Commit Graph

4624 Commits

Author SHA1 Message Date
Olle Johansson
73d0ba053b Issue 9487 - stop media flows at hangup of call
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:37:44 +00:00
Christian Richter
d682a74e26 Merged revisions 64902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line

we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 08:58:51 +00:00
Joshua Colp
7a8ca54257 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-17 16:10:12 +00:00
Olle Johansson
4ae20ba8e4 Fix auth on BYE. (Different patch than for 1.2)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:59:28 +00:00
Olle Johansson
5acc63d688 Issue #9681 - Handle www-auth on BYE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:38:18 +00:00
Olle Johansson
374b52f717 Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:05:47 +00:00
Olle Johansson
70ad556544 Issue #9439 - properly handle username parameters in SIP uri.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 09:57:22 +00:00
Olle Johansson
bf1a15b9bf 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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2007-05-16 09:12:34 +00:00
Olle Johansson
56af259505 Merged following patch with a lot of changes for 1.4
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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/branches/1.4@64516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 08:46:18 +00:00
Christian Richter
d17174cfc1 Merged revisions 64513 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line

in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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2007-05-16 08:44:51 +00:00
Olle Johansson
1f2afa0ff1 Change -2 to XMIT_ERROR to clarify a bit more
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 19:26:50 +00:00
Russell Bryant
b340fcd7d0 Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
will trigger an error and cause sounds to stop, which in this case, is ringing.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 19:13:00 +00:00
Olle Johansson
e041d57175 Handle network errors, like host or network unreachable, in a better way. This means that
calls to hosts or qualify (OPTION) messages will fail quicker if the TCP/IP stack tells us
that there is an issue.

Since this is an unconnected UDP socket, we will not get error messages directly
in most cases, but maybe on the second and third try.

This is already implemented in trunk.


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2007-05-14 18:52:09 +00:00
Steve Murphy
8c635fb2c1 As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 13:58:42 +00:00
Joshua Colp
9a73c07fb1 This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 22:27:04 +00:00
Joshua Colp
699aa6ad74 Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 21:10:45 +00:00
Joshua Colp
026000bae8 Ensure the onhold flag is set no matter what when being put on hold.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 16:32:15 +00:00
Jason Parker
c2c91c7fef Merged revisions 63828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-10 23:15:37 +00:00
Olle Johansson
ca1ae5e81a 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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2007-05-10 20:46:41 +00:00
Joshua Colp
67a0bbaa83 Do not prematurely go on hold if sendonly was not actually set.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 17:43:30 +00:00
Matthew Fredrickson
a181c6a777 Merged revisions 63653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-09 17:25:21 +00:00
Joshua Colp
c83c0072bb 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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2007-05-09 16:54:56 +00:00
Christian Richter
3e63c9d542 Merged revisions 62945,63402,63519 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 13:17:10 +00:00
Olle Johansson
d30faa1dc2 Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 13:04:14 +00:00
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07 21:26:58 +00:00
Joshua Colp
95de3fbf0c 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/branches/1.4@62989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 16:44:00 +00:00
Christian Richter
460e677ea6 Merged revisions 61357,61770,62885 via svnmerge from
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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2007-05-03 14:36:32 +00:00
Tilghman Lesher
8bcfcfca88 Merged revisions 62691 via svnmerge from
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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2007-05-02 17:43:48 +00:00
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 17:10:50 +00:00
Olle Johansson
d5fda03428 Don't unlock a channel that we already know does not exist (propably isue 8228)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 06:15:43 +00:00
Russell Bryant
3595c5fed1 Merged revisions 62417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30 15:58:28 +00:00
Russell Bryant
8936804377 Fix a bug that made the "language" setting in zapata.conf not
functional.  (issue #9626, reported and fixed by sergee)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-29 05:50:37 +00:00
Russell Bryant
6b76a95f09 Fix a weird problem where when a caller talking to someone sitting behind an
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
2007-04-27 21:10:51 +00:00
Olle Johansson
34f9e0c4dd Merged revisions 62126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-27 14:04:07 +00:00
Olle Johansson
b0121ca8a3 Issue #9608 - fix some annoying DEBUG messages not controlled by option_debug (DEA). Thanks!
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2007-04-27 08:10:43 +00:00
Joshua Colp
0525347958 Merged revisions 62037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-26 16:33:52 +00:00
Kevin P. Fleming
0e525813a8 Merged revisions 61913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-25 22:29:53 +00:00
Russell Bryant
5574da0c12 Merged revisions 61866 via svnmerge from
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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2007-04-25 21:59:07 +00:00
Russell Bryant
e6aa1145ed Merged revisions 61862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-25 21:13:15 +00:00
Russell Bryant
77c52fd8bf Merged revisions 61798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-25 16:22:07 +00:00
Dwayne M. Hubbard
db55615090 Merged revisions 61777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-24 18:43:53 +00:00
Joshua Colp
b271179161 Merged revisions 61771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-24 16:07:02 +00:00
Olle Johansson
32be6621f6 Merged revisions 61685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-19 08:37:08 +00:00
Olle Johansson
dc6d309bdb #9483, half of patch by twilson to solve 302 redirect issues
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-16 14:47:05 +00:00
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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2007-04-13 17:19:53 +00:00
Steve Murphy
8a386a9816 A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13 17:01:02 +00:00
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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2007-04-13 16:32:03 +00:00
Russell Bryant
5858778f40 Merged revisions 61476 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 16:05:29 +00:00
Nadi Sarrar
3bd81c31fb Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 15:39:14 +00:00
Russell Bryant
a0d75effd5 Merged revisions 61426 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 15:09:39 +00:00