6359 Commits (d1fbf03fdfd4238db24a3e8ba7f799edee19863a)

Author	SHA1	Message	Date
Matthew Nicholson	9ed4d9d89c	This commit modifies the way polling is done on TLS sockets. ... Because of the buffering the TLS layer does, polling is unreliable. If poll is called while there is data waiting to be read in the TLS layer but not at the network layer, the messaging processing engine will not proceed until something else writes data to the socket, which may not occur. This change modifies the logic around TLS sockets to only poll after a failed read on a non-blocking socket. This way we know that there is no data waiting to be read from the buffering layer. (closes issue #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	0640d5f616	Merged revisions 319652 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	e59b9085d7	Unlink a peer from peers_by_ip when expiring a registration ... Review: https://reviewboard.asterisk.org/r/1218/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d06dd1262f	Fixes issue with peer ref-counting during handle_request_subscribe. ... (closes issue #19293) Reported by: irroot git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	229e1541be	Make sure tcptls_session exists before dereferencing it. ... (closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9172658d42	Merged revisions 318734 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500 (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The applicable fixes for v1.4 are the SIP deadlock and the in progress masquerade check for multiple parties trying to pickup the same call. issue18654_v1.4.patch uploaded by rmudgett (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch uploaded by rmudgett (license 664) ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d12154f767	Merged revision 222981 from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 Similar deadlock possible when running the Pickup application internally. ------------------------------------------------------------------------ r222981 | dvossel | 2009-10-08 17:04:41 -0500 (Thu, 08 Oct 2009) | 13 lines Deadlock between ast_cel_report_event and ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner channel while only the pvt lock is held. Since pbx_exec calls ast_cel_report_event which attempts to lock the channel, invalid locking order occurs. Channels should be locked before pvt's. (closes issue #15512) Reported by: lmsteffan Patches: ast_cel_deadlock_15512.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	fe41f00a47	Clean up several chan_sip reference leaks ... Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	12e2f9cdeb	Don't offer video to directmedia callee unless caller offered it as well ... Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d693f81ab6	Fixes cases where sip_set_rtp_peer can return too early during media path reset. ... (closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	5081b488fe	Merged revisions 317719 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317719 | rmudgett | 2011-05-06 12:59:05 -0500 (Fri, 06 May 2011) | 11 lines Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.) Uninitialized variable. (issue #18640) (closes issue #18785) Reported by: pnlarsson Patches: issue18785_enegaard.patch uploaded by enegaard (license 1197) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	5fa33b9e28	Merged revisions 317211 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines chan_sip: fix broken realtime peer count, fix memory leak This patch addresses two bugs in chan_sip: 1) The count of realtime peers and users was off. The increment checked the value of the caching option, while the decrement did not. 2) Add a missing regfree() for a regex. (closes issue #19108) Reported by: vrban Patches: missing_regfree.patch uploaded by vrban (license 756) sip_object_counter.patch uploaded by vrban (license 756) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	b9bf3d604e	Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer ... abruptly disappears. This mostly occurs after a successful registration. (closes issue #17544) Reported by: marcelloceschia Patches: (modified) tcptls.patch uploaded by st (license 907) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	6b94fbacfa	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/1.6.2@316616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	abf3b8cf58	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/1.6.2@316329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	96ebbd8ddb	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/1.6.2@315893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6d933f5b74	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/1.6.2@315672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	8ac2b5b078	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/1.6.2@315212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	1bc9545d2e	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/1.6.2@315052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	826448948f	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/1.6.2@314958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	2939c487cd	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/1.6.2@314620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	dd044c85d7	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/1.6.2@314549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f33df943bc	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/branches/1.6.2@313435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	a9c87b24e4	white space change ... ........ 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/1.6.2@313433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	41a0ce02eb	fixes 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/1.6.2@313432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cef5e9d434	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/1.6.2@313189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	389b2a6907	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/1.6.2@312574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	a31990174b	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/1.6.2@309255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a3ff344e12	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/1.6.2@309083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	7c140ca0b5	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/1.6.2@308678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	368f8fc4cd	Merged revisions 307623 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307623 | rmudgett | 2011-02-10 18:29:17 -0600 (Thu, 10 Feb 2011) | 13 lines Reentrancy problem if outgoing call gets different B channel than requested. The chan_dahdi pri_fixup_principle() routine needs to protect the Asterisk channel with the channel lock when it changes the technology private pointer to a new private structure. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. Partial backport from v1.8 -r300714. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@307624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	5fd699dad0	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/1.6.2@306973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	bf7aea5263	Merged revisions 306617 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@306618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	56ec0fc138	Merged revisions 306119 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@306126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	050a3c39ec	Merged revisions 305888 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Andrew Latham	54dcf83e13	Replace link to old doc with new wiki page. ... Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	5f6b31daae	Merged revisions 305341 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines Obtain the pri lock for PRI queue counters. Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	1dbde4531e	Merged revisions 305252 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	afaf018322	Merged revisions 304241 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	48e5b9d57c	Merged revisions 304148 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	11110b1ead	Merged revisions 303906 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines Guard against retransmitting BYEs indefinitely In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0585c53d3f	Fix "sip show user <tab>", so that it actually shows results, instead of just completing the last entry. ... (closes issue #16675) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	569f4c6604	Merged revisions 303765 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	6d24696968	Merged revisions 303284 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	28cbf8801c	Merged revisions 302311 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines URI encode the user part of the contact header. ABE-2705 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a16089f243	Merged revisions 293493 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 [^] ........ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines 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 0017403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ [^] ........ Backporting a bugfix that should have been included. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	b381052ca0	Don't reject all SUBSCRIBE auth requests ... When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	f97b1161de	Fix backwards and broken XML documentation. ... (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	2722ad135e	Merged revisions 300216 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4c916b5474	Merged revisions 299624 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) | 5 lines Move check for extension existence below variable inheritance, due to the possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago