Commit Graph

3890 Commits

Author SHA1 Message Date
Olle Johansson
e4f9cc15e6 Issue #8536 - Caller ID not set in CDR for jingle
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:48:55 +00:00
Joshua Colp
0df2a42f96 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/branches/1.4@65839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:42:12 +00:00
Olle Johansson
6cfe6a550e Issue 8409 - phsultan - Fix "login" as component to jabber server.
...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits
   of BYE requests.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:38:30 +00:00
Christian Richter
9f54cd55af Merged revisions 65767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line

we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
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2007-05-24 09:37:32 +00:00
Kevin P. Fleming
e1518f42ae start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 20:59:19 +00:00
Kevin P. Fleming
ca6b421be4 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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2007-05-23 20:51:56 +00:00
Kevin P. Fleming
b89faf596b clear the 'delay PBX' flag when we are ready to start the PBX
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 20:35:50 +00:00
Kevin P. Fleming
09dc4253d8 don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 20:30:24 +00:00
Kevin P. Fleming
f608d90283 Merged revisions 65676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines

if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it

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2007-05-23 20:07:59 +00:00
Russell Bryant
7398856cea Merged revisions 65588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines

Revert revision 62417 as someone reported problems with it to Mark.  This was
related to issue #9588.

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2007-05-23 13:07:13 +00:00
Russell Bryant
c556cc222c List res_smdi as a dependency for app_voicemail and chan_zap
(Thanks to mnicholson for pointing it out)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-22 18:40:38 +00:00
Christian Richter
e7355ec53b Merged revisions 65328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line

we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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2007-05-22 08:12:20 +00:00
Olle Johansson
4483fa12e8 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/branches/1.4@65123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 18:16:09 +00:00
Olle Johansson
7fe3608300 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/branches/1.4@65076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 15:18:13 +00:00
Olle Johansson
50f79ba4b2 - Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp.
- In order to not duplicate code, I restructured some of the code for putting calls on/off hold.

Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 12:58:39 +00:00
Christian Richter
012fe116fa Merged revisions 65007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line

fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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2007-05-18 12:40:46 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 09:12:34 +00:00
Olle Johansson
56af259505 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/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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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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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)


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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27 14:04:07 +00:00