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1dc2b9c0f76f25a93a99facfcc7c4e4971c9d936
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2919 Commits

Author SHA1 Message Date
Joshua Colp
1dc2b9c0f7 Actually check to make sure a PBX was started on one of the Local channels instead of blindly assuming it was. (issue #10112 reported by makoto) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@73318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-05 13:26:02 +00:00
Christian Richter
ce99e9d955 bchannel configurations like echocancel and volume control, need to be setuped on inbound calls too. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@73252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-04 14:50:58 +00:00
Christian Richter
479d7e4738 bad bug in overlapdial case, we called start_pbx multiple times, because the state wasn't changed.. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@73207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-04 08:20:54 +00:00
Christian Richter
2676d6c595 fixed issue, that misdn_l2l1_check could only be called from mISDN Source channels.. #9449 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@73004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-03 08:04:35 +00:00
Russell Bryant
defd4eb3e2 Backport changes that make chan_iax2 not start the PBX on an incoming channel 
until the three-way call setup is completed.  These changes are already in 1.4
and trunk.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@72629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-29 16:30:56 +00:00
Christian Richter
fc4111b44f check if the bchannel stack id is already used, if so don't use it a second time. Also added a release_chan lock, so that the same chan_list object cannot be freed twice. chan_misdn does not crash anymore on heavy load with these changes. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@72585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-29 13:08:26 +00:00
Christian Richter
3637816f4c simplified generation for dummy bchannels, also we mark them as dummies, so they are not used later as real-bchannels, optimized the RESTART mechanisms, we block a channel now on cause:44, and send out a RESTART automatically, then on reception of RESTART_ACKNOWLEDGE we unblock the channel again. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@72099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-27 13:22:37 +00:00
Christian Richter
5f272436b9 simplified channel finding and locking a lot. removed unnecessary #ifdefed areas. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@72087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-27 09:26:53 +00:00
Christian Richter
085065ac35 isdn_lib.c didn't compile 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@72041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-27 07:54:30 +00:00
Christian Richter
16ecedee04 for inbound TE calls, we setup the bchannel when we get the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. removed some #if 0 areas which weren't used anymore. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@72040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-27 07:49:27 +00:00
Joshua Colp
76b4eb5daa Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@71414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-25 01:02:49 +00:00
Christian Richter
d8d4454ae5 we activate the bchannels in TE mode on incoming calls only when we want to connect the call. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@70672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-21 13:11:29 +00:00
Joshua Colp
6357ad5659 Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@70551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 22:20:16 +00:00
Russell Bryant
cfaead2b9c Fix a problem where an established call would not be properly disconnected 
when a PRI disconnect is received depending on which cause code was received.
(issue #9588, original patch by softins, updated patch from jtexter3, and some
 additional feedback from mhardeman)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@70396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 18:45:38 +00:00
Christian Richter
f5f018a209 forgot one place .. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@70342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 15:42:39 +00:00
Christian Richter
7fc236e53b fixed a bug that was introduced by copy and paste in the last commit ..bchannels weren't cleaned properly. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@70341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 15:29:09 +00:00
Christian Richter
ede913f976 on receiption of cause:44 we mark the channel as in use and inform the user about the situation, we need to test the RESTART stuff then. Also shuffled the empty_chan_in_stack function after the bchannel cleaning functions, to avoid race conditions. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@70311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 14:47:59 +00:00
Russell Bryant
6e0248318b Backport fix for crashes related to subscriptions from 1.4 ... 
Fix a crash that could occur when handing device state changes.
When the state of a device changes, the device state thread tells the extension
state handling code that it changed.  Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent.  However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use
 the sip_pvt lock wrappers by eliel)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@69990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-19 16:45:37 +00:00
Christian Richter
3322095dea when we send out a SETUP, but get no response, we should cleanup everything after reception of a hangup. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@69887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-19 13:23:04 +00:00
Joshua Colp
dc41ce9857 Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@69765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 18:13:03 +00:00
Christian Richter
ba372aa9a4 restart indicator 0x80 is correct, at least that's what libpri does. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@69053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-13 09:55:54 +00:00
Christian Richter
37ded96cfa if the bridged partner is mISDN too we should not send dtmf tones, they are transmitted inband always 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@68887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-12 08:35:22 +00:00
Christian Richter
7bb272f942 if we have already some digits, we just stop the tones. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@68874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-12 07:48:52 +00:00
Christian Richter
9809905c76 added check for NULL Pointer when calling misdn_new. Asterisk does not allow us to create channels anymore when stop gracefully is used :). also modified the restart_indicator to 0 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@68732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-11 16:49:00 +00:00
Christian Richter
5cc2b1078e fixed problem that the dummybc chanels had no lock, checking for the lock now. Also fixed the channel restart stuff, we can now specify and restart particular channels too. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@68631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-11 09:18:01 +00:00
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
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
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
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
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
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
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
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
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