2833 Commits (ffb13d7d3866963247ddbea30d302af0a0fd7bf2)

Author	SHA1	Message	Date
Matthew Nicholson	ffb13d7d38	unlock pvt when we drop voice frames received in early media when in t.38 mode ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	70bf4a47f4	fix for previous commit ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	52c395fb1f	Merged revisions 322646 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	14 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 years ago
Matthew Nicholson	077b8b9956	Merged revisions 299194,299198,299220 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines Respond as soon as possible with a 202 Accepted to refer requests. This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 ........ r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines Remove changes to via processing that were not supposed to go into the last commit. ........ r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use ast_free() instead of free() ABE-2656 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	c448bc9800	Typos: recieved => received ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	1511efcbb3	Merged revisions 297959 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b6e84113ef	Merged revisions 297603 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	5b02a16a61	Merged revisions 297185 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5 lines If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event. If we answer 481 the subscription that we don't want will be cancelled. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	0ad0a9b85e	Merged revisions 297072 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5aaefe7f08	Merged revisions 295628 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@295672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	744ece2574	Merged revisions 294688 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@294733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	9e8ad57efc	Go off hold when we get an empty reinvite telling us to. ... (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@294242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	58c28ab1dc	Merged revisions 293722 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@293723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	4f361397b9	Update the LDIF file for LDAP. ... The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@292786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3442a369aa	Merged revisions 291392 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@291393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago