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${ noResults }
3573 Commits (66e993de951dcb3fe536f51b6f4b77a2c5e52180)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Mark Michelson
|
66e993de95 |
Fix potential memory leaks.
ABE-1998 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Kevin P. Fleming
|
1c9fe00920 |
Recorded merge of revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthias Nick
|
00bb578898 |
Prevents from division by zero
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Sean Bright
|
0e805a51ec |
Modify VoiceMailMain()'s a() argument to allow mailboxes to be specified by name.
(closes issue #14740) Reported by: pj Patches: issue14740_09022009.diff uploaded by seanbright (license 71) Tested by: seanbright, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Sean Bright
|
3ac28f1e0f |
Clarify documentation for VoiceMailMain()'s a() option.
We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221085 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
|
4b4432b67e |
Fix options 'm' and 's'. They were swapped in the code. Also document the fact that app_confbridge does not automatically answer the channel.
(closes issue #15964) Reported by: shrift git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
|
a9154a905a |
Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't actually being expunged. When imapgreetings was set to no, the file based message was not being deleted at all. All good now! (closes issue #14949) Reported by: noahisaac Patches: vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
f150b48bc0 |
Add bridge related dial flags to the bridge app
Most of the functionality here is gained simply by setting the feature flag on the bridge config. However, the dial limit functionality has been moved from app_dial to the features code and has been made public so both app_dial and the bridge app can use it. (closes issue #13165) Reported by: tim_ringenbach Patches: app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
|
1cf5422dc8 |
Merged revisions 220288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
9298b2602a |
Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward.
(closes issue #15739) Reported by: DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by tilghman (license 14) 20090922__issue15739.diff.txt uploaded by tilghman (license 14) Tested by: DLNoah, jeffg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
|
6c3a3dabe0 |
Merged revisions 219816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines When IMAP variables were changed during a reload, Voicemail did not use the new values. This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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340d4b1a93 |
Missing value setting line for maxsecs/maxmessage
(closes issue #15696) Reported by: fhackenberger Patches: maxsecs.patch uploaded by fhackenberger (license 592) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219412 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Sean Bright
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a48d489568 |
Get this compiling under dev-mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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cd88adfc6a |
Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up.
In addition, there's a bit of cleanup to the arguments and documentation, in which I discovered that the last feature added to this application duplicated an option (oops!) and changed that option so that it now works. (closes issue #14909) Reported by: junky Patches: __20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10) Tested by: amilcar, junky, flujan, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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e876206693 |
Merged revisions 218730 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines If the user enters the same password as before, don't signal an error when the change does nothing. (closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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95da50292e |
Merged revisions 218577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines Ensure FollowMe sets language in channels it creates. Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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a873ad7a9b |
Recorded merge of revisions 218331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
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3b2023290f |
Merged revisions 218223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines Ensure we don't pickup ourselves when doing pickup by exten. (closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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85f18fcb8f |
Merged revisions 217989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines Don't ring another channel, if there's not enough time for a queue member to answer. (Fixes AST-228) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217990 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Sean Bright
|
245b163755 |
Fix compilation of app_meetme.
Reported by ebroad in #asterisk-bugs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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555ed0464f |
Merged revisions 217156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines When MOH is playing on the channel, announcements sent through the conference are not heard. (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Sean Bright
|
40d83cf748 |
Use ast_free() instead of free().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Olle Johansson
|
98f18d56b8 |
Merged revisions 216430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Dwayne M. Hubbard
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b5c89b7d0c |
Merged revisions 215270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names In general channel names are in the form Foo/Bar-Z, but the channel name could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the channel name at the last hyphen. (closes issue #15810) Reported by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard (license 733) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
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72365bdede |
Fix storage of greetings when using IMAP_STORAGE
The store macro was not getting called preventing storage of IMAP greetings at all. This has been corrected along with fixing checking if the imapgreetings option is turned on to store the greeting in IMAP. Lastly, the attachment filename was incorrectly using the full path instead of just the basename, which was causing problems with retrieval of the greeting. (closes issue #14950) Reported by: noahisaac (closes issue #15729) Reported by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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89e87e32d2 |
Ensure that realtime mailboxes properly report status on subscription.
This patch modifies app_voicemail's response to mailbox status subscriptions (via the internal event system) to ensure that a subscription triggers an explicit poll of the mailbox, so the subscriber can get an immediate cached event with that status. Previously, the cache was only populated with the status of non-realtime mailboxes. (closes issue #15717) Reported by: natmlt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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1b08c27c1a |
Add original position, when logging a caller entering a queue.
(closes issue #15146) Reported by: arabe Patches: asterisk-trunk.patch uploaded by arabe (license 786) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
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b8e571fb44 |
Fix greeting retrieval from IMAP
Properly check for the current voicemail state and if it doesn't exist, create it. (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch uploaded by mmichelson (license 60) Tested by: jpeeler git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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f6a10614a4 |
Merged revisions 213283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213283 | jpeeler | 2009-08-20 14:53:34 -0500 (Thu, 20 Aug 2009) | 2 lines Make all the symbols for the C-client callbacks global ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
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3d0314d5c0 |
Merged revisions 213103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) | 8 lines Fixes memory leak caused by incorrectly freeing mixmonitor (closes issue #15699) Reported by: edantie Patches: mixmonitor.patch uploaded by edantie (license 862) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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80cdd9b61d |
Small doxygen changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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c2612973d0 |
Check the return value of opendir(3), or we may crash.
(closes issue #15720) Reported by: tobias_e git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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160eb55c47 |
Merged revisions 211953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug 2009) | 10 lines This patch adds additional checking when generating queue log TRANSFER events. The additional checks prevent generation of false TRANSFER events in certain situations. (closes issue #14536) Reported by: aragon Patches: queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) Tested by: aragon, mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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642bec4d6f |
AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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e05992889b |
Check for NULL frame, before dereferencing pointer.
(closes issue #15617) Reported by: rain git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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20102765bf |
Merged revisions 211038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername. This is a partial revert of revision 82590, which was an attempted cleanup, but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended as a method by which component interfaces could be queried from the queue. Membername isn't useful here, because that field cannot be used to obtain further information about the member. See the documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various AMI commands which take a member argument for further justification. (closes issue #15664) Reported by: rain Patches: app_queue-queue_member_list.diff uploaded by rain (license 327) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211040 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
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a737df8603 |
Allow Gosub to recognize quote delimiters without consuming them.
(closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
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eb449d514e |
Minor improvements to app_fax.
This patch makes some small changes to handle watchdog timeouts in a better way, and also uses a 'cleaner' method of including the spandsp header files. (closes issue #14769) Reported by: andrew Patches: app_fax-20090406.diff uploaded by andrew (license 240) v1-14769.patch uploaded by dimas (license 88) Tested by: freh, deti, caspy, dimas, sgimeno, Dovid git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
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104974cdb4 |
Merged revisions 209838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines Modify how Playtones() is used in Milliwatt() to resolve gain issue. When Milliwatt() was changed internally to use Playtones() so that the proper tone was used, it introduced a drop in gain in the output signal. So, use the playtones API directly and specify a volume argument such that the output matches the gain of the original Milliwatt() code. (closes issue #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff uploaded by russell (license 2) Tested by: rue_mohr ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Brooks
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48363c16e1 |
Fixes numerous spelling errors. Patch submitted by alecdavis.
(closes issue #15595) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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7c5f20e46b |
Cleanup T.38 negotiation changes.
Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages, clean up some looping logic, and correct an improper use of ast_free() for freeing an ast_frame. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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b6b3fed0c7 |
Make T.38 switchover in ReceiveFAX synchronous.
In receive mode, if the channel that ReceiveFAX is running on supports T.38, we should *always* attempt to switch T.38, rather than listening for an incoming CNG tone and only triggering on that. The channel may be using a low-bitrate codec that distorts the CNG tone, the sending FAX endpoint may not send CNG at all, or there could be a variety of other reasons that we don't detect it, but in all those cases if T.38 is available we certainly want to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Brooks
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d81d6d3415 |
Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"
(closes issue #15571) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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0b2b01a1fa |
Merged revisions 208592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines Do not log an ERROR if autoservice_stop() returns -1. This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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0a6e06c7ff |
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
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d009896670 |
Restore an int declaration on PPC platforms.
This x is one crafty little bugger... It was used for 2 different things (one of which was only done on PPC) in 1.4. One of the uses were removed in trunk, and with it went the declaration. (closes issue #14038) Reported by: ffloimair git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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33852cfaf6 |
Fix the crash in directed pickups. For real this time.
A shallow pointer copy was causing an ast_party_connected_line structure to be freed multiple times, thus causing a crash. (closes issue #15441) Reported by: lmsteffan Patches: 15441.patch uploaded by mmichelson (license 60) Tested by: lmsteffan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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d040266a17 |
Okay, that didn't fix the crash. It didn't really do anything useful.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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b276189912 |
Initialize connected line instance when doing a directed pickup.
This helps to prevent a crash which may occur due to our freeing garbage due to a struct being uninitialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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98e4ab5716 |
Flag field in wrong position.
Reported by "Hoggins!" on asterisk-dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |