Joshua Colp
026000bae8
Ensure the onhold flag is set no matter what when being put on hold.
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18 years ago
Jason Parker
c2c91c7fef
Merged revisions 63828 via svnmerge from
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r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines
Fix an issue with trying to kill a thread before it gets created.
Issue 9709, patch by nic_bellamy.
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18 years ago
Olle Johansson
ca1ae5e81a
Merged revisions 63748 via svnmerge from
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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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18 years ago
Joshua Colp
67a0bbaa83
Do not prematurely go on hold if sendonly was not actually set.
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18 years ago
Matthew Fredrickson
a181c6a777
Merged revisions 63653 via svnmerge from
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r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines
Make sure we only create a DSP if it's requested on SUB_REAL
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18 years ago
Joshua Colp
c83c0072bb
Merged revisions 63610 via svnmerge from
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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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18 years ago
Christian Richter
3e63c9d542
Merged revisions 62945,63402,63519 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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18 years ago
Olle Johansson
d30faa1dc2
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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18 years ago
Joshua Colp
b50b92ab77
Minor backport of revision 59083 in trunk. Don't queue an unhold frame up if the call was never on hold to begin with.
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18 years ago
Joshua Colp
95de3fbf0c
Merged revisions 62987 via svnmerge from
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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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18 years ago
Christian Richter
460e677ea6
Merged revisions 61357,61770,62885 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line
some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
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18 years ago
Tilghman Lesher
8bcfcfca88
Merged revisions 62691 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.
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18 years ago
Steve Murphy
55f4eb3e3d
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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18 years ago
Olle Johansson
d5fda03428
Don't unlock a channel that we already know does not exist (propably isue 8228)
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18 years ago
Russell Bryant
3595c5fed1
Merged revisions 62417 via svnmerge from
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588 , reported and patched by softins)
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18 years ago
Russell Bryant
8936804377
Fix a bug that made the "language" setting in zapata.conf not
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functional. (issue #9626 , reported and fixed by sergee)
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18 years ago
Russell Bryant
6b76a95f09
Fix a weird problem where when a caller talking to someone sitting behind an
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agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615 , #9616 , reported by jiddings and BigJimmy, and fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
34f9e0c4dd
Merged revisions 62126 via svnmerge from
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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
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18 years ago
Olle Johansson
b0121ca8a3
Issue #9608 - fix some annoying DEBUG messages not controlled by option_debug (DEA). Thanks!
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18 years ago
Joshua Colp
0525347958
Merged revisions 62037 via svnmerge from
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
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18 years ago
Kevin P. Fleming
0e525813a8
Merged revisions 61913 via svnmerge from
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286 )
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18 years ago
Russell Bryant
5574da0c12
Merged revisions 61866 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines
If the callerid= option is specified, but empty, clear any previous data.
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18 years ago
Russell Bryant
e6aa1145ed
Merged revisions 61862 via svnmerge from
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines
Ensure that callerid settings are reset on a reload.
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18 years ago
Russell Bryant
77c52fd8bf
Merged revisions 61798 via svnmerge from
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587 , reported and patched by xrg)
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18 years ago
Dwayne M. Hubbard
db55615090
Merged revisions 61777 via svnmerge from
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r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) | 1 line
removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor()
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18 years ago
Joshua Colp
b271179161
Merged revisions 61771 via svnmerge from
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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines
Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)
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18 years ago
Olle Johansson
32be6621f6
Merged revisions 61685 via svnmerge from
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r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 lines
Send NOTIFY to Contact: in SUBSCRIBE
- as reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp.
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18 years ago
Olle Johansson
dc6d309bdb
#9483 , half of patch by twilson to solve 302 redirect issues
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18 years ago
Joshua Colp
eb631c00a3
For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg)
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18 years ago
Steve Murphy
8a386a9816
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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18 years ago
Joshua Colp
c06e063e1a
Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
5858778f40
Merged revisions 61476 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines
If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all. It is an optional header, anyway. Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason. (issue #9488 , reported by makoto, fixed by me)
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18 years ago
Nadi Sarrar
3bd81c31fb
Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..
