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${ noResults }
3514 Commits (c9a23f6d78da2bf98d3784dc2ca74194e212800d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Kevin P. Fleming
|
c9a23f6d78 |
Backport audio handling loop fixes from trunk version of app_fax.
This backport resolves some issues handling audio frames during FAX processing, and ensures that the FAX application doesn't accidentally get notified of a T.38 switchover at the end of a successful FAX. (closes issue #16127) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@225871 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
|
12fb1e8bde |
Merged revisions 225445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@225489 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
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c6a9fbf2c9 |
Merged revisions 225360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@225363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Joshua Colp
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73e58ba5e9 |
Merged revisions 224567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@224571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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bc5b04795d |
Merged revisions 224448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines Allow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's. This corrects an issue reported on the -users list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@224451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
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a2bbda8b49 |
Merged revisions 224178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@224181 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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544acf5a22 |
Merged revisions 223832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@223835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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f555c4431f |
Merged revisions 223652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@223655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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02db749cd7 |
Merged revisions 223330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@223333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
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503d9f8ab0 |
Merged revisions 223215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@223257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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6410aa2418 |
Merged revisions 222176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines 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/branches/1.6.2@222187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthias Nick
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0b1d6d601c |
Merged revisions 221436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 | mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines Prevents from division by zero ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Sean Bright
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6e0b29a9cd |
Merged revisions 221085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep 2009) | 9 lines 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/branches/1.6.2@221089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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4ddccae801 |
Remove optional_api from 1.6.2 branch, since it is not currently working.
This is a blocking issue for the 1.6.2 release. (closes issue #15914) Reported by: mbeckwell Branch: http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162 Tested by: mbeckwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
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bfa2271b46 |
Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked.
(closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@220934 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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cb30042138 |
Merged revisions 220904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep 2009) | 5 lines 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/branches/1.6.2@220905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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f9298dc0cb |
Merged revisions 220833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines 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/branches/1.6.2@220836 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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94740de344 |
Merged revisions 220289 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines 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/branches/1.6.2@220292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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aebef98bca |
Merged revisions 219987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009) | 8 lines 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/branches/1.6.2@219989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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2e1d6748ec |
Merged revisions 219818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines 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/branches/1.6.2@219821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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c1dc6a4b14 |
Merged revisions 219412 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) | 6 lines 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/branches/1.6.2@219415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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eb99291c98 |
Merged revisions 218731 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500 (Tue, 15 Sep 2009) | 13 lines 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/branches/1.6.2@218733 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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14e5422156 |
Merged revisions 218579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) | 16 lines 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/branches/1.6.2@218582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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b292004c84 |
Merged revisions 218361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines 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/branches/1.6.2@218364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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61055d3660 |
Merged revisions 218224 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500 (Mon, 14 Sep 2009) | 14 lines 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/branches/1.6.2@218238 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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b509ba227f |
Merged revisions 217990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) | 10 lines 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/branches/1.6.2@218052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Sean Bright
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78927bffe3 |
Merged revisions 217286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 | seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4 lines Fix compilation of app_meetme. Reported by ebroad in #asterisk-bugs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@217299 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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4e6e286b51 |
Merged revisions 217199 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) | 14 lines 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/branches/1.6.2@217217 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Olle Johansson
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9e57fd58e2 |
Merged revisions 216438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines 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/branches/1.6.2@216647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Sean Bright
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3532b989b8 |
Merged revisions 216593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep 2009) | 1 line Use ast_free() instead of free(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Dwayne M. Hubbard
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b999f4a99d |
Merged revisions 215338 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500 (Tue, 01 Sep 2009) | 18 lines 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/branches/1.6.2@215376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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6f4a27e6a0 |
Merged revisions 213833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) | 14 lines 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/branches/1.6.2@213836 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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5f84d4ae53 |
Merged revisions 213697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug 2009) | 12 lines 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/branches/1.6.2@213698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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c3e98c3534 |
Merged revisions 213404 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009) | 12 lines 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/branches/1.6.2@213413 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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5de49da50d |
Merged revisions 213113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500 (Wed, 19 Aug 2009) | 14 lines 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/branches/1.6.2@213128 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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d701cc5e7c |
Merged revisions 212627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009) | 4 lines 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/branches/1.6.2@212631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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7a0b0feb3c |
Merged revisions 211957 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug 2009) | 17 lines 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/branches/1.6.2@211959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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9e94589304 |
AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@211580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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834ad31c4d |
Merged revisions 211232 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 | tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines Check for NULL frame, before dereferencing pointer. (closes issue #15617) Reported by: rain ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@211235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
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6abc7aa68c |
Merged revisions 211113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) | 11 lines Recorded merge of revisions 211112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) | 4 lines Resolve a deadlock involving app_chanspy and masquerades. (ABE-1936) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@211122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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74659ff9d6 |
Merged revisions 211040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) | 21 lines 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/branches/1.6.2@211051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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fe6107aba0 |
Merged revisions 210908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines 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/branches/1.6.2@210911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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d058620f28 |
Merged revisions 209839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r209839 | russell | 2009-08-01 06:02:07 -0500 (Sat, 01 Aug 2009) | 20 lines 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/branches/1.6.2@209842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Brooks
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503bb8de9c |
Merged revisions 209554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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144defd78a |
Merged revisions 209279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 | kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 lines 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/branches/1.6.2@209282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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eb0a873dab |
Merged revisions 209256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 | kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 lines 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/branches/1.6.2@209265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Brooks
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f9ba130264 |
Just replacing typos "recieved" with "received".
From issue #15360, forgot to apply to trunk and other branches. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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b560ce9cde |
Merged revisions 208593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines 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/branches/1.6.2@208596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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03d73280b3 |
Merged revisions 208464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines 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/branches/1.6.2@208501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jason Parker
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4ec6629fed |
Merged revisions 208113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 | qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines 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/branches/1.6.2@208116 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |