Commit Graph

2149 Commits

Author SHA1 Message Date
Olle Johansson
4c1068c136 oops. Thanks Terry.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 19:53:40 +00:00
Olle Johansson
ee3a0af16a Merged revisions 66503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 lines

Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 19:35:43 +00:00
Olle Johansson
6d6c525b10 Merged revisions 66474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines

Don't issue hangup on hangup on hangup on hangup (for jcmoore)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 19:17:49 +00:00
Olle Johansson
0b67a7d80a Merged revisions 66414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines

Don't reset hangupcause if we already have one

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 16:19:53 +00:00
Olle Johansson
f4e81d7a54 Merged revisions 66404 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines

Tracking down hanging channels, killing them one by one. Issue #9235 and related

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 16:17:17 +00:00
Olle Johansson
36f15091bb Merged revisions 66363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines

Merged revisions 66349 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines

Issue #9802 - Change inuse counter on CANCEL

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 10:02:31 +00:00
Joshua Colp
4d03b4f268 Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-28 23:28:52 +00:00
Joshua Colp
39e9b3112c Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-28 23:24:04 +00:00
Russell Bryant
4b3a3fb14c Add a new API call for creating detached threads. Then, go replace all of the
places in the code where the same block of code for creating detached threads
was replicated.  (patch from bbryant)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 18:30:19 +00:00
Joshua Colp
202fbe363a Merged revisions 65839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines

Merged revisions 65837 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines

Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:43:49 +00:00
Olle Johansson
9b95428cce Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there
but is still ok... Sorry. Lack of Tea, really.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:41:43 +00:00
Kevin P. Fleming
0ec502099f Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 11:38:20 +00:00
Russell Bryant
d1ba4f90de - Remove debug variable that was only used in one place
- convert string handling to the ast_str API
 - Convert strdup() to ast_strdup() and check the result
 - Minor formatting changes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 03:28:39 +00:00
Mark Spencer
04e45cfda3 Add SendURL support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 02:23:08 +00:00
Kevin P. Fleming
5dc23536ec Merged revisions 65683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines

Merged revisions 65682 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines

ensure that variables are set on a newly created channel before we start a PBX on it

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 20:53:10 +00:00
Olle Johansson
a3f9350ec2 Related to issue #9235 btw.
Merged revisions 65123 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines

Merged revisions 65122 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines

Not getting an ACK to a 200 OK in the initial invite is critical to the call.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 18:18:59 +00:00
Olle Johansson
1bc7fdeb6b Merged revisions 65076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines

Merged revisions 65075 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines

Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.

A special Thank You to WeBRainstorm that gave me access to his system.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 15:20:39 +00:00
Olle Johansson
491827a28a Merged revisions 64974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines

Issue 9487 - stop media flows at hangup of call

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:41:31 +00:00
Olle Johansson
ac343d43c8 Makeup, darling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:28:56 +00:00
Olle Johansson
f3ec447a23 Another fix for the support for recordings controlled by INFO-packets
We still lack a setting to enable/disable this per peer


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 08:49:34 +00:00
Joshua Colp
a769766c53 Merged revisions 64754 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines

Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-17 16:11:26 +00:00
Olle Johansson
d83dcae6b1 Below patches with some re-structuring for trunk
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Merged revisions 64602 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines

Issue #9681 - Handle www-auth on BYE

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:58:22 +00:00
Olle Johansson
c472b899ef Merged revisions 64578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines

Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:09:42 +00:00
Olle Johansson
c352f7b0d5 Merged revisions 64543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines

Merged revisions 64535 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines

Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:02:06 +00:00
Olle Johansson
09aec2f622 Merged revisions 64516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines

Merged following patch with a lot of changes for 1.4
------

Merged revisions 64514 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines

Issue #9726 - rlister - Better logging for ACL denials

While at it, also added better logging and handling of peers that are not supposed to register.

My patch, stole the issue report from Russell. My apologies, Russell :-)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 08:51:39 +00:00
Olle Johansson
7532b0bc4b Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 07:58:43 +00:00
Olle Johansson
90bad9d2f5 Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 07:35:56 +00:00
Olle Johansson
fa2622cb1d Merged revisions 64324 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines

Change -2 to XMIT_ERROR to clarify a bit more

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 19:35:58 +00:00
Joshua Colp
4fbb449444 If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 18:21:30 +00:00
Olle Johansson
fc057b1890 Improve handling network errors on transmission to hosts that don't reply or are unreachable
With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-13 19:20:36 +00:00
Joshua Colp
b71e691b3e Merged revisions 64114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines

This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 22:28:09 +00:00
Joshua Colp
38e951cfda Merged revisions 64086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines

Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 21:12:18 +00:00
Joshua Colp
82a30356da Merged revisions 64044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines

Ensure the onhold flag is set no matter what when being put on hold.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 16:33:34 +00:00
Olle Johansson
aa320037d2 Merged revisions 63749 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines

Merged revisions 63748 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines

Do not allocate SIP pvt's for PEERs we can not reach. 

This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 20:51:59 +00:00
Joshua Colp
7e10164e20 Merged revisions 63611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines

Merged revisions 63610 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines

Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 16:56:45 +00:00
Olle Johansson
c358b18a5a Merged revisions 63532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines

Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 13:07:44 +00:00
Russell Bryant
314c874d7d I noted this on the dev list but got no response, so I just did it myself.
Lock the call features when being used in chan_sip.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-08 16:41:35 +00:00
Olle Johansson
d326d84ae0 - Adding some missing spaces
- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-05 08:05:38 +00:00
Steve Murphy
02337303ef a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 17:49:20 +00:00
Steve Murphy
3ee0077f04 Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 16:37:23 +00:00
Olle Johansson
1b15d8852d Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
is needed to follow the call through the PBX.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 13:56:25 +00:00
Joshua Colp
81cade7a4c Merged revisions 62989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines

Merged revisions 62987 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines

When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 16:45:39 +00:00
Olle Johansson
e1ec3f917c Add a small message that we're doing something. On my systems, there's a long
dead period with a non-responsive CLI after I issue "load chan_sip.so"


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 12:12:02 +00:00
Olle Johansson
1d51b2e161 More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
But first, some serious SIP testing :-)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 12:00:03 +00:00
Olle Johansson
8fee67c83b Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 09:41:03 +00:00
Olle Johansson
daefa6a8b4 Merged revisions 62624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines

Don't unlock a channel that we already know does not exist (propably isue 8228)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 09:35:14 +00:00
Russell Bryant
b419fc1134 Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30 16:16:26 +00:00
Russell Bryant
5cb08adc7a Don't crash when invalid arguments are provided to the CHANNEL() function
for a SIP channel.
(issue #9619, reported by jtodd, original patch by Corydon76, committed patch
 slightly modified by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30 15:37:23 +00:00
Russell Bryant
b6b1bf3213 Merge changes from team/russell/events
This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.

This set of changes introduces the first use of the API, as well.  I have
restructured the way that MWI (message waiting indication) is handled.  It is
now event based instead of polling based.  For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes.  app_voicemail will generate events
when changes occur.

See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective.  For developer information, see the text in
include/asterisk/event.h.

As always, additional feedback is welcome on the asterisk-dev mailing list.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-28 21:01:44 +00:00
Olle Johansson
240bd841b0 Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27 14:40:28 +00:00