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18 years ago
Russell Bryant
a0d75effd5
Merged revisions 61426 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines
Fix a bug with switching between host=dynamic and using specific hosts for
peers. The code would only reset the peer's address when it is dynamic if
it was a new peer structure. Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515 , reported by caio1982, fixed by me)
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18 years ago
Russell Bryant
9ce476a89c
Merged revisions 61376 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
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18 years ago
Nadi Sarrar
4e2ab7de64
Export AOCD variables on misdn_hangup.
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18 years ago
Nadi Sarrar
b31fe9be83
Ignore facility messages in case we don't have a corresponding channel object.
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18 years ago
Nadi Sarrar
5e08746b28
AOCD's are now exported to asterisk channel variables.
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18 years ago
Nadi Sarrar
50d606bbf8
Merged revisions 61170 via svnmerge from
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r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 lines
msns config parameter defaults to '*'
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18 years ago
Olle Johansson
3e5c59f235
Merged revisions 61038 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 lines
- Don't send ActionID before Response: header.
- Don't use a blank in an AMI header
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18 years ago
Steve Murphy
7d5a79a0b9
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
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18 years ago
Tilghman Lesher
1d3fbc214a
Merged revisions 60846 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines
Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid.
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18 years ago
Tilghman Lesher
dc54d451dc
Off by one error, resulting in a crash (Issue 9500)
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18 years ago
Matthew Fredrickson
e7e3929740
Make sure we check the faxdetect option before doing fax processing
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18 years ago
Matthew Fredrickson
96a8eadf29
Merged revisions 60456 via svnmerge from
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r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 lines
There should only be one code path for doing DTMF conditionals on channels. This fixes it.
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18 years ago
Joshua Colp
a85156a737
Merged revisions 60213 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines
Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa)
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18 years ago
Russell Bryant
0b94c7c27c
Add a Content-Length of 0 to the response built by transmit_response_with_unsupported().
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(issue #9454 , reported by makoto, fixed by me)
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18 years ago
Russell Bryant
3e058dab84
Merged revisions 60083 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines
Fix the return value of handle_common_options() so that it always properly
indicates whether it handled the option or not.
(issue #9455 , reported by Netview, fixed by me)
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18 years ago
Russell Bryant
dff65e38f4
Merged revisions 59938 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines
Don't attempt to report configuration errors in build_user(). oej pointed out
that for a "friend" entry, this won't work, because all user options are valid
for peers, but not the other way around.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
f8773a1e80
Merged revisions 59916 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines
Make chan_sip report when it encounters an unknown option.
(issue #9440 , reported by nightcrawler)
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18 years ago
Nadi Sarrar
911a8f501d
Merged revisions 59788,59803 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines
Use the new sysfs way of mISDN 1.2 to check if a port is NT or not.
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r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines
ptp is the 5th bit, not the 4th.
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18 years ago
Christian Richter
30ace959bb
Merged revisions 59623-59624,59639 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line
we can now make 30 channels on a PRI (before we forgot chan 31..)
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r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line
don't be verbose if no need
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r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line
added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
8acd1e8349
When the IAX2 read callback gets called, return NULL instead of a "null frame".
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This will cause Asterisk to hangup the call instead of keep trying whatever it
was doing. Under normal conditions, this function would *never* be called.
However, the author of this patch says an error will occur that will cause it
to get called every 100 thousand calls or so. When this does happen, it puts
the channel in a loop that eventually brings down the system. So, hangup up
the call is certainly a better alternative. (issue #8286 , john)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
a2fc7b9e36
Fix the check that ensures that the CHANNEL function's first argument is "rtpqos".
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Thanks, Corydon. :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
8fdf6e3c47
Merged revisions 59258 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | 4 lines
Fix the use of the "sourceaddress" option when "bindaddr" is set to 0.0.0.0
instead of having each interface explicitly listed.
(issue #7874 , patch by stevens)
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18 years ago
Russell Bryant
c0aca26f30
Convert the RTPQOS function to just be additional parameter of the CHANNEL
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function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
10760fce2c
Merged revisions 59252 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 Mär 2007) | 1 line
fixed #9355
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18 years ago
Tilghman Lesher
0220349ca4
Oops, this should be case insensitive
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18 years ago
Tilghman Lesher
ec028bd1b2
Change this to a single dp function to make oej happy.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
cd3238d543
Rename the new dialplan functions to match the variable name
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
fa97f6c381
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Nadi Sarrar
980b0bc785
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
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* 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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
3b6f376425
Merged revisions 59194 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
cf58b2ba46
don't allow string input to overrun the buffer to hold it (ASA-2007-010)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
34f55a9bf6
remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
067876afbc
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
bfeb3bcf18
Copy len variable as well, should fix remaining IAX2 DTMF issues.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
30165b1bf2
Merged revisions 58849-58850,59062-59063 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line
added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers
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r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line
fixed the crypt_keys stuff
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r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one.
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r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line
modified a loglevel
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18 years ago
Olle Johansson
156c178143
Issue #9313 , Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
6548086af0
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
4fc7d081d3
Some people like to put "limitonpeer" instead of "limitonpeers" in their
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configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
0eda92609d
Merged revisions 58847 via svnmerge from
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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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19 years ago
Olle Johansson
f546453573
Don't hangup the call on OK or errors on MESSAGE and INFO
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inside of a dialog (like video update requests).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
83bb6718fc
Issue #9251 - Clear From URI from user attributes (tgrman)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Christian Richter
5923e48d7f
Merged revisions 57034,57523,57753,58558 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line
fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302 , #303 , #304
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r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line
fixed typo
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r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line
fixed another place where the out_cause was hardcoded to 16
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r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line
we can free channel 31 as well, since we can occupy it
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19 years ago
Christian Richter
160cfaf1c6
added UU transceiving and corect handling for rdnis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
fa866efb5c
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
8ae4e264d1
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
b9124234e9
Merged revisions 58579 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
2f54661cf8
Add a missing break statement so that handling the above event does not
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incorrectly destroy the channel. (issue #9242 , andrew)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
5603ca69eb
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
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(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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
f34c30860e
(This bug was reported to me by Kinsey Moore)
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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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19 years ago
Joshua Colp
240355d78b
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Steve Murphy
81ef9a4f18
Merged revisions 58115 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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19 years ago
Olle Johansson
275abf4e08
Merged revisions 58052 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
905238839b
Return an error of transmit_response is called without a session.
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(issue #9002 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
94c9dcda60
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
53b9bc89c0
Merged revisions 57475 via svnmerge from
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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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19 years ago
Joshua Colp
219f4b4913
Merged revisions 57317 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
43f841cadf
Merged revisions 57092 via svnmerge from
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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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19 years ago
Russell Bryant
83ce3720bc
Restore the behavior of Asterisk 1.2 where if a device was not specified in
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alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
dd6b58edcd
Fix a crash in my last change to iax2_indicate(). (issue #9150 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
268fc8a9cc
Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
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(Problem brought up by Ben Smithurst on the asterisk-dev list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
a9e4e0eefa
Make sure to set a speeddials parent on creation.
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Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
76a643978e
Merged revisions 56406 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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/branches/1.4@56407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
4565c1483c
Merged revisions 56230 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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19 years ago
Olle Johansson
d67ef52159
Move message from verbose to debug
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
9a7d1cc182
Restructure a little bit of code to reduce nesting. There is no functionality
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change here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
71921a8329
Merged revisions 56010 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines
If we receive a frame that is not in any of the negotiated formats, then drop
it. (potentially issue #8781 and SPD-12)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
b937d7e8bc
Fix locking issue, and accept "transport-accept" as a valid accept message.
...
This should solve issues 8970 and 8503.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago