asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	e8c0be8fc2	Merged revisions 324479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324479 \| rmudgett \| 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) \| 1 line Comments and whitespace in chan_sip.c ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	b005d8dd53	Fixes issue with finding correct extension when message context is used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	ece8a5702a	Merged revisions 324237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324237 \| twilson \| 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) \| 12 lines Ignore media offers with a port of 0 Section 5.1 of RFC3264 states: A port number of zero in the offer indicates that the stream is offered but MUST NOT be used. (closes issue ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch uploaded by jacco (license 1277) Tested by: jacco, twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	34e2305ae7	Merged revisions 324048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 \| twilson \| 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) \| 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	abd7ef817e	Merged revisions 323370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 \| twilson \| 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) \| 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	00181729b4	Merged revisions 323371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323371 \| jrose \| 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) \| 12 lines Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this was causing NAT=Yes to always use rport when present which was against 1.6.2 behavior and the check itself was redundant since the only way this segment of code could be reached was if RPORT_PRESENT was already evaluated as true earlier. (closes issue ASTERISK-17789) Reported by: byronclark Patches: use_sip_nat_force_rport.patch uploaded by byronclark (license 1200) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	379370a396	Store sip peer name as var data on a outofcall msg. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	0bd877621e	Addition of "outofcall_message_context" sip.conf option. Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	4c459c2c85	Merged revisions 323040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323040 \| mnicholson \| 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) \| 5 lines Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop. (closes issue ASTERISK-17798) tested by mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	53ef4bfc16	Merged revisions 322807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322807 \| mnicholson \| 2011-06-09 12:37:07 -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/trunk@322808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	4cd9bc43c2	Merged revisions 322322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322322 \| irroot \| 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) \| 18 lines Make handle_request_publish do dialog expiration and destruction. This patch fixes handle_request_publish so that it does dialog expiration and destruction. Without this patch the incoming PUBLISH requests will get stuck in the dialog list. Restarting asterisk is the only way to remove them. Personal observation on one system the server hung up while looping through the channels rendering asterisk unusable and all sip phones unregisterd when they try reregister more requests are added. (closes issue #18898) Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: https://reviewboard.asterisk.org/r/1253 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ba625fa7d5	Correct some whitespace and a reference debug message. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	397c379a7d	Merged revisions 321812-321813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 \| rmudgett \| 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) \| 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 \| rmudgett \| 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) \| 1 line Constify subscription description parameter string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	9cd3cf2e71	Fix message destination extension. Don't send all messages to 's'. Get the destination from the request URI. (Found using automated test cases). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3f4d0e8743	Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	42907d40cd	Merged revisions 321511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321511 \| lmadsen \| 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) \| 8 lines Enhance NOTICE message to know who couldn't access the dialplan. (closes issue #19390) Reported by: lmadsen Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Murawki	9a7f807278	Merged revisions 321155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321155 \| markm \| 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) \| 10 lines Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic. Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails. (closes issue #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Murawki	0648d9595b	Merged revisions 321100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321100 \| markm \| 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) \| 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null pointer Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables (closes issue #19346) Reported by: kobaz Patches: netsock2.patch uploaded by kobaz (license 834) Tested by: kobaz, Marquis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	dbfac9cb55	Merged revisions 320883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320883 \| rmudgett \| 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) \| 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC request has incorrect To/From SIP headers. They are reversed and the dialog tags are the same when they should not be. If pedantic mode was disabled, then the cancel would have succeeded despite the incorrect message. * The SIP_OUTGOING flag was not set correctly for the dialog and I had to move some CC subscribe handling code as a result. * Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes in and the CC instance is not found, the 404 response was duplicated. JIRA AST-568 JIRA SWP-3493 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	12d7d81e6c	Merged revisions 320504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320504 \| jrose \| 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) \| 10 lines Fixes segfault occuring in chan_sip.c at __set_address_from_contact Checks to see if domain contains anything before sending it off to ast_sockaddr_resolve which is where the segfault was occuring due to null str. (closes issue #18857) Reported by: sybasesql Review: https://reviewboard.asterisk.org/r/1225/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	81bd779c24	Merged revisions 320180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320180 \| mnicholson \| 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) \| 16 lines 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/trunk@320181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f90bc95f0d	Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 \| jrose \| 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) \| 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	573108e63c	Merged revisions 319654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319654 \| twilson \| 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) \| 22 lines Merged revisions 319653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319653 \| twilson \| 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) \| 15 lines 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/trunk@319661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	