asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	a77913a22d	One way audio before answering call waiting call on analog port. * Analog call waiting Caller ID spills could get stuck resulting in one way audio until the waiting call is answered. This only happens on the second (and later) call waiting call if the active call is not the first call. * The CLI/AMI "dahdi show channel" command could report the wrong channel information. Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer in sync. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	5153fbef97	Merged revisions 295710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295710 \| russell \| 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) \| 29 lines Fix cache of device state changes for multiple servers. This patch addresses a regression where device states across multiple servers were not being processing completely correctly. The code works to determine the overall state by looking at the last known state of a device on each server. However, there was a regression due to some invasive rewrites of how the cache works that led to the cache only storing the last device state change for a device, regardless of which server it was on. The code is set up to cache device state change events by ensuring that each event in the cache has a unique device name + entity ID (server ID). The code that was responsible for comparing raw information elements (which EID is) always returned a match due to a memcmp() with a length of 0. There isn't much code to fix the actual bug. This patch also introduces a new CLI command that was very useful for debugging this problem. The command allows you to dump the contents of the event cache. (closes issue #18284) Reported by: klaus3000 Patches: issue18284.rev1.txt uploaded by russell (license 2) Tested by: russell, klaus3000 (closes issue #18280) Reported by: klaus3000 Review: https://reviewboard.asterisk.org/r/1012/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	1930461fa7	Merged revisions 295672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295672 \| twilson \| 2010-11-19 13:55:48 -0800 (Fri, 19 Nov 2010) \| 15 lines Merged revisions 295628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 \| twilson \| 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) \| 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	ddb80391f6	Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (closes issue #18031) Reported by: rain Review: https://reviewboard.asterisk.org/r/1018/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7fbafbd5df	Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed initial value of struct analog_pvt.use_callerid. It may get forced on depending upon other config options. * Call analog_dnd() instead of manual inlined code. * Removed unused struct analog_pvt.usedistinctiveringdetection. * Removed the struct analog_pvt.unknown_alarm flag. It was really the struct analog_pvt.inalarm flag. * Use ast_debug() instead of ast_log(LOG_DEBUG). * Rename several function's index variable to idx. * Some formatting tweaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	b007bcc9b0	'sip notify clear-mwi' needs terminating CRLF. (closes issue #18275) Reported by: klaus3000 Patches: fix_body_CRLF_patch.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	f284b66dda	Merged revisions 295440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295440 \| pabelanger \| 2010-11-18 12:51:34 -0500 (Thu, 18 Nov 2010) \| 4 lines Fix compiler warnings when using openssl-dev 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	66e95a72d3	Add RedHat specific dependencies git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	14bf83e9d3	Uncomment settings under [global], to surpress warning when loading Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b63471b048	Merged revisions 295281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295281 \| rmudgett \| 2010-11-16 16:57:07 -0600 (Tue, 16 Nov 2010) \| 9 lines Merged revisions 295280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 \| rmudgett \| 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) \| 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	73547b0961	Check for pdftotext and give a useful error if not found. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	6a35147c4f	Remove intentional typo I had added when testing the check. oops. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	49750bb6e8	Check for wikiexport.py in PATH and give a useful error message if not found. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	583ca101ad	Merged revisions 295062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295062 \| tilghman \| 2010-11-15 12:24:02 -0600 (Mon, 15 Nov 2010) \| 9 lines Merged revisions 295026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295026 \| tilghman \| 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) \| 2 lines Create test verifying results of expression parser ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ab199924ac	Merged revisions 294988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294988 \| tilghman \| 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) \| 8 lines It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	f1abd401b9	Merged revisions 294910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294910 \| jpeeler \| 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) \| 4 lines Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	06ac20454e	Merged revisions 294904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294904 \| jpeeler \| 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) \| 23 lines Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 \| jpeeler \| 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) \| 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ac9434aecc	Merged revisions 294822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294822 \| rmudgett \| 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) \| 18 lines Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 \| rmudgett \| 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) \| 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2052ac8ecf	Remove CCSS architecture PDF. It has been moved to: https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f68807f063	Remove most of the contents of the doc dir in favor of the wiki content. This merge does the following things: * Removes most of the contents from the doc/ directory in favor of the wiki - http://wiki.asterisk.org/ * Updates the build_tools/prep_tarball script to know how to export the contents of the wiki in both PDF and plain text formats so that the documentation is still included in Asterisk release tarballs. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	01f31e0c50	Merged revisions 294733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 \| jpeeler \| 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) \| 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 \| jpeeler \| 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) \| 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	aafc5f4e99	Merged revisions 294639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294639 \| jpeeler \| 2010-11-11 13:31:00 -0600 (Thu, 11 Nov 2010) \| 53 lines Merged revisions 294384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294384 \| jpeeler \| 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) \| 47 lines Fix a deadlock in device state change processing. Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	c6973b1187	Fixing the Mac OS X build (bamboo warning) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1c692b86e5	Properly queue files with inotify(7). (closes issue #18089) Reported by: abelbeck Patches: 20101021__issue18089.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a4715fdf2d	Tweak a couple of CLI commands back to their original form. The "module" in this case is two parts, so there are two words before the verb of the CLI command. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2ee0c4ddff	Merged revisions 294500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294500 \| russell \| 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) \| 7 lines Improve a debug message to be more readable and consistent. (closes issue #18282) Reported by: klaus3000 Patches: ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1334febddc	Allow ast_do_masquerade() failure to be reported again. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	2d6a9d607f	Merged revisions 294429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294429 \| tilghman \| 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) \| 8 lines Detect GMime properly on systems where gmime flags and libs are configured with pkg-config. (closes issue #16155) Reported by: jcollie Patches: 20100917__issue16155.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3f9644b7db	Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b12f27f231	Merged revisions 294312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294312 \| jpeeler \| 2010-11-08 16:30:49 -0600 (Mon, 08 Nov 2010) \| 1 line add missing unlock not present in 294277 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	375b5b6b09	Merged revisions 294277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 \| jpeeler \| 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) \| 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	529b8fc988	Merged revisions 294242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 \| mnicholson \| 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) \| 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	3bb6f58420	Set a default waittime, and make sure to convert it to milliseconds git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4e39e38ac5	valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ******** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ******** JIRA SWP-2423 JIRA ABE-2614 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	e1a1360451	Fixed deadlock avoidance issues while locking channel when adding the Max-Forwards header to a request. (closes issue #17949) (closes issue #18200) Reported by: bwg Review: https://reviewboard.asterisk.org/r/997/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	225da545c5	Corret spelling and example git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	10317dacb6	Tell people to use the correct common name for clients as well git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Shaun Ruffell	c45ac8cc9c	Merged revisions 293969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293969 \| sruffell \| 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) \| 25 lines Merged revisions 293968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293968 \| sruffell \| 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) \| 17 lines codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically commit 9034) added the capability for the wctc4xxp to return more than a single packet of data in response to a read. However, when decoding packets, codec_dahdi was still assuming that the default number of samples was in each read. In other words, each packet your provider sent you, regardless of size, would result in 20 ms of decoded data (30 ms if decoding G723). If your provider was sending 60 ms packets then codec_dahdi would end up stripping 40 ms of data from each transcoded frame resulting in "choppy" audio. This would only affect systems where G729 packets are arriving in sizes greater than 20ms or G723 packets arriving in sizes greater than 30ms. DAHDI-744. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c34f40e710	Fixes ringback tone on sip semi-attended transfer. ABE-2168 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5cf1c93fd5	Do not output port in IPaddress for AMI sippeers. (closes issue #18248) Reported by: orn Patches: ami_sippeers.patch uploaded by pabelanger (license 224) Tested by: orn git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	252cffd38d	Merged revisions 293806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293806 \| rmudgett \| 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) \| 27 lines Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 \| rmudgett \| 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) \| 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6dfc9dddd8	Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	9a482b0724	Merged revisions 293723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 \| jpeeler \| 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) \| 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 \| jpeeler \| 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) \| 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	643d500d66	Merged revisions 293647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293647 \| rmudgett \| 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) \| 13 lines Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 \| rmudgett \| 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) \| 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	2c98e867f4	If manager and tls are disabled, do not display TCP/TLS Bindaddress. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ea00bff969	Analog 3-way call would not connect all parties if one was using sig_pri. Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	8b53a54dc3	Use ast_sockaddr_from_sin function not memcpy This resolves some IAX2 registration issue report on the asterisk-users mailing list. (closes issue #18202) Reported by: pabelanger Patches: update_registry.patch.v2 uploaded by pabelanger (license 224) Tested by: pabelanger, Nic Colledge (mailing list) Review: https://reviewboard.asterisk.org/r/993 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	de440fa10d	Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	651d43c698	Merged revisions 293417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293417 \| rmudgett \| 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) \| 9 lines Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 \| rmudgett \| 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) \| 1 line Remove some more code that serves no purpose. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	55a1b13da8	Merged revisions 293340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293340 \| rmudgett \| 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) \| 9 lines Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 \| rmudgett \| 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) \| 1 line Remove some code that serves no purpose. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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