6884 Commits (e5e49b9c52a0556f19fd6a5a126f25318e83601c)

Author	SHA1	Message	Date
David Vossel	e389fab7dd	Merged revisions 316616 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) | 12 lines Fixes session-timers=refuse not being enforced for *caller* During handle_request_invite, the session timer mode was retrieved from a cached variable. This patch forces a peer lookup of the session timer mode in the case of an incoming invite. (closes issue #18804) Reported by: wdoekes Patches: issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	1c252efce5	Use htons() instead of ntohs() in some places. ... (closes issue #19200) Reported by: wdoekes Patches: issue19200-trunk.patch uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	27ef94eb74	Merged revisions 316329 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500 (Tue, 03 May 2011) | 17 lines Merged revisions 316328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (closes issue #19053) Reported by: oej Tested by: oej Review: https://reviewboard.asterisk.org/r/1158/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	a82f1bb995	Fix a bunch of compiler warnings generated by gcc 4.6.0. ... Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c5ad2f12a0	The dahdi_hangup() call does not clean up the channel fully. ... After dahdi_hangup() has supposedly hungup an ISDN channel there is still traffic on the S0-bus because the channel was not cleaned up fully. Shuffled the hangup code to include some missing cleanup. Also fixed some code formatting in the area. I think the primary missing clean up code was the call to tone_zone_play_tone() to turn off any active tones on the channel. (closes issue #19188) Reported by: jg1234 Patches: issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested by: jg1234 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	981716535a	Never put the Require: timer header in an Invite. ... This has already been discussed and should have been resolved earlier. View revsion 285565's log for more information about why it is important to not put timer in the Require header. (closes issue #18704) Reported by: mfrager git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	e8210addf8	Merged revisions 315893 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e4ef679575	Merged revisions 315672 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines Make sure unregistering a peer unlinks it from the peer container Instead of mostly copying the code from expire_register, just use the function that "does the right thing". (closes issue #16033) Reported by: kkm Patches: 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) Tested by: kkm, tilghman, twilson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	4e99831b16	chan_local: resolve a deadlock. ... This patch resolves a fairly complex deadlock that can occur with the combination of chan_local and a dialplan switch, such as dynamic realtime extensions, which pulls autoservice into the picture when doing a dialplan lookup. (closes issue #18818) Reported by: nic Patches: issue18818.patch uploaded by jthurman (license 614) 18818.v1.txt uploaded by russell (license 2) Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ced679eef9	When using MGCP realtime gateway definitions, random crashes occur. ... Fixed incorrect linked list node removal for realtime gateways. (closes issue #18291) Reported by: nahuelgreco Patches: dangling-pointers-when-pruning.patch uploaded by nahuelgreco (license 162) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	f575dd5397	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/branches/1.8@315213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	d67b2b00b3	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/branches/1.8@315053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	f014ffa9d0	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/branches/1.8@315001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1e0234afd6	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/branches/1.8@314959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	51be3664e1	Fix a few typos (shown by Lintian) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1310fd4175	Correct DAHDIShowChannels XML documentation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	4468fe047e	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/branches/1.8@314628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	459ab135c2	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/branches/1.8@314550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	4c1dd375f7	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/branches/1.8@314067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	2998c62fc4	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/branches/1.8@314017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	89f98df5d8	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/branches/1.8@313780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	2600de8c9f	fixing stupid mistake with putting code before variable declaration ... ........ Merged revisions 313435 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/branches/1.8@313436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	833c42ce4b	Merged revisions 313432 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/branches/1.8@313434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	46067d2dc4	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/branches/1.8@313190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	20ef1e9c95	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/branches/1.8@313001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4242cb82f4	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/branches/1.8@312949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ddc3fac28b	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/branches/1.8@312889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0acdb60dbd	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/branches/1.8@312866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	458a57d1d3	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/branches/1.8@312575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d01f7b6dd8	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/branches/1.8@312509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	32e0a3510c	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/branches/1.8@312022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	28bfbccfb7	Update some setup_dahdi_int() comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	a54ab29087	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/branches/1.8@311612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2f95620a2f	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/branches/1.8@311558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	7cf95da39a	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/branches/1.8@311352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4f3cf039f4	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/branches/1.8@311297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	d409098f0e	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/branches/1.8@310231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ed3e04e831	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/branches/1.8@310088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8bfde13607	Make pri parameter description consistent. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	56cd7709a5	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/branches/1.8@309808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Moises Silva	3770b4d7cb	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/branches/1.8@309720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5b9f9f78ca	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/branches/1.8@309445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	546b652f1f	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/branches/1.8@309256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	10e378b07c	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/branches/1.8@309126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	0a577f6cdd	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/branches/1.8@309084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	73254a28cb	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/branches/1.8@308945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f0992a0b5e	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/branches/1.8@308679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7b353a26ae	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/branches/1.8@308622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a5f6367057	No response sent for SIP CC subscribe/resubscribe request. ... Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	cd3b672f45	Merged revisions 306973 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago