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Author SHA1 Message Date
Joshua Colp
084ede4507 Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-07 00:09:13 +00:00
Russell Bryant
3a2e0e1ed9 We have some bug reports showing crashes due to a double free of a channel. 
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 16:40:51 +00:00
Joshua Colp
76fdb9418b Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 13:28:34 +00:00
Joshua Colp
a05ab92728 Revert channel name splitting fix for Zap. The moral of the story is don't use - in your user/peer names. (issue #9668 reported by stevedavies) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 12:18:36 +00:00
Christian Richter
f002ad09a3 briding is a bool, fixed copy and paste issue. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 15:42:03 +00:00
Christian Richter
e7590d0aec simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 15:39:43 +00:00
Nadi Sarrar
e0f4f4969c Backport of the overlap_dial functionality from asterisk-1.4's chan_misdn. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 11:18:45 +00:00
Christian Richter
3cd1c84e8d added possibility to deactivate bridging per port 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 10:05:45 +00:00
Tilghman Lesher
dd412388a1 According to MATH, 0+1181000386 = 1181000448. Oops. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 23:41:49 +00:00
Tilghman Lesher
c78acd2896 Add revision Id tags (by request of tzafrir) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 17:10:30 +00:00
Joshua Colp
22fe1b73cc It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@66764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-31 16:12:39 +00:00
Tilghman Lesher
c0ce087e43 Issue 9818 - Fix for issue 8329 breaks pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime, but only because we lack core API to do it. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@66744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-31 15:58:45 +00:00
Tilghman Lesher
fc6d28a932 If the value of a variable passed to FIELDQTY is blank, then FIELDQTY should return 0, not 1. 
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 
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 
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) 
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. 
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 
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 
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 
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. 
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 
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. 
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. 
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 
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. 
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. 
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? 
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... 
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 
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!) 
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 
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. 
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) 
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 
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. 
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. 
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 
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) 
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) 
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. 
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 
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. 
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) 
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) 
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. 
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) 
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. 
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. 
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 
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