99aaceacad	Merged revisions 319552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319552 \| twilson \| 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) \| 11 lines Unbreak the storing of registrations for restart The fix for issue 18882 broke retrieving non-realtime peers from the ast_db on restart/reload. This patch tries to unbreak things while leaving the intent of the original fix intact. (closes issue #19318) Reported by: remiq Patches: diff.txt uploaded by twilson (license 396) Tested by: lmadsen, remiq ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d34d46a16e	Merged revisions 319204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319204 \| twilson \| 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) \| 11 lines Merged revisions 319202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319202 \| twilson \| 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) \| 4 lines 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/trunk@319212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	980d896bde	Merged revisions 319145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319145 \| dvossel \| 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) \| 9 lines Merged revisions 319144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319144 \| dvossel \| 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) \| 2 lines 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/trunk@319146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	8e719c62b0	Merged revisions 319142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319142 \| mnicholson \| 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) \| 8 lines 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/trunk@319143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	32d43ebe19	When a error in T.38 negotiation happens or its rejected on a channel the state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	547490144c	Merged revisions 318917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318917 \| bbryant \| 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) \| 11 lines This patch allows TCP peers into the ast_db where they were previously restricted. (closes issue #18882) Reported by: cmaj Patches: patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	892b7a2efd	Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ 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/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	475c264bd2	Merged revisions 318550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318550 \| twilson \| 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) \| 2 lines Comment out the REF_DEBUG that slipped in during debugging ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	da4016544e	Merged revisions 318549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318549 \| twilson \| 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) \| 27 lines Merged revisions 318548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318548 \| twilson \| 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) \| 19 lines 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/ Review: https://reviewboard.asterisk.org/r/1210/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	07b3742ad2	Merged revisions 318337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318337 \| twilson \| 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) \| 18 lines Merged revisions 318331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318331 \| twilson \| 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) \| 12 lines 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/trunk@318400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4c35291c6b	Merged revisions 318233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318233 \| dvossel \| 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) \| 14 lines Merged revisions 318230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318230 \| dvossel \| 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) \| 7 lines 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/trunk@318234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	33b7cc2ef6	Merged revisions 317867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317867 \| russell \| 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) \| 10 lines chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. Don't duplicate variables on the sip_pvt. Just reset the variable list each time. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	ae8dbde4a8	Merged revisions 317865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317865 \| russell \| 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) \| 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't block doing silly deadlock avoidance. Just return and try again later. The funciton gets called often enough that it's fine. Also, this change was already made in trunk. (closes issue #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch uploaded by irroot (license 52) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	307f148adb	Merged revisions 317670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317670 \| rmudgett \| 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) \| 22 lines Fix SIP connected line updates. This patch fixes a couple SIP connected line update problems: 1) The connected line needs to be updated when the initial INVITE is sent if there is a peer callerid configured. Previously, the connected line information did not get reported until the call was connected so SIP could not report connected line information in ringing or progress messages. 2) The connected line should not be updated on initial connect if there is no connected line information. Previously, all it did was wipe out any default preset CONNECTEDLINE information set by the dialplan with empty strings. (closes issue #18367) Reported by: GeorgeKonopacki Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	0938974902	Merged revisions 317478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 \| russell \| 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) \| 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f0f5e237bf	Merged revisions 317474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317474 \| russell \| 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) \| 2 lines Fix more "set but unused" warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	932e34ee62	Merged revisions 317283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317283 \| jrose \| 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) \| 10 lines Resolves a deadlock that occurs during sip_new This is based on an uncommitted patch by jpeeler for the issue. Instead of relocking and then unlocking the channel though, we keep the lock on the channel until we are finished doing what we need to the channel. (closes issue #18441) Reported by: Alric ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	4d612d126b	Merged revisions 317281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317281 \| russell \| 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) \| 29 lines Merged revisions 317255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317255 \| russell \| 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) \| 22 lines 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/trunk@317282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	89da27b780	Merged revisions 317196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317196 \| mnicholson \| 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) \| 8 lines 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/trunk@317198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	1f96380da5	Reverts rev 316218 as it breaks parsing the [general] section of sip.conf. The functionality this patch attempts to achieve should already be possible using [general](+) in the config file. issue #17957 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	3bf4b09a6e	Merged revisions 316617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316617 \| dvossel \| 2011-05-04 08:44:41 -0500 (Wed, 04 May 2011) \| 19 lines 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/trunk@316618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ed56ae3ef7	If multiple [general] contexts occur from sip.conf (usually due to external includes), merge them. The original implementation of this did the merging of all contexts with the same name in the realtime layer, but that implementation severely breaks drivers which use the same context name (e.g. iax.conf, type={peer,user}). Therefore, the implementation needs to do the merging for particular entries only, based upon what contexts would allow that in the channel driver itself. This implementation is for chan_sip only, but others could be added in the future. (closes issue #17957) Reported by: marcelloceschia Patches: chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	db72ee299a	Merged revisions 316217 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316217 \| dvossel \| 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) \| 9 lines 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/trunk@316218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	e87639fc26	Merged revisions 315894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315894 \| mnicholson \| 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) \| 28 lines 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/trunk@315895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	181661c617	Merged revisions 315673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315673 \| twilson \| 2011-04-26 15:56:19 -0700 (Tue, 26 Apr 2011) \| 25 lines 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/trunk@315675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	bd354a0378	Make sure to create the caps structure for autocreated peers Because crashing is bad. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	1c14c67ce8	Merged revisions 315213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315213 \| russell \| 2011-04-25 14:04:28 -0500 (Mon, 25 Apr 2011) \| 14 lines Merged revisions 315212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r315212 \| russell \| 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) \| 7 lines Don't link non-cached realtime peers into the peers_by_ip container. (closes issue #18924) Reported by: wdoekes Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded by wdoekes (license 717) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	b8f253161b	Merged revisions 314550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314550 \| twilson \| 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) \| 13 lines Merged revisions 314549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314549 \| twilson \| 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) \| 6 lines Don't allocate more space than necessary for a sip_pkt This extra allocation is a hold-over from when pkt->data was a character array. Now that it is an allocated string, just allocate enough for the sip_pkt. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	642249c360	Merged revisions 314067 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314067 \| dvossel \| 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) \| 22 lines Remove the need for deadlock avoidance in chan_sip do_monitor. Deadlock avoidance between the sip pvt and the pvt->owner is very difficult. Now that channel's are ao2 objects, this complication is no longer necessary. It turns out the pvt's msg queue only exists because of deadlock avoidance (when deadlock avoidance fails msgs were added to a queue to be processed later), so this goes away as well. The technique used in the new sip_lock_pvt_full() function should be used as a template for replacing all locations where deadlock avoidance occurs between a channel tech_pvt and the pvt's owner. My hope is that this will begin a reversal of the invalid channel driver locking architecture we have been using for so long. This patch also resolves an issue where the pvt->owner gets unlocked during processing the msg queue. (closes issue #18690) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/1182/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4b4549106b	Merged revisions 314017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314017 \| dvossel \| 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) \| 17 lines sip codec negotiation of dynamic rtp payloads error fix This patch fixes how chan_sip handles dynamic rtp payload types it does not understand. At the moment if a dynamic payload's mime type does not match one we understand, the payload does not get removed from our payload table. As a result of this, the payload is set to whatever dynamic codec we use internally for that payload number on outgoing INVITES. This is incorrect. This patch fixes this by properly checking the rtpmap set function's return code to make sure it was found. The function can return both -1 and -2 depending on the source of the mismatch. We were just checking -1 explicitly. Review: https://reviewboard.asterisk.org/r/1169/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	b8b1d085db	Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ad30fa7569	Merged revisions 312889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312889 \| rmudgett \| 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) \| 5 lines Add 416 response to OPTIONS packet. RFC3261 Section 11.2 says the response code to an OPTIONS packet needs to be the same as if it were an INVITE. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e005f07b7d	Merged revisions 312866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312866 \| rmudgett \| 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) \| 15 lines Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension. The get_destination() function was not using the "s" extension when the request URI did not specify an extension. This is a regression caused when the URI parsing code was extracted into parse_uri(). Made get_destination() substitute the "s" extension when the parsed URI results in an empty string. (closes issue #18348) Reported by: shmaize Patches: issue18348_v1.8.patch uploaded by rmudgett (license 664) Tested by: shmaize ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f91462e7ca	Merged revisions 311352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311352 \| jrose \| 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) \| 10 lines Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL. This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings. (closes issue #18759) Reported by: bklang Patches: null-strings.patch uploaded by bklang (license 919) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	0d66e03bf4	Merged revisions 310231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310231 \| mmichelson \| 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) \| 9 lines Be more tolerant of what URI we accept for call completion PUBLISH requests. (closes issue #18946) Reported by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	070cb4ef87	Merged revisions 309256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309256 \| qwell \| 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) \| 15 lines Merged revisions 309255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309255 \| qwell \| 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) \| 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing is going to be done, so delme doesn't need to be set at all. Strangely, when this was added, this was