asterisk

Commit Graph

Author	SHA1	Message	Date
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

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3102 Commits (c33e1b0e27065d4cd1d22ff19f26633cf082e20f)