asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	1c14c67ce8	Merged revisions 315213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315213 \| russell \| 2011-04-25 14:04:28 -0500 (Mon, 25 Apr 2011) \| 14 lines Merged revisions 315212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r315212 \| russell \| 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) \| 7 lines Don't link non-cached realtime peers into the peers_by_ip container. (closes issue #18924) Reported by: wdoekes Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded by wdoekes (license 717) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	472e9aca3f	Merged revisions 315053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315053 \| alecdavis \| 2011-04-25 19:14:32 +1200 (Mon, 25 Apr 2011) \| 23 lines Merged revisions 315052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315052 \| alecdavis \| 2011-04-25 19:11:12 +1200 (Mon, 25 Apr 2011) \| 16 lines Merged revisions 315051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315051 \| alecdavis \| 2011-04-25 19:06:29 +1200 (Mon, 25 Apr 2011) \| 11 lines chan_local:check_bridge() misplaced misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked. (closes issue #19176) Reported by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis (license 585) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	73d8795841	Merged revisions 315001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315001 \| alecdavis \| 2011-04-23 10:59:18 +1200 (Sat, 23 Apr 2011) \| 12 lines chan_dahdi: Can't return to normal ring after distinctive ring on FXS clear a previous distinctivering pattern before each new call (closes issue #18985) Reported by: bromont Patches: bug18985.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, bromont ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	fae1fd4b1d	Merged revisions 314959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314959 \| mnicholson \| 2011-04-22 16:20:08 -0500 (Fri, 22 Apr 2011) \| 24 lines Merged revisions 314958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314958 \| mnicholson \| 2011-04-22 15:49:45 -0500 (Fri, 22 Apr 2011) \| 17 lines Merged revisions 311203,314908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311203 \| mnicholson \| 2011-03-17 14:14:37 -0500 (Thu, 17 Mar 2011) \| 4 lines Don't hold the pvt lock while streaming a file. ABE-2756 ........ r314908 \| mnicholson \| 2011-04-22 15:01:48 -0500 (Fri, 22 Apr 2011) \| 4 lines Prevent the login thread and the app threads from using the asterisk channel at the same time. ABE-2756 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	2b56cf085c	Merged revisions 314779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314779 \| tzafrir \| 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) \| 2 lines Fix a few typos (shown by Lintian) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0f1ff9141e	Implement AMI action PRIShowSpans. PRIShowSpans works like the AMI action DAHDIShowChannels but for PRI spans. It is similar to the CLI command "pri show spans". (closes issue #15980) Reported by: dwery git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	13e925b276	Simplify sig_pri.c:build_status(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c2676dc9dc	Merged revisions 314732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314732 \| rmudgett \| 2011-04-21 17:38:44 -0500 (Thu, 21 Apr 2011) \| 1 line Correct DAHDIShowChannels XML documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	b8f253161b	Merged revisions 314550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314550 \| twilson \| 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) \| 13 lines Merged revisions 314549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314549 \| twilson \| 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) \| 6 lines Don't allocate more space than necessary for a sip_pkt This extra allocation is a hold-over from when pkt->data was a character array. Now that it is an allocated string, just allocate enough for the sip_pkt. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	37274c73ee	Problems with ISDN MWI to phones. The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	642249c360	Merged revisions 314067 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314067 \| dvossel \| 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) \| 22 lines Remove the need for deadlock avoidance in chan_sip do_monitor. Deadlock avoidance between the sip pvt and the pvt->owner is very difficult. Now that channel's are ao2 objects, this complication is no longer necessary. It turns out the pvt's msg queue only exists because of deadlock avoidance (when deadlock avoidance fails msgs were added to a queue to be processed later), so this goes away as well. The technique used in the new sip_lock_pvt_full() function should be used as a template for replacing all locations where deadlock avoidance occurs between a channel tech_pvt and the pvt's owner. My hope is that this will begin a reversal of the invalid channel driver locking architecture we have been using for so long. This patch also resolves an issue where the pvt->owner gets unlocked during processing the msg queue. (closes issue #18690) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/1182/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4b4549106b	Merged revisions 314017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314017 \| dvossel \| 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) \| 17 lines sip codec negotiation of dynamic rtp payloads error fix This patch fixes how chan_sip handles dynamic rtp payload types it does not understand. At the moment if a dynamic payload's mime type does not match one we understand, the payload does not get removed from our payload table. As a result of this, the payload is set to whatever dynamic codec we use internally for that payload number on outgoing INVITES. This is incorrect. This patch fixes this by properly checking the rtpmap set function's return code to make sure it was found. The function can return both -1 and -2 depending on the source of the mismatch. We were just checking -1 explicitly. Review: https://reviewboard.asterisk.org/r/1169/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	6b01eb3324	Consolidate all new call calls to run through new setsubstate_ringout. (closes issue #17907) Reported by: wedhorn Patches: cleanup.stateringout.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4f8d56a824	Merged revisions 313780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313780 \| rmudgett \| 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) \| 20 lines Leftover debug messages unconditionally sent to the console. Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config option enabled outputs the following debug messages unconditionally: Dialing T1847555121 on 1 Dialing www2w on 1 * Made debug messages in my_dial_digits() normal debug messages that do not get output unless enabled. * Reworded some debug messages in my_dial_digits() to be clearer. * Replace strncpy() with ast_copy_string() in my_dial_digits() which does the same job better. (closes issue #18847) Reported by: vmikhelson Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	b8b1d085db	Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	a6695b84ce	Merged revisions 313435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 also went ahead and fixed the problem it introduces before committing. ........ r313435 \| jrose \| 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) \| 1 line fixing stupid mistake with putting code before variable declaration ........ Merged revisions 313433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313432 \| jrose \| 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) \| 14 lines reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ ........ ........ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bc907695bf	Merged revisions 313190 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313190 \| rmudgett \| 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) \| 39 lines Merged revisions 313189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313189 \| rmudgett \| 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) \| 32 lines Merged revisions 313188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313188 \| rmudgett \| 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) \| 25 lines Stuck channel using FEATD_MF if caller hangs up at the right time. The cause was actually a caller hanging up just at the end of the Feature Group D DTMF tones that setup the call. The reason for this is a "guard timer" that's implemented using ast_safe_sleep(100). If the caller happens to hang up AFTER the final tone of the DTMF string but BEFORE the end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero. This causes the code to bounce to the end of ss_thread(), but it does NOT tear down the call properly. This should be a rare occurrence because the caller has to hang up at EXACTLY the right time. Nonetheless, it was happening quite regularly on the reporter's system. It's not easily reproducible, unless you purposely increase the guard-time to 2000 or more. Once you do that, you can reproduce it every time by watching the DTMF debug and hanging up just as it ends. Simply add an ast_hangup() before goto quit. (closes issue #15671) Reported by: jcromes Patches: issue15671.patch uploaded by pabelanger (license 224) Tested by: jcromes ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ce17f956dc	Add private lock deadlock avoidance callback to PRI and SS7. Factor out the equivalent function for analog. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	1e33d71b79	Merged revisions 313001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313001 \| alecdavis \| 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) \| 13 lines Fix ISDN calling subaddr User Specified Odd/Even Flag Calculation of the Odd/Even flag was wrong. Implement correct algo, and set odd/even=0 if data would be truncated. Only allow automatic calculation of the O/E flag, don't let dialplan influence. (closes issue #19062) Reported by: festr Patches: bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by: festr, alecdavis, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	698a356737	Merged revisions 312949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312949 \| rmudgett \| 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) \| 6 lines Crash if ISDN span layer 1 is down on initial load. Regression from -r312575 B channel shifting during negotiation. * Also combine updating the alarm flag with clearing the resetting flag. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ad30fa7569	Merged revisions 312889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312889 \| rmudgett \| 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) \| 5 lines Add 416 response to OPTIONS packet. RFC3261 Section 11.2 says the response code to an OPTIONS packet needs to be the same as if it were an INVITE. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e005f07b7d	Merged revisions 312866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312866 \| rmudgett \| 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) \| 15 lines Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension. The get_destination() function was not using the "s" extension when the request URI did not specify an extension. This is a regression caused when the URI parsing code was extracted into parse_uri(). Made get_destination() substitute the "s" extension when the parsed URI results in an empty string. (closes issue #18348) Reported by: shmaize Patches: issue18348_v1.8.patch uploaded by rmudgett (license 664) Tested by: shmaize ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	121b90a47d	Remove the channel parameter from sig_pri_handle_subcmds(). It was only used in a debug message and may not be correct anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e1ceb52b51	Merged revisions 312575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312575 \| rmudgett \| 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) \| 52 lines Merged revisions 312574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312574 \| rmudgett \| 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) \| 45 lines Merged revisions 312573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312573 \| rmudgett \| 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) \| 38 lines Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6826b083ec	Merged revisions 312509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312509 \| rmudgett \| 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) \| 22 lines When a call going out an NT-PTMP port gets rejected, Asterisk crashes. If a call is sent to an ISDN phone that rejects the call with RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes. I could not get my setup to crash. However, I could see the possibility from a race condition between queuing an AST_CONTROL_BUSY to the core and then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is processed before the AST_CONTROL_HANGUP is queued, the ast_channel could be destroyed out from under chan_misdn. Avoid this particular crash scenario by not queueing the AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy JIRA SWP-2679 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312510 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	759bf6b840	Fixing bad line break from 312384 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	846cfa0ef0	New Feature for chan_dahdi. 4 length pattern matching. In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ee44bf7257	Merged revisions 312022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312022 \| rmudgett \| 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) \| 14 lines chan_misdn segfaults when DEBUG_THREADS is enabled. The segfault happens because jb->mutexjb is uninitialized from the ast_malloc(). The internals of ast_mutex_init() were assuming a nonzero value meant mutex tracking initialization had already happened. Recent changes to mutex tracking code to reduce excessive memory consumption exposed this uninitialized value. Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc(). Also eliminated redundant zero initialization code in the routine. (closes issue #18975) Reported by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8dce4dbe2a	Merged revisions 311874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311874 \| rmudgett \| 2011-03-29 20:56:05 -0500 (Tue, 29 Mar 2011) \| 1 line Update some setup_dahdi_int() comments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	15f633294d	Merged revisions 311612 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311612 \| bbryant \| 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) \| 9 lines Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null value. (closes issue #18821) Reported by: cmaj Patches: patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx uploaded by cmaj (license 830) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	82ef85f20b	Merged revisions 311558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311558 \| twilson \| 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) \| 5 lines Don't use static declared buf in parse_name_andor_addr This function isn't used anywhere yet, but we definitely don't want to keep the same value for buf between calls to the function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f91462e7ca	Merged revisions 311352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311352 \| jrose \| 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) \| 10 lines Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL. This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings. (closes issue #18759) Reported by: bklang Patches: null-strings.patch uploaded by bklang (license 919) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d000b76ebc	Merged revisions 311297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311297 \| rmudgett \| 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) \| 12 lines Race condition when ISDN CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY could sometimes be processed before the call_forward dial string is recognized. * Moved setting the call_forwarding dial string after sending a response to the initiator and just queue an empty frame to wake up the media thread instead of an AST_CONTROL_BUSY. * Added check for empty rerouting/deflection number and respond with an error. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	0d66e03bf4	Merged revisions 310231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310231 \| mmichelson \| 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) \| 9 lines Be more tolerant of what URI we accept for call completion PUBLISH requests. (closes issue #18946) Reported by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f7b7223fb6	Merged revisions 310088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310088 \| jrose \| 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) \| 9 lines Returns with an error notice if CHANNEL function of SIP channel is read without arguments. (Closes issue #18653) Reported by: wuwu Patches: diff.patch uploaded by jrose (license 1225) Tested by: jrose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c551e9105d	Merged revisions 309994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309994 \| rmudgett \| 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) \| 1 line Make pri parameter description consistent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6de1332214	Merged revisions 309808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 \| tilghman \| 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) \| 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 \| tilghman \| 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) \| 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Moises Silva	0f207dce6e	Merged revisions 309720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309720 \| moy \| 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) \| 6 lines Fix caller id passed to openr2_chan_make_call (closes issue #18894) Reported by: malufrj Tested by: moy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c1ba13c1ea	Fix a buglet that prevented chan_nbs from loading (and subsequently stopped Asterisk). In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	928ec2b990	Merged revisions 309445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 \| rmudgett \| 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) \| 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	070cb4ef87	Merged revisions 309256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309256 \| qwell \| 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) \| 15 lines Merged revisions 309255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309255 \| qwell \| 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) \| 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing is going to be done, so delme doesn't need to be set at all. Strangely, when this was added, this was being set to 1 in 1.6, and 0 in trunk. (issue AST-439) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	72260849b7	Merged revisions 309126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309126 \| rmudgett \| 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) \| 16 lines Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal. Looks like an unintended change when sig_analog.c was extracted from chan_dahdi.c. Removed useless conditional around needed code and fixed resulting compiler warning. (closes issue #18667) Reported by: enegaard Patches: issue18667.patch uploaded by enegaard (license 1197) Tested by: enegaard JIRA SWP-2965 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309127 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	8e603ab4e1	Merged revisions 309084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309084 \| dvossel \| 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) \| 15 lines Merged revisions 309083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309083 \| dvossel \| 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) \| 9 lines Fixes thread blocking issue in the sip TCP/TLS implementation. (closes issue #18497) Reported by: vois Patches: issues_18497.diff uploaded by dvossel (license 671) Tested by: vois, rossbeer, kowalma, Freddi_Fonet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	b6e37118c9	Merged revisions 308945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308945 \| alecdavis \| 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) \| 21 lines Fix Deadlock with attended transfer of SIP call Call path sip_set_rtp_peer (locks chan then pvt) transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (locks p->owner) But by the time p->owner lock was attempted, seems as though chan and p->owner were different. So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods. (closes issue #18837) Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5deb544d06	Merged revisions 308679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 \| twilson \| 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) \| 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 \| twilson \| 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) \| 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	2b063d4dca	Merged revisions 308622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308622 \| rmudgett \| 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) \| 9 lines sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails. (closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b79adb645e	Add more verbage to CLI command 'pri show channels' usage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4a48600231	Add CLI "pri show channels" command. List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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6924 Commits (afc8368deaab417a5e481c1f8709737cfb025560)