being set to 1 in 1.6, and 0 in trunk. (issue AST-439) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	8e603ab4e1	Merged revisions 309084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309084 \| dvossel \| 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) \| 15 lines Merged revisions 309083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309083 \| dvossel \| 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) \| 9 lines Fixes thread blocking issue in the sip TCP/TLS implementation. (closes issue #18497) Reported by: vois Patches: issues_18497.diff uploaded by dvossel (license 671) Tested by: vois, rossbeer, kowalma, Freddi_Fonet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	b6e37118c9	Merged revisions 308945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308945 \| alecdavis \| 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) \| 21 lines Fix Deadlock with attended transfer of SIP call Call path sip_set_rtp_peer (locks chan then pvt) transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (locks p->owner) But by the time p->owner lock was attempted, seems as though chan and p->owner were different. So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods. (closes issue #18837) Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5deb544d06	Merged revisions 308679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 \| twilson \| 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) \| 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 \| twilson \| 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) \| 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b2ef13cb60	Merged revisions 307879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 \| rmudgett \| 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) \| 37 lines 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/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	08460fc094	Fixes bug in chan_sip where nativeformats are not set correctly. The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4f57a3bb7c	Merged revisions 306979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306979 \| twilson \| 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) \| 16 lines 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/trunk@307061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	a974d1a4ce	Merged revisions 306619 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306619 \| twilson \| 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) \| 24 lines Merged revisions 306618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306618 \| twilson \| 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) \| 17 lines 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/trunk@306670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2db3c9e058	Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	285d953fdf	Merged revisions 306215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306215 \| jpeeler \| 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) \| 20 lines Fix SIP deadlock involving state changes. Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) has caused locking problems. Both of these functions lock the channel when the channel argument is passed in! In this case, the suspected problem (the backtrace makes it impossible to tell) was the private being locked in sip_set_rtp_peer and then: transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to verify that the fix was only required in 1.8 and later.) (closes issue #18491) Reported by: cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f71322f239	Merged revisions 305923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 \| rmudgett \| 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) \| 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 \| rmudgett \| 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) \| 17 lines 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/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	175dd0ebf6	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/trunk@305759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	6908539952	Merged revisions 305254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305254 \| qwell \| 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) \| 24 lines Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 \| qwell \| 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) \| 17 lines 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/trunk@305255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	48a9694ed0	Merged revisions 304245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 \| mnicholson \| 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) \| 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 \| mnicholson \| 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) \| 13 lines 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/trunk@304246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	cd9221d2f6	Merged revisions 303962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303962 \| twilson \| 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) \| 30 lines Merged revisions 303960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303960 \| twilson \| 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) \| 23 lines 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/trunk@303963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	50c432324b	Merged revisions 303860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303860 \| tilghman \| 2011-01-25 12:55:27 -0600 (Tue, 25 Jan 2011) \| 12 lines Merged revisions 303858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303858 \| tilghman \| 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) \| 5 lines 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/trunk@303861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	e706b5706e	According to section 19.1.2 of RFC 3261: For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	06ac89965c	Merged revisions 302414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302414 \| seanbright \| 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) \| 7 lines Initialize an uninitialized variable. (closes issue #18640) Reported by: jcovert Patches: chan_sip.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	785e3a1417	Merged revisions 302314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302314 \| mnicholson \| 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) \| 18 lines Merged revisions 302313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302313 \| mnicholson \| 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) \| 11 lines 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/trunk@302315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	ae6b55e4a3	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 #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a0e4c4ee5b	Merged revisions 301790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301790 \| jpeeler \| 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) \| 42 lines Resolve deadlock involving REFER. Two fixes: 1) One must always have the private unlocked before calling pbx_builtin_setvar_helper to not invalidate locking order since it locks the channel. 