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${ noResults }
5190 Commits (6353bddc57751ea2db044637fdd3f5b35b96f44c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
fced823c08 |
Change the default for the pridialplan parameter to the far more common case of
'unknown', and better document the use of each parameter. (closes issue #12633) Reported by: tzafrir Patches: pridialplan_unknown_2.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117182 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Luigi Rizzo
|
775542f753 |
trap potential failures of SDL when SDL_WINDOWID is pointing to a
random window. This commit is essentially a workaround for some undesirable behaviour of SDL; we should not be doing this in the application, but in the library. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
0894cae92c |
Merged revisions 117081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6 lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682) Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby (license 430) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Luigi Rizzo
|
391f5ffcf1 |
Some fixes to the code to support running on an externally
supplied window. SDL (at least recent 1.2.x versions) has the ability to run the graphic output into an externally supplied window, whose ID in the environment variable SDL_WINDOWID. Ideally, applications should run unchanged irrespective of who creates the window. Unfortunately, SDL does not subscribe to mouse, key and resize events on externally supplied windows, so we need to do ask for these events explicitly. On passing, also add some code to handle SDL_ACTIVEEVENT so if the X11 window is killed while we are active, we call "stop now" to terminate the asterisk instance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Luigi Rizzo
|
db8475bb4e |
Allow users to specify 'startgui=1' in oss.conf so that the
graphic screen for the video console is activated at startup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
affbbe3bd2 |
Merged revisions 116978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) | 4 lines Avoid access of uninitialized memory. This caused a bunch of crashes for me while doing load testing of development branch where I'm working on some performance improvements. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116979 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
29a9d477df |
Remove duplicate colon on Reason header
(closes issue #12678) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116919 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
4943cbcf2c |
Improve native transfers when a chain of IAX2 connections are in use.
(closes issue #7567) Reported by: tjd Patches: bug_7567_update_v2.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116884 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
30aedbade7 |
Try to fix attended transfers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116848 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
df6cd7a879 |
Merged revisions 116799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 lines Check to make sure an RTP structure exists before calling ast_rtp_new_source on it. (closes issue #12669) Reported by: sbisker ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116800 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Fredrickson
|
74c9d35cb5 |
Try to see if we can make our ringback situation a little better
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Sean Bright
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1e65b27439 |
Compile under dev-mode, please.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116765 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jim Dixon
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76707a409c |
Bring all app_rpt and chan_usbradio stuff up to date
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
f97d547aba |
Fixes a problem I was having with two SIP phones using Packet2Packet bridging dropping audio nearly immediately. The problem was that the lock on the SIP dialog was not being unlocked while the bridge was still active. (Related to issue #12566)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
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46423f6e09 |
Fix pedanticness.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
08f91c1192 |
Merged revisions 116463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Olle Johansson
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eecea3268e |
Don't add linefeed on received MESSAGE
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
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f07454f25d |
Properly declare charset for text messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116239 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Olle Johansson
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bb386c84e7 |
Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Olle Johansson
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47bf217ee8 |
Merged revisions 116230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines Accept text messages even with Content-Type: text/plain;charset=Södermanländska ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Olle Johansson
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29b1d73567 |
Add support for codec settings in originate via call file and manager.
This is to enable video and text in originated calls. Development sponsored by Omnitor AB, Sweden. (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
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9c2956a3b0 |
Reformatting
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Olle Johansson
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615ed013d3 |
Adding comments
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
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0ebec7fa4f |
Undo inadvertent changes to chan_skinny caused by the merging of urgent messaging
support. Thanks to Damien Wedhorn for pointing out the problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116138 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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739a3c88a5 |
Merged revisions 116038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines Fix a deadlock involving channel autoservice and chan_local that was debugged and fixed by mmichelson and me. We observed a system that had a bunch of threads stuck in ast_autoservice_stop(). The reason these threads were waiting around is because this function waits to ensure that the channel list in the autoservice thread gets rebuilt before the stop() function returns. However, the autoservice thread was also locked, so the autoservice channel list was never getting rebuilt. The autoservice thread was stuck waiting for the channel lock on a local channel. However, the local channel was locked by a thread that was stuck in the autoservice stop function. It turned out that the issue came down to the local_queue_frame() function in chan_local. This function assumed that one of the channels passed in as an argument was locked when called. However, that was not always the case. There were multiple cases in which this channel was not locked when the function was called. We fixed up chan_local to indicate to this function whether this channel was locked or not. The previous assumption had caused local_queue_frame() to improperly return with the channel locked, where it would then never get unlocked. (closes issue #12584) (related to issue #12603) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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8d18723961 |
Merged revisions 115944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines Use the right flag to open the audio in non-blocking. (closes issue #12616) Reported by: nicklewisdigiumuser ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Fredrickson
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a439ea6fe2 |
Need to clear calling_party_cat variable after we retrieve it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115941 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Fredrickson
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df175cebc3 |
Add support for receiving calling party category
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115939 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Brett Bryant
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9575b82389 |
A small change to fix iax2 native bridging.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Fredrickson
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5e3d36e4aa |
Add Zap MTP2 support to chan_zap
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Matthew Fredrickson
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1a492c49d4 |
Open up audio channel when we get ACM on SS7 event
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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7daebcd610 |
Adding support for "urgent" voicemail messages. Messages which are
marked "urgent" are considered to be higher priority than other messages and so they will be played before any other messages in a user's mailbox. There are two ways to leave an urgent message. 1. send the 'U' option to VoiceMail(). 2. Set review=yes in voicemail.conf. This will give instructions for a caller to mark a message as urgent after the message has been recorded. I have tested that this works correctly with file and ODBC storage, and James Rothenberger (who wrote initial support for this feature) has tested its use with IMAP storage. (closes issue #11817) Reported by: jaroth Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent Tested by: putnopvut, jaroth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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b280054c38 |
Merged revisions 115568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008) | 2 lines Remove debug output. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
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c961d9637f |
Merged revisions 115565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines Fix a race condition that bbryant just found while doing some IAX2 testing. He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes, however, the audio was extremely choppy. We looked at a packet trace and saw a storm of INVAL and VNAK frames being sent from one box to another. It turned out that what had happened was that one box tried to send a CONTROL frame before the 3 way handshake had completed. So, that frame did not include the destination call number, because it didn't have it yet. Part of our recent work for security issues included an additional check to ensure that frames that are supposed to include the destination call number have the correct one. This caused the frame to be rejected with an INVAL. The frame would get retransmitted for forever, rejected every time ... This race condition exists in all versions that got the security changes, in theory. However, it is really only likely that this would cause a problem in Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_ beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing all versions that could potentially be affected by the introduced race condition. These changes are what bbryant and I came up with to fix the issue. Instead of simply dropping control frames that get sent before the handshake is complete, the code attempts to wait a little while, since in most cases, the handshake will complete very quickly. If it doesn't complete after yielding for a little while, then the frame gets dropped. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
c02cf176e1 |
Merged revisions 115561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines Don't give up on attempting an outbound registration if we receive a 408 Timeout. (closes issue #12323) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Matthew Fredrickson
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4465c8704d |
Remove unused code as well as demote an error message to a debug message
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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25c75f6772 |
Let chan_h323 build in dev mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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9c549e6cf5 |
Merged revisions 115512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines Remove remnants of dlinkedlists. I didn't actually use them in the final version of my IAX2 improvements. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
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4555f32184 |
Remove redundant header getting.
(closes issue #12597) Reported by: hooi git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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e9f62e1d41 |
Change some NOTICE log messages to debug.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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27521f9e63 |
Remove my rant, since I have now replaced the rant with code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115315 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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2a966cdb03 |
Merged revisions 115304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines Avoid putting opaque="" in Digest authentication. This patch came from switchvox. It fixes authentication with Primus in Canada, and has been in use for a very long time without causing problems with any other providers. (closes issue AST-36) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115305 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
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b11854445b |
Add attributes to various API calls, to help track down bugs (and remove a deprecated function)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Brett Bryant
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4f3e4e22ef |
Add two new console commands "pri show version" and "ss7 show version" that will show the version of each library respectively.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115078 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
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c48c37909c |
Allow dringXrange to properly default to 10, as was done in 1.4.
dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified. (closes issue #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded by qwell (license 4) Tested by: bjm git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114922 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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f4237076bf |
Add support for specifying the registration expiry on a per registration basis in the register line. This comes from a Switchvox patch. (issue AST-24)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
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4c3aecfc55 |
Merged revisions 114890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines Don't crash on bad SIP replys. Fix created in Huntsville together with Mark M (putnopvut) (closes issue #12363) Reported by: jvandal Tested by: putnopvut, oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114899 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
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59f170973e |
Merged revisions 114891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4 These changes address a critical performance issue introduced in the latest release. The fix for the latest security issue included a change that made Asterisk randomly choose call numbers to make them more difficult to guess by attackers. However, due to some inefficient (this is by far, an understatement) code, when Asterisk chose high call numbers, chan_iax2 became unusable after just a small number of calls. On a small embedded platform, it would not be able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels. Ouch. These changes address some performance issues of the find_callno() function that have bothered me for a very long time. On every incoming media frame, it iterated through every possible call number trying to find a matching active call. This involved a mutex lock and unlock for each call number checked. So, if the random call number chosen was 20000, then every media frame would cause 20000 locks and unlocks. Previously, this problem was not as obvious since Asterisk always chose the lowest call number it could. A second container for IAX2 pvt structs has been added. It is an astobj2 hash table. When we know the remote side's call number, the pvt goes into the hash table with a hash value of the remote side's call number. Then, lookups for incoming media frames are a very fast hash lookup instead of an absolutely insane array traversal. In a quick test, I was able to get more than 3600% more IAX2 channels on my machine with these changes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114892 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Jeff Peeler
|
7cfd8389ac |
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
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63f5e27842 |
Merged revisions 114880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114884 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |