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certified/13.1-cert3
${ noResults }
5059 Commits (352ca6584bcb1aa2742da08fbadc46f6415e643c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Joshua Colp
|
438361c0b8 |
Add expiry value to the sip show subscriptions CLI command.
(closes issue #12025) Reported by: agx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111083 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jason Parker
|
6412a96e43 |
Large cleanup of DSP code
Per comments from dimas: 1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones. 2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before. 3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best. Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too. 4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used. 5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel. This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality. (closes issue #11968) Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas (license 88) v4-11968-zap.patch uploaded by dimas (license 88) Tested by: dimas, qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
a3d7dc8903 |
Merged revisions 111020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be. (closes issue #11995) Reported by: fall ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
febd162ed2 |
Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Tilghman Lesher
|
ef4eff9a9b |
Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
|
13787bc595 |
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110726 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Mark Michelson
|
a49b6591f5 |
Oops here too. I need to stop coding for a while...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
67efba6e50 |
Merged revisions 110635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines When reverting a commit, I accidentally left in this bit which was an experiment to see what would happen. It passed the compile test, and I didn't notice I had left this change in too. So this is a revert of a revert...sort of. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
738e4ec94e |
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Olle Johansson
|
676d9d3303 |
Use the "Server" header when responding to SIP requests.
(closes issue #12278) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110625 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
c05501d812 |
Remove the "Event: registration" header from Asterisk-generated
SIP REGISTER requests. rjain points out that RFC 3265 specifies that the Event: header is not a valid header for REGISTER requests and that the "registration" value is not defined at IANA. (closes issue #12279) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
625f6bd203 |
Merged revisions 110618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines This is a revert for revision 108288. The reason is that that revision was not for an actual bug fix per se, and so it really should not have been in 1.4 in the first place. Plus, people who compile with DO_CRASH are more likely to encounter a crash due to this change. While I think the usage of DO_CRASH in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4 and should be done instead in a developer branch based on trunk so that all scheduler functions are fixed at once. I also am reverting the change to trunk and 1.6 since they also suffer from the DO_CRASH potential. (closes issue #12272) Reported by: qq12345 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Russell Bryant
|
c087390452 |
Merged revisions 110614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines Turn a NOTICE into a DEBUG message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
5a77d16eda |
Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog.
(closes issue #12280) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110610 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
2860d9f83c |
Merged revisions 110336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines Fix some very broken code that was introduced in 1.2.26 as a part of the security fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address structure that a background thread continuously updates. However, in these cases, a stack variable was passed. That means that the dnsmgr thread would be continuously writing to bogus memory. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110337 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
0be1d98624 |
Fix recent trunk breakage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
9e3603dac9 |
Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading. Early cleanup for issue #11968 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
8d4276578a |
Rename very poorly named function to reflect what it actually does. This was causing quite a bit of confusion for me...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jeff Peeler
|
515ec9d92f |
This change adds DNS manager support for registrations not referencing a peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
5f7e81b564 |
Set req->data to NULL after free'ing to ensure that it never gets accidentally
double free'd. (reported by dhubbard directly to me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
37fcf0e0fe |
Fix a typo which caused a double free in chan_zap. This was discovered
by Juggie while attempting to load chan_zap. Apparently this would happen if an error were encountered while trying to process zapata.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
8bceb4f2a1 |
Since a sip request's data field is now a stringfield, we not only have to check
if the string is zero-length, but also if the data field is non-null. (closes issue #12250) Reported by: caio1982 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109681 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
f1181cb04e |
Merged revisions 109575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar 2008) | 6 lines Make sure an agent doesn't try to send dtmf to a NULL channel closes issue #12242 Reported by Yourname ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Kevin P. Fleming
|
3a8268bc29 |
fix up various warnings found via the addition of format string checking... some of these were really, really bad code
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109475 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Terry Wilson
|
b02bc230af |
Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
10cdbe28a8 |
Merged revisions 109386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
263c658a6b |
Do not return with a successful authentication if the From header ends up empty.
(AST-2008-003) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109389 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
0de4eba640 |
Add manager peerstatus events when peer can't authenticate.
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Michiel van Baak
|
5729b0ec40 |
Update the directory of placed calls on skinny phones
when dialing a channel that does not provide progress (analog ZAP lines) The phone does handle the double update on calls to channels that do provide progress and wont insert duplicate items (closes issue #12239) Reported by: DEA Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
7980ac1261 |
Remove something that is never ever used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
|
ba63fd28c2 |
Merged revisions 109107 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 lines 200 OKs in response to a reinvite need to be sent reliably. If the remote side does not receive one the dialog will be torn down. (closes issue #12208) Reported by: atrash ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
f7f58d194e |
Make sure that the temporary sip_request structure is empty so that copy_request doesn't think it already has an ast_str.
(closes issue #12231) Reported by: IgorG git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
5bd4e32a4c |
Merged revisions 108796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines Fix a channel name issue. chan_oss registers the "Console" channel type, but it created channels with an "OSS" prefix. (closes issue #12194, reported by davidw, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
086d4f0e56 |
Merged revisions 108737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines Fix a race condition in the SIP packet scheduler which could cause a crash. chan_sip uses the scheduler API in order to schedule retransmission of reliable packets (such as INVITES). If a retransmission of a packet is occurring, then the packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if a response is received from the packet as previously transmitted, then when we ACK the response, we will remove the packet from the scheduler and free the packet. The problem is that both the ACK function and retrans_pkt attempt to acquire the same lock at the beginning of the function call. This means that if the ACK function acquires the lock first, then it will free the packet which retrans_pkt is about to read from and write to. The result is a crash. The solution: 1. If the ACK function fails to remove the packet from the scheduler and the retransmit id of the packet is not -1 (meaning that we have not reached the maximum number of retransmissions) then release the lock and yield so that retrans_pkt may acquire the lock and operate. 2. Make absolutely certain that the ACK function does not recursively lock the lock in question. If it does, then releasing the lock will do no good, since retrans_pkt will still be unable to acquire the lock. (closes issue #12098) Reported by: wegbert (closes issue #12089) Reported by: PTorres Patches: 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
5f58a11ff2 |
Merged revisions 108530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines Make a tweak that gets the LEDs on polycom phones to blink when an extension that has been subscribed to goes on hold. Otherwise, they just stay on like it does when an extension is in use. (closes issue #11263) Reported by: russell Patches: notify_hold.rev1.txt uploaded by russell (license 2) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
1a2b358588 |
Merge changes from team/jamesgolovich/chan_sip-ast_str
This set of changes removes the hard coded maximum packet size of 4kB from chan_sip. It now starts by allocating 1kB, and growing the buffer as needed to accommodate large packets. (closes issue #8556, reported by mikma, patch by jamesgolovich) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108439 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
8bbef5f996 |
Rename ast_tcptls_server_instance to session_instance, since this pertains to
server and client usage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
|
ff1527de3d |
Let's get this to compile
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108293 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
|
39cc1b4f36 |
Merged revisions 108288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip. The scheduler callback will always return 0. This means that this id is never rescheduled, so it makes no sense to loop trying to delete the id from the scheduler queue. If we fail to remove the item from the queue once, it will fail every single time. (Yes I realize that in this case, the macro would exit early because the id is set to -1 in the callback, but it still makes no sense to use that macro in favor of calling ast_sched_del once and being done with it) This is the first of potentially several such fixes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Kevin P. Fleming
|
a3a8aa6547 |
add support for named sections in zapata.conf, and fix a few bugs in config file parsing
(closes issue #9503) Reported by: tzafrir Patches: fix_cleanups uploaded by tzafrir (license 46) zapata_sections uploaded by tzafrir (license 46) skipchannel_options uploaded by tzafrir (license 46) conf_sample uploaded by tzafrir (license 46) patches updated by me to better conform to coding guidelines and fix some problems git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
|
5ebfa5638a |
Merged revisions 108086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP closes issue #11475 Reported by: andrebarbosa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108191 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
582d3b4ba7 |
Deadlock fixes
(closes issue #12143) Reported by: kactus Patches: 20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14) Tested by: kactus git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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1c0bc928d1 |
Merged revisions 107714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber). (closes issue #12014) Reported by: junky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
|
3ee2872f40 |
Merged revisions 107713 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar 2008) | 2 lines get chan_vpb to build properly in dev mode ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107715 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
|
acb73c7448 |
fix another potential bug found by gcc 4.3
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
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4925e7b835 |
Merged revisions 107464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines fix various other problems found by gcc 4.3 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Terry Wilson
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a9a3f001e0 |
Merged revisions 107290 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) | 2 lines If we fail to alloc a channel, we should re-lock the pvt structure before returning. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
6477f207f1 |
Merged revisions 107173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | 5 lines Make sure to reenable echo can after a "failed" (canceled, etc) three-way call. (closes issue #11335) Reported by: rebuild ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107177 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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362b184c9c |
If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38.
(closes issue #8677) Reported by: alex-911 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Kevin P. Fleming
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f5bee50ad1 |
Merged revisions 106945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down" ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |