Commit Graph

4570 Commits

Author SHA1 Message Date
Russell Bryant
7c884d76ea Merged revisions 59256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines

Convert the RTPQOS function to just be additional parameter of the CHANNEL
function.  This way, it will be possible for other RTP based channel drivers
to expose this information in the future.

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2007-03-27 16:25:02 +00:00
Tilghman Lesher
0e0600a446 Merged revisions 59227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines

Change this to a single dp function to make oej happy.

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2007-03-26 21:44:59 +00:00
Russell Bryant
46b15992c7 Merged revisions 59209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | 1 line

Rename the new dialplan functions to match the variable name
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2007-03-26 17:57:50 +00:00
Russell Bryant
08e3a9bdc8 Merged revisions 59207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines

The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup.  So, there are common situations where
the variables will not be available in the dialplan at all.  So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)

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2007-03-26 17:51:27 +00:00
Matthew Fredrickson
e4a7d1b35f Fix bug in which parameter type we are passing. This shouldn't be a problem
since both types are the same underneath.


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2007-03-26 16:48:29 +00:00
Matthew Fredrickson
df42d78569 Small API related SS7 updates.
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2007-03-26 16:46:07 +00:00
Nadi Sarrar
24d8595d00 Merged revisions 59202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines

* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
  (the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).

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2007-03-26 15:59:56 +00:00
Joshua Colp
cc22f60f30 Merged revisions 59195 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59195 | file | 2007-03-23 21:39:44 -0400 (Fri, 23 Mar 2007) | 10 lines

Merged revisions 59194 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines

Only try to handle a response if it has a response code. (ASA-2007-011)

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2007-03-24 01:42:11 +00:00
Kevin P. Fleming
e8e9e5e23c Merged revisions 59182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007) | 2 lines

don't allow string input to overrun the buffer to hold it (ASA-2007-010)

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2007-03-22 23:41:37 +00:00
Kevin P. Fleming
a6d4f89769 Merged revisions 59180 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22 Mar 2007) | 2 lines

remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)

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2007-03-22 23:39:26 +00:00
Joshua Colp
af9c17025f Minor tweak. Only queue up an unhold control frame if we are actually on hold. This would have shown itself when a call was initially being setup and the SDP data was being parsed in.
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2007-03-21 03:33:57 +00:00
Joshua Colp
1d5be2d1c7 Merged revisions 59081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2 lines

Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown)

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2007-03-21 03:27:58 +00:00
Joshua Colp
2874a4317d Merged revisions 59076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar 2007) | 2 lines

Copy len variable as well, should fix remaining IAX2 DTMF issues.

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2007-03-20 20:44:58 +00:00
Olle Johansson
dddb57b242 Merged revisions 59037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3 lines

Issue #9313, Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)


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2007-03-18 20:39:37 +00:00
Joshua Colp
4787adb4e9 Merged revisions 58923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2 lines

Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado)

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2007-03-15 18:21:03 +00:00
Russell Bryant
1bc728ed4b Merged revisions 58906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | 4 lines

Some people like to put "limitonpeer" instead of "limitonpeers" in their
configuration.  While we're at it, support "limitonpeerz" and 
"limitonpeerssssss".  (inspired by issue #9172)

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2007-03-14 19:19:00 +00:00
Olle Johansson
f9c3f60ab9 Merged revisions 58848 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, 13 Mar 2007) | 10 lines

Merged revisions 58847 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines

Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)

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2007-03-14 16:59:35 +00:00
Olle Johansson
b4490af8ac Merged revisions 58845 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3 lines

Don't hangup the call on OK or errors on MESSAGE and INFO 
inside of a dialog (like video update requests).

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2007-03-13 10:14:13 +00:00
Olle Johansson
32f9227ca4 Merged revisions 58843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2 lines

Issue #9251 - Clear From URI from user attributes (tgrman)

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2007-03-13 09:15:17 +00:00
Olle Johansson
d008232b27 Change URL to OpenH323 (thanks, Tzafrir!)
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2007-03-12 09:37:13 +00:00
Joshua Colp
ea226e9d77 Merged revisions 58779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines

Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)

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2007-03-12 00:54:13 +00:00
Russell Bryant
5dc7f1d898 Merged revisions 58705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | 6 lines

Fix a few more places in chan_iax2 where the ast_frame used for receiving a
frame was not properly initialized.
 - Interpolating a frame when the jitterbuffer is in use
 - decrypting a frame when IAX2 encryption is on
 - frames in an IAX2 trunk

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2007-03-10 18:15:41 +00:00
Joshua Colp
5629ee0292 Merged revisions 58584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, 09 Mar 2007) | 10 lines

Merged revisions 58579 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines

If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.

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2007-03-09 20:53:37 +00:00
Russell Bryant
d04a9e0bc0 Merged revisions 58510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | 3 lines

Add a missing break statement so that handling the above event does not
incorrectly destroy the channel. (issue #9242, andrew)

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2007-03-08 22:08:28 +00:00
Russell Bryant
9266a6b8d8 Merged revisions 58320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | 6 lines

If we receive ZT_EVENT_REMOVED, destroy the specified channel. 
(issue #7256, tzafrir)

Also, update the configure script to make sure that we don't try to build 
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.