2) Unlock the channel before calling pbx_find_extension, which starts and stops autoservice during the lookup. The problem scenario as illustrated by the reporter: Thread: do_monitor ----------------------- handle_request_do handle_incoming handle_request_refer ast_parking_ext_valid pbx_find_extension ast_autoservice_stop while (chan_list_state == as_chan_list_state) { usleep(1000); } Thread: autoservice_run ----------------------- autoservice_run chan = ast_waitfor_n ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / complex (depending on your system) ast_channel_lock(c[x]); handle_request_do and schedule_process_request_queue locks the owner if it exists. The autoservice thread is waiting for the channel lock, which wasn't ever released since the do_monitor thread was waiting for autoservice operations to complete. Solved by unlocking the channel but keeping a reference to guarantee safety. (closes issue #18403) Reported by: jthurman Patches: 20110103-blind_deadlock.diff uploaded by jthurman (license 614) issue18403.patch uploaded by jpeeler (license 325) Tested by: jthurman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c6858b9a1d	Merged revisions 301683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301683 \| twilson \| 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) \| 15 lines Merged revisions 301682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301682 \| twilson \| 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) \| 9 lines 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/trunk@301684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	783ea39ba1	Merged revisions 300521 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300521 \| lmadsen \| 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) \| 17 lines Merged revisions 300520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 \| lmadsen \| 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) \| 9 lines 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/trunk@300522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	94ef793caa	Merged revisions 300301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300301 \| twilson \| 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) \| 29 lines Merged revisions 300298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300298 \| twilson \| 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) \| 22 lines 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/trunk@300302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	ef23c07447	Merged revisions 299353 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299353 \| mnicholson \| 2010-12-21 09:25:03 -0600 (Tue, 21 Dec 2010) \| 30 lines Merged revisions 299242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299242 \| mnicholson \| 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) \| 23 lines 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/trunk@299355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	59ec959844	Merged revisions 299248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299248 \| mmichelson \| 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) \| 20 lines Fix a couple of CCSS issues. * Make sure to allocate a cc_params structure when creating autopeers. * Use sip_uri_cmp when retrieving SIP CC agents and monitors in case parameters appear in the URI. (closes issue #18504) Reported by: kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches: 18338.diff uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	cc0b7e7df5	Some scheduler API cleanup and improvements. Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	6307b6fe3a	Typos: recieved => received git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brad Watkins	806d69dc93	Merged revisions 298773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298773 \| marquis \| 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) \| 10 lines Fix parsing of mwi => lines in sip.conf Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. (closes issue #18350) Reported by: gbour Tested by: Marquis, gbour Review: https://reviewboard.asterisk.org/r/1053/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8ba7ff54b4	Merged revisions 298539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298539 \| tilghman \| 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) \| 8 lines Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. (closes issue #18464) Reported by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG (license 20) (plus a few additional lines by tilghman) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	30f81f902d	Merged revisions 297965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297965 \| twilson \| 2010-12-09 16:18:19 -0600 (Thu, 09 Dec 2010) \| 28 lines Merged revisions 297960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297960 \| twilson \| 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) \| 21 lines 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/trunk@297972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	537d235460	Merged revisions 297607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297607 \| jpeeler \| 2010-12-06 16:06:37 -0600 (Mon, 06 Dec 2010) \| 25 lines Merged revisions 297605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297605 \| jpeeler \| 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) \| 18 lines 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/trunk@297608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a46bd43ae8	Merged revisions 297075 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297075 \| jpeeler \| 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) \| 37 lines Merged revisions 297073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297073 \| jpeeler \| 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) \| 30 lines 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/trunk@297076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	40cc550f1f	Merged revisions 296628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296628 \| russell \| 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) \| 6 lines Complete some error handling in transmit_publish() in chan_sip.c. This error handling block caught my eye. It was missing a couple of things, but it should be safe now. Thanks to mmichelson for the quick peer review on IRC. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brad Watkins	ad56a4d16e	Merged revisions 296352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296352 \| marquis \| 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010) \| 12 lines Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. This had the effect of sending one NOTIFY for every time a sip peer made a call, in one case eventually overwhelming the phone and causing it to reboot. (closes issue #18342) Reported by: nivek Patches: issue0018342p1.patch uploaded by nivek (license 636) Tested by: nivek Review: https://reviewboard.asterisk.org/r/1029/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	e5ede71934	Merged revisions 295673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295673 \| twilson \| 2010-11-19 14:06:10 -0800 (Fri, 19 Nov 2010) \| 22 lines Merged revisions 295672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295672 \| twilson \| 2010-11-19 13:55:48 -0800 (Fri, 19 Nov 2010) \| 15 lines Merged revisions 295628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 \| twilson \| 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) \| 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	99a698efb7	Merged revisions 294734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294734 \| jpeeler \| 2010-11-11 15:58:25 -0600 (Thu, 11 Nov 2010) \| 32 lines Merged revisions 294733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 \| jpeeler \| 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) \| 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 \| jpeeler \| 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) \| 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	2df9e23e35	Merged revisions 294243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294243 \| mnicholson \| 2010-11-08 14:56:30 -0600 (Mon, 08 Nov 2010) \| 15 lines Merged revisions 294242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 \| mnicholson \| 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) \| 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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