asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	f67e8ec980	Merged revisions 118251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118251 \| tilghman \| 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) \| 12 lines Realtime flag affects construction in multiple ways, so consulting whether rtcachefriends was set was done too soon (needed to be done inside build_peer, not just as a flag to build_peer). Also, fullcontact needed to be reconstructed, because realtime separates the embedded ';' into multiple fields. (closes issue #12722) Reported by: barthpbx Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) Tested by: barthpbx (Much of the discussion happened on #asterisk-dev for diagnosing this issue) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	1457c99ea8	Merged revisions 118163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118163 \| jpeeler \| 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) \| 1 line Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	de98d48a0d	- remove whitespaces between tags in received XML packets before giving them to the parser ; - report Gtalk error messages from a buddy to the console. This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation work with Empathy. Note that this is only true for audio streams, not video. Thank you to PH for his great help! (closes issue #12647) Reported by: PH Patches: trunk-12647-1.diff uploaded by phsultan (license 73) Tested by: phsultan, PH git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a668a87a80	Split the compile flags out and wire up some dependencies git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3c6aa2f5dc	Fix trunk breakage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	d46f9af7fa	A couple more places the frame data change was missed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	0985a2331a	one more place I forgot git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	6d018f0774	chan_console fixes because of ast_frame.data => ast_frame.data.ptr git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	5ceec8b052	oops git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	dbcef163a2	forgot chan_misdn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	04689cc5b3	Merged revisions 117582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117582 \| jpeeler \| 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) \| 2 lines Ensure that passed in zt_chan_conf structure is not modified in mkintf. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	19fd7beeb9	Merged revisions 117462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117462 \| jpeeler \| 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) \| 3 lines Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure. Another commit is following to make sure the zt_chan_conf structure is not modified. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	c126127fd5	Merged revisions 117574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117574 \| file \| 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) \| 2 lines Apply the autoframing setting to dialogs that do not get matched against a user or peer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Luigi Rizzo	e1ae86f643	do not die on SDL_ACTIVEEVENT reporting lost focus. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
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	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	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	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	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	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
Olle Johansson	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
Olle Johansson	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
Olle Johansson	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	9c2956a3b0	Reformatting git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	615ed013d3	Adding comments git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	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
Russell Bryant	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
Joshua Colp	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
Matthew Fredrickson	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
Matthew Fredrickson	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	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
Matthew Fredrickson	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	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
Mark Michelson	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
Russell Bryant	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	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

1 2 3 4 5 ...

5205 Commits (982959c04f8d25f5513c6c52ec47cf6c166fa084)