ec
Add distinctive ring detection with Caller ID for Australia, New Zealand, and other countries. (issue #3596 reported by deon patch by dbowerman with minor mods by moi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 16:51:47 +00:00