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2007-03-08 01:06:00 +00:00
Russell Bryant
f86aee41f7 Merged revisions 58243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines

(This bug was reported to me by Kinsey Moore)

Merged revisions 58242 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines

Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call.  This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it.  However, this is not always
correct.  So, I have made it change this name after authentication happens
since at that point, we have an exact match.

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2007-03-07 18:20:51 +00:00
Joshua Colp
e7da006562 Merged revisions 58240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines

Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)

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2007-03-07 17:55:11 +00:00
Steve Murphy
5bb1d16f47 Merged revisions 58121 via svnmerge from
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r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, 06 Mar 2007) | 9 lines

Merged revisions 58115 via svnmerge from 
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line

Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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2007-03-06 23:19:59 +00:00
Olle Johansson
52750892b0 Merged revisions 58053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, 06 Mar 2007) | 10 lines

Merged revisions 58052 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines

Change error message to proper message

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2007-03-06 20:45:29 +00:00
Russell Bryant
d7aa9b6e2d Merged revisions 58023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines

Return an error of transmit_response is called without a session.
(issue #9002)

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2007-03-06 18:02:35 +00:00
Joshua Colp
b7e47198da Add zap show version CLI command. This pulls the version/echo canceller in use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai)
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2007-03-05 20:13:51 +00:00
Joshua Colp
2adf4fa4e0 Merged revisions 57914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 lines

Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)

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Joshua Colp
e23433e50e Merged revisions 57477 via svnmerge from
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r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines

Merged revisions 57475 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines

If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.

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Joshua Colp
b3bc675402 Merged revisions 57318 via svnmerge from
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r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines

Merged revisions 57317 via svnmerge from 
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines

Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)

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2007-03-01 22:23:26 +00:00
Joshua Colp
fd2db34f15 Merged revisions 57093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r57093 | file | 2007-02-28 13:57:52 -0500 (Wed, 28 Feb 2007) | 10 lines

Merged revisions 57092 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines

Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)

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2007-02-28 19:00:24 +00:00
Tilghman Lesher
a3da18c244 Issue 7789 - some telcos want the TON set based on the number, but without the NANP prefix removed
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2007-02-27 00:11:32 +00:00
Russell Bryant
b3776415aa Merged revisions 56888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | 4 lines

Restore the behavior of Asterisk 1.2 where if a device was not specified in
alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0.  (issue #9139)

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2007-02-26 20:43:18 +00:00
Russell Bryant
c617945b5c Merged revisions 56847 via svnmerge from
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r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) | 2 lines

Fix a crash in my last change to iax2_indicate(). (issue #9150)

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Russell Bryant
82179d1202 Merged revisions 56785 via svnmerge from
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r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) | 3 lines

Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
(Problem brought up by Ben Smithurst on the asterisk-dev list)

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2007-02-26 16:54:17 +00:00
Russell Bryant
5873c9c90f There is no need to look in the iaxs array for the pvt struct when we already
have a pointer to it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-26 00:37:03 +00:00
Olle Johansson
75d387acbc Doxygen additions, corrections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 20:29:41 +00:00
Olle Johansson
e916cf45da Doxygen updates and corrections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 19:49:11 +00:00
Olle Johansson
bc01e39174 Creating new doxygen macro "\extref" to create page that lists
external libraries and URLs to these. Please help me add these
references.

We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 19:27:50 +00:00
Jason Parker
97ab07a9e8 Allow a Skinny device to monitor a dialplan hint (w00t!).
See skinny.conf.sample for configuration example.


Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 02:23:43 +00:00
Jason Parker
a23096985b Support devicestate requests.
Now you should be able to subscribe to a Skinny device/line.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 02:06:44 +00:00
Jason Parker
639aee2ec7 Merged revisions 56569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines

Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 02:04:34 +00:00
Russell Bryant
b85a5ffbe3 Make the hashing function calculate something that makes more sense.
(Thanks to bmd on #asterisk-dev for pointing out my pointless math).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-23 22:26:00 +00:00
Russell Bryant
9fb1404279 Merge team/russell/iax2_performance.
There is not a large amount of code here and the changes are not very invasive.
However, they should significantly improve performance of chan_iax2 under load.

IAX2 media frames only carry the *source* call number.  So, when one arrives,
the correct session that it is a part of has to be matched on IP address, port
number, and call number, instead of just a call number.  Had these frames
carried the *destination* call number, this would not be an issue, because that
would be a unique identifier that would make it easy to immediately identify
the correct session.

The way that chan_iax2 did this matching was extremely inefficient.  It starts
at the first available call number and traverses each call number sequentially,
locking and unlocking a mutex for each one, to try to match against it.  It
would do this regardless of whether the call number was in use or not.  So,
for a call with a local call number of 25000, every single incoming media
frame would require a traversal that required 25000 mutex lock and unlock
operations.  (Note that the max call number is about 32k).

I have introduced a hash table of active IAX2 calls to improve this lookup
process.  The hash is done on the IP address, port number, and call number.  
So, for the previous example, a few lock/unlock operations may be done versus 
25000 for each frame.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-23 21:20:33 +00:00
Russell Bryant
70ce880338 Merged revisions 56407 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines

Merged revisions 56406 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines

Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).

........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-23 20:21:35 +00:00
Joshua Colp
afc99294fa Merged revisions 56231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines

Merged revisions 56230 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines

Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.

........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22 18:53:22 +00:00