Tilghman Lesher
fc6d28a932
If the value of a variable passed to FIELDQTY is blank, then FIELDQTY should return 0, not 1.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@66537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 21:49:35 +00:00
Olle Johansson
c4e7d9fef5
Issue #9802 - Change inuse counter on CANCEL
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@66349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 07:53:14 +00:00
Tilghman Lesher
2cb2558eb1
Issue 9791 - Fix pronunciation of seconds in Dutch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@66127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-25 13:46:35 +00:00
Joshua Colp
ad2f350d39
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/1.2@65837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:40:38 +00:00
Christian Richter
17175c7d54
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 09:19:58 +00:00
Kevin P. Fleming
cba8e2f704
ensure that variables are set on a newly created channel before we start a PBX on it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 20:46:22 +00:00
Kevin P. Fleming
9edd1e094c
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 20:06:13 +00:00
Russell Bryant
2f0f1f5e00
Revert revision 62417 as someone reported problems with it to Mark. This was
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related to issue #9588 .
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 13:06:17 +00:00
Russell Bryant
03ad135134
Fix a memory leak that I just noticed in the device state handling in app_queue.
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On most device state changes, it would leak roughly 8 to 64 bytes (the length of
the name of the device).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-22 13:07:03 +00:00
Christian Richter
0b6da8d56e
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-22 07:46:39 +00:00
Steve Murphy
fd1fc0a9c1
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 20:56:20 +00:00
Olle Johansson
86882515a8
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/1.2@65122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 18:10:46 +00:00
Olle Johansson
21ea4dc3f1
Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 15:12:09 +00:00
Christian Richter
58bcd919d5
fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 11:23:11 +00:00
Christian Richter
06b2955d26
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/1.2@64902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 08:24:08 +00:00
Tilghman Lesher
b2f84c347f
How is it that we never caught that this is returning the opposite of our documentation, until now?
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@64819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-17 21:14:36 +00:00
Jason Parker
dfd24c33d2
If we have a negative current message, we shouldn't go back even further...
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Issue 9727.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@64758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-17 16:52:38 +00:00
Olle Johansson
9ebfde54a1
Fixing possible bug in auth of BYE
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@64603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:55:16 +00:00
Olle Johansson
80e4abca3d
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/1.2@64535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 09:08:22 +00:00
Olle Johansson
aa9ff74af5
Issue #9726 - rlister - Better logging for ACL denials
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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 :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@64514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 08:25:56 +00:00
Christian Richter
b60fd4bc20
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@64513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 08:23:42 +00:00
Joshua Colp
3b1ad79633
Only perform stripping of - strings from the channel name for Zap channels. Anywhere else we might remove a legitimate part of a device name. (issue #9668 reported by stevedavies)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@64275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 18:34:06 +00:00
Tilghman Lesher
29aa7c809b
Issue 9121 - fixups for safe_asterisk script
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-11 16:31:03 +00:00
Jason Parker
074cc21291
Fix an issue with trying to kill a thread before it gets created.
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Issue 9709, patch by nic_bellamy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 23:14:55 +00:00
Olle Johansson
07ba0e379b
Do not allocate SIP pvt's for PEERs we can not reach.
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This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 20:38:54 +00:00
Matthew Fredrickson
818c25352e
Make sure we only create a DSP if it's requested on SUB_REAL
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 17:20:20 +00:00
Joshua Colp
7dc491d090
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 16:51:03 +00:00
Tilghman Lesher
47047d5004
Replicate fix from 51158 (app_voicemail) to app_directory (Issue 9224)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 14:48:06 +00:00
Christian Richter
fcad34fd9f
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/1.2@63519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 11:26:16 +00:00
Tilghman Lesher
25930c842f
Issue 9602 - segfault in app_macro
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-08 22:19:15 +00:00
Christian Richter
048000f9d6
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-08 15:07:37 +00:00
Tilghman Lesher
616e3502e7
Issue 9527 - upon entering a folder, no message is selected (curmsg == -1), so deleting causes memory corruption (beyond bounds)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-08 06:20:16 +00:00
Joshua Colp
71befa9884
Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07 21:39:52 +00:00
Olle Johansson
999633ab4f
Releasing the whole in-memory configuration while you're adding to it is not a good thing.
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Please review this change. Caused by additions (+) to non-existing contexts.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@63195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07 14:06:50 +00:00
Joshua Colp
3a218f3a09
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/1.2@62987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 16:42:19 +00:00
Christian Richter
2102128500
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 15:39:21 +00:00
Christian Richter
70e95c4846
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 13:59:00 +00:00
Kevin P. Fleming
de32406990
doh... initializing the pointer variable will work just a bit better
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 00:23:00 +00:00
Kevin P. Fleming
0b78d1542c
increase reliability and efficiency of static Realtime config loading via ODBC:
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don't request fields we aren't going to use
don't request sorting on fields that are pointless to sort on
explicitly request the fields we want, because we can't expect the database to always return them in the order they were created
(reported by blitzrage in person (!), patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 23:53:46 +00:00
Steve Murphy
26cab38b64
Some tweaks to satisfy CDR bug 8796, where being in 'h' extension louses up the dst field
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 20:10:32 +00:00
Tilghman Lesher
9b71a5799b
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
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IAX connection, the remote end will receive garbage characters tacked onto the
end.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 17:38:16 +00:00
Russell Bryant
c8a90488d6
Remove an unnecessary check that makes it so if you hang up after doing an
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attended transfer before the target extension answers the channel, the transfer
is not successful. (issue #9338 , patch by svanlund)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-01 21:55:19 +00:00
Russell Bryant
8da91a20e3
Add indications.conf information for the Philippines.
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(issue #9525 , reported and patched by loloski)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-01 16:26:23 +00:00
Russell Bryant
274afdc345
This patch fixes an issue where depending on the cause code, when the network
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sends a PRI disconnect, the call may not be properly hung up.
(issue #9588 , reported and patched by softins)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30 15:57:26 +00:00
Joshua Colp
80d88816df
Update copyright notice. It's now the year 2007!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30 14:34:07 +00:00
Jason Parker
d97b6e3d34
This transcoder message needn't be a NOTICE.
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I've seen it cause confusion more than a few times.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27 16:16:16 +00:00
Olle Johansson
3bafdca29f
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
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final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27 13:57:45 +00:00
Joshua Colp
babe9cbbef
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@62037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-26 16:30:57 +00:00
Joshua Colp
897b4c0ca3
Don't count failed include attempts against the configuration include level. (issue #9593 reported by mostyn)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@61958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-26 01:25:03 +00:00
Kevin P. Fleming
cd3ef82ba4
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@61913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-25 22:24:59 